Cisco IOS Voice Command Reference. C O N T E N T S. Cisco IOS Voice
Commands: A VR-1 aaa nas port voip VR-2 aaa username VR-3 access-list (
voice ...
Cisco IOS Voice Command Reference November 2010
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Cisco IOS Voice Commands: Z VR-3127 zone access zone bw
VR-3128
VR-3130
zone circuit-id
VR-3131
zone cluster local
VR-3133
zone cluster remote zone local zone prefix
VR-3134
VR-3136 VR-3138
zone remote
VR-3142
zone subnet
VR-3144
Cisco IOS Voice Command Reference
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Contents
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A This chapter contains commands to configure and maintain Cisco IOS voice applications. The commands are presented in alphabetical order. Some commands required for configuring voice may be found in other Cisco IOS command references. Use the Cisco IOS Master Commands List online to find these commands. For detailed information on how to configure these applications and features, refer to the Cisco IOS Voice Configuration Library.
Cisco IOS Voice Command Reference
VR-1
Cisco IOS Voice Commands: A aal2-profile custom
aal2-profile custom To specify custom numbers and user-to-user information (UUI) code points for ATM adaptation layer 2 (AAL2) profiles and codecs, use the aal2-profile custom command in global configuration mode. To disable the configuration, use the no form of this command. aal2-profile custom number number number {clear-channel | g711alaw | g711ulaw | g726r32 | g729br8 | g720r8 | llcc} packet-length minimum-UUI-codepoint maximum-UUI-codepoint no aal2-profile custom number
Syntax Description
number
AAL profile number. For more information, use the question mark (?) online help function.
clear-channel | g711alaw | Specifies the types of codec as follows: g711ulaw | g726r32 | • Clear Channel g729br8 | g720r8 | llcc • G.711 a-law •
G.711-mu-law
•
G.726r32
•
G.729 ANNEX-B 8000 bits per second
•
G.729 8000 bps
•
Lossless Compression
packet-length
Packet length in octets. The range is from 5 to 64.
minimum-UUI-codepoint
Minimim UUI code point. The range is from 0 to 15.
maximum-UUI-codepoint
Maximum UUI code point. The range is from 0 to 15.
Command Default
One of the predefined International Telecommunication Union - Telecommunication Standardization Sector (ITU-T) profiles can be used.
Command Modes
Global configuration (config)
Command History
Release
Modification
15.0(1)M
This command was introduced in a release earlier than Cisco IOS Release 15.0(1)M.
Usage Guidelines
AAL2 custom profiles are used to define additional profiles that are not present in the ITU-T specifications. After defining a custom profile, apply that profile under a Voice over ATM (VoATM) dial peer for it to take affect using the codec aal2-profile command. The codec aal2-profile command can be used only if the session protocol is "aal2-trunk".
Cisco IOS Voice Command Reference
VR-2
Cisco IOS Voice Commands: A aal2-profile custom
Examples
The following example shows how to specify custom numbers and UUI cod epoints for AAL2 profiles and codecs: Router# configure terminal Router(config)# aal2-profile custom
2 1 1 g711ulaw 6 3 3
Cisco IOS Voice Command Reference
VR-3
Cisco IOS Voice Commands: A aaa nas port voip
aaa nas port voip To send out the standard NAS-port attribute (RADIUS IETF Attribute 5) on voice interfaces, use the aaa nas port voip command in global configuration mode. To disable the command, use the no form of the command. aaa nas port voip no aaa nas port voip
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled
Command Modes
Global configuration
Command History
Release
Modification
12.2(11)T
This command was introduced on the Cisco AS5300.
Usage Guidelines
This command brings back the original behavior of the Authentication, Authorization, and Accounting (AAA). NAS-Port on VoIP interfaces. By default this feature is disabled.
Examples
The following example shows how to return to the original behavior of the AAA NAS-Port: aaa nas port voip
Related Commands
Command
Description
aaa nas port extended
Replaces the NAS-port attribute with RADIUS IETF attribute 26 and displays extended field information.
Cisco IOS Voice Command Reference
VR-4
Cisco IOS Voice Commands: A aaa username
aaa username To determine the information with which to populate the username attribute for Authentication, Authorization, and Accounting (AAA). billing records, use the aaa username command in SIP UA configuration mode. To achieve default capabilities, use the no form of this command. aaa username {calling-number | proxy-auth} no aaa username
Syntax Description
calling-number
Uses the FROM: header in the SIP INVITE (default value). This keyword is used in most implementations.
proxy-auth
Parses the Proxy-Authorization header. Decodes the Microsoft Passport user ID (PUID) and password, and then populates the PUID into the username attribute and a "." into the password attribute. The username attribute is used for billing, and the “.” is used for the password, because the user has already been authenticated before this point.
Command Default
calling-number
Command Modes
SIP UA configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced on the Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco AS5300, Cisco AS5350, and the Cisco AS5400.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. This command does not support the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
12.2(11)T
This command was integrated Cisco IOS Release 12.2(11)T and was implemented on the Cisco AS5850. This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5800 in this release.
Usage Guidelines
Parsing the Proxy-Authorization header, decoding the PUID and password, and populating the username attribute with the PUID must be enabled through this command. If this command is not issued, the Proxy-Authorization header is ignored. The keyword proxy-auth is a nonstandard implementation, and Session Initiation Protocol (SIP) gateways do not normally receive or process the Proxy-Authorization header.
Cisco IOS Voice Command Reference
VR-5
Cisco IOS Voice Commands: A aaa username
Examples
The following example enables the processing of the SIP username from the Proxy-Authorization header: Router(config)# sip-ua Router(config-sip-ua)# aaa username proxy-auth
Related Commands
Command
Description
show call active voice
Displays sactive call information for voice calls or fax transmissions in progress.
show call history voice Displays the voice call history table.
Cisco IOS Voice Command Reference
VR-6
Cisco IOS Voice Commands: A access-list (voice source-group)
access-list (voice source-group) To assign an access list to a voice source group, use the access-list command in voice source-group configuration mode. To delete the access list, use the no form of this command. access-list access-list-number no access-list access-list-number
Syntax Description
access-list-number
Command Default
No default behavior or values
Command Modes
Voice source-group configuration
Command History
Release
Modification
12.2(11)T
This command was introduced in voice source-group configuration mode.
Usage Guidelines
Number of an access list. The range is from 1 to 99.
An access list defines a range of IP addresses for incoming calls that require additional scrutiny. Two related commands are used for voice source groups: •
Use the access-list access-list-number {deny | permit} source [source-wildcard] [log] command in global configuration mode to define the contents of the access list.
•
Use the access-list access-list-number command in voice source-group configuration mode to assign the defined access list to the voice source group.
The terminating gateway uses the source IP group to identify the source of the incoming VoIP call before selecting an inbound dial peer. If the source is found in the access list, then the call is accepted or rejected, depending on how the access list is defined. The terminating gateway uses the access list to implement call blocking. If the call is rejected, the terminating gateway returns a disconnect cause to the source. Use the disconnect-cause command to specify a disconnect cause to use for rejected calls. Use the show access-lists privileged EXEC command to display the contents of all access lists. Use the show ip access-list privileged EXEC command to display the contents of one access list.
Examples
The following example assigns access list 1 to voice source-group alpha. Access list 1 was defined previously using another command. An incoming source IP group call is checked against the conditions defined for access list 1 and is processed based on the permit or deny conditions of the access list. Router(config)# voice source-group alpha Router(cfg-source-grp)# access-list 1
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A access-list (voice source-group)
Related Commands
Command
Description
carrier-id (dial peer)
Specifies the carrier as the source of incoming VoIP calls (for carrier ID routing).
disconnect-cause
Specifies a cause for blocked calls.
h323zone-id (voice source group)
Associates a zone for an incoming H.323 call.
show access-lists
Displays the contents of all access lists.
show ip access-list
Displays the contents of one access list.
translation-profile (source group)
Associates a translation profile with incoming source IP group calls.
trunk-group-label (voice source group)
Specifies the trunk group as the source of incoming VoIP calls (for trunk group label routing).
voice source-group
Initiates the source IP group profile definition.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A access-policy
access-policy To require that a neighbor be explicitly configured in order for requests to be accepted, use the access-policy command in Annex G configuration mode. To reset the configuration to accept all requests, use the no form of this command. access-policy [neighbors-only] no access-policy
Syntax Description
neighbors-only
Command Default
Border elements accept any and all requests if service relationships are not configured.
Command Modes
Annex G configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
(Optional) Requires that a neighbor be configured.
Usage Guidelines
Border elements accept any and all requests if service relationships are not configured. The access-policy command eliminates arbitrary requests from unknown border elements, and is a required prerequisite for configuring service relationships.
Examples
The following example shows how to enable the service relationship between border elements: Router(config-annexg)# access-policy neighbors-only
Related Commands
Command
Description
call-router
Enables the Annex G border element configuration commands.
domain-name
Sets the domain name reported in service relationships.
Cisco IOS Voice Command Reference
VR-9
Cisco IOS Voice Commands: A accounting (gatekeeper)
accounting (gatekeeper) To enable and define the gatekeeper-specific accounting method, use the accounting command in gatekeeper configuration mode. To disable gatekeeper-specific accounting, use the no form of this command. accounting {username h323id | vsa} no accounting
Syntax Description
username h323id
Enables H323ID in the user name field of accounting record.
vsa
Enables the vendor specific attribute accounting format.
Command Default
Accounting is disabled.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
12.1(5)XM
The vsa keyword was added.
12.2(2)T
The vsa keyword was integrated into Cisco IOS Release 12.2(2)T.
12.2(2)XB1
This command was implemented on the Cisco AS5850 universal gateway.
12.2(33)SRA
This command was integrated into Cisco IOS Release 12.2(33)SRA.
12.3(9)T
This username h323id keyword was added.
Usage Guidelines
To collect basic start-stop connection accounting data, the gatekeeper must be configured to support gatekeeper-specific H.323 accounting functionality. The accounting command enables you to send accounting data to the RADIUS server via IETF RADIUS or VSA attriibutes. Specify a RADIUS server before using the accounting command. There are three different methods of accounting. The H.323 method sends the call detail record (CDR) to the RADIUS server, the syslog method uses the system logging facility to record the CDRs, and the VSA method collects VSAs.
Examples
The following example enables the gateway to report user activity to the RADIUS server in the form of connection accounting records: aaa accounting connection start-stop group radius gatekeeper accounting
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A accounting (gatekeeper)
The following example shows how to enable VSA accounting: aaa accounting connection start-stop group radius gatekeeper accounting exec vsa
The following example configures H.323 accounting using IETF RADIUS attributes: Router(config-gk)# accounting username h323id
The following example configures H.323 accounting using VSA RADIUS attributes: Router(config-gk)# accounting vsa
Related Commands
Command
Description
aaa accounting
Enables AAA accounting of requested services for billing or security purposes.
gatekeeper
Enters gatekeeper configuration mode.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A accounting method
accounting method To set an accounting method at login for calls that come into a dial peer, use the accounting method command in voice class AAA configuration mode. To disable the accounting method set at login, use the no form of this command. accounting method MethListName [out-bound] no accounting method MethListName [out-bound]
Syntax Description
MethListName
Defines an accounting method list name.
out-bound
(Optional) Defines the outbound leg.
Command Default
When this command is not used to specify an accounting method, the system uses the aaa accounting connection h323 command as the default. If the method list name is not specified, the outbound call leg uses the same method list name as the inbound call leg
Command Modes
Voice class AAA configuration
Command History
Release
Modification
12.2(11)T
This command was introduced on the Cisco 3660, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
Usage Guidelines
This command sets the accounting method for dial peers in voice class AAA configuration mode. To initially define a method list, refer to the Cisco IOS Security Configuration Guide, Release 12.2. If the outbound option is specified, the outbound call leg on the dial peer uses the method list name specified in the command. If the method list name is not specified, by default, the outbound call leg uses the same method list name as the inbound call leg.
Examples
The following example sets the dp-out method for the outbound leg: voice class aaa 1 accounting method dp-out out-bound
Related Commands
Command
Description
aaa accounting connection h323 Defines the accounting method list H.323 with RADIUS, using stop-only or start-stop accounting options. voice class aaa
Cisco IOS Voice Command Reference
VR-12
Enables dial-peer-based VoIP AAA configurations.
Cisco IOS Voice Commands: A accounting suppress
accounting suppress To disable accounting that is automatically generated by a service provider module for a specific dial peer, use the accounting suppress command in voice class AAA configuration mode. To allow accounting to be automatically generated, use the no form of this command. accounting suppress [in-bound | out-bound] no accounting suppress [in-bound | out-bound]
Syntax Description
in-bound
(Optional) Defines the call leg for incoming calls.
out-bound
(Optional) Defines the call leg for outbound calls.
Command Default
Accounting is automatically generated by the service provider module.
Command Modes
Voice class AAA configuration
Command History
Release
Modification
12.2(11)T
This command was introduced on the Cisco 3660, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
Usage Guidelines
If a call leg option is not specified by the command, accounting is disabled for both inbound and outbound calls. For accounting to be automatically generated in the service provider module, you must first configure gw-accounting aaa command in global configuration mode before configuring dial-peer-based accounting in voice class AAA configuration mode.
Examples
In the example below, accounting is suppressed for the incoming call leg. voice class aaa 1 accounting suppress in-bound
Related Commands
Command
Description
gw-accounting aaa
Enables VoIP gateway accounting.
suppress
Turns off accounting for a call leg on a POTS or VoIP dial peer. This command is used in gw-accounting aaa configuration mode.
voice class aaa
Enables dial-peer-based VoIP AAA configurations.
Cisco IOS Voice Command Reference
VR-13
Cisco IOS Voice Commands: A accounting template
accounting template To allow each dial peer to choose and send a customized accounting template to the RADIUS server, use the accounting template command in voice class AAA configuration mode. To disable the dial peer from choosing and sending a customized accounting template, use the no form of this command. accounting template acctTempName [out-bound] no accounting template acctTempName [out-bound]
Syntax Description
acctTempName
Defines an accounting template name.
out-bound
(Optional) Defines the outbound leg.
Command Default
The dial peer does not choose and send a customized accounting template to the RADIUS server.
Command Modes
Voice class AAA configuration
Command History
Release
Modification
12.2(11)T
This command was introduced on the Cisco 3660, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
Usage Guidelines
By default, non-RFC-mandatory vendor-specific attributes (VSAs) are not included in accounting records if you do not configure the accounting template. The accounting template enables you to manage accounting records at a per-VSA level. When an accounting template is used for customizing the accounting record, the VSA name release source has to be included in the template file so that it is included in the accounting record and sent to the RADIUS server. This command overrides the acct-template command in gateway accounting AAA configuration mode when a customized accounting template is used. If you use a Tool Command Language (Tcl) script, the Tcl verb aaa accounting start [-t acctTempName] takes precedence over the accounting template command in voice class AAA configuration mode.
Examples
The following example sets the template temp-dp for the outbound leg voice class aaa 1 accounting template temp-dp out-bound
Related Commands
Command
Description
acct-template
Sends a selected group of voice accounting VSAs.
voice class aaa
Enables dial-peer-based VoIP AAA configurations.
Cisco IOS Voice Command Reference
VR-14
Cisco IOS Voice Commands: A acc-qos
acc-qos To define the acceptable quality of service (QoS) for any inbound and outbound call on a VoIP dial peer, use the acc-qos command in dial peer configuration mode. To restore the default QoS setting, use the no form of this command. acc-qos {best-effort | controlled-load | guaranteed-delay} [audio | video] no acc-qos
Syntax Description
best-effort
Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation. This is the default.
controlled-load
Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to ensure that preferential service is received even when the bandwidth is overloaded.
guaranteed-delay
Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded.
audio
(Optional) Configures acceptable QoS for audio traffic.
video
(Optional) Configures acceptable QoS for video traffic.
Command Default
best-effort
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
12.1(5)T
The description of the command was modified.
12.3(4)T
The audio and video keywords were added.
Usage Guidelines
This command is applicable only to VoIP dial peers. When VoIP dial peers are used, the Cisco IOS software uses RSVP to reserve a certain amount of bandwidth so that the selected QoS can be provided by the network. Call setup is aborted if the RSVP resource reservation does not satisfy the acceptable QoS for both peers. To select the most appropriate value for this command, you need to be familiar with the amount of traffic this connection supports and what kind of impact you are willing to have on it. The Cisco IOS software generates a trap message when the bandwidth required to provide the selected quality of service is not available. If audio or video is not configured, the bearer capability information element (IE) is not checked against max values during SETUP.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A acc-qos
You must use the ip rsvp bandwidth command to enable RSVP on an IP interface before you can specify RSVP QoS. In order to use this command, you have to have the req-qos statement present.
Examples
The following example selects guaranteed-delay as the acceptable QoS for inbound and outbound audio calls on VoIP dial peer 10: dial-peer voice 10 voip acc-qos guaranteed-delay
The following example selects controlled-load as the acceptable QoS for audio and video: dial-peer voice 100 voip acc-qos controlled-load audio acc-qos controlled-load video
Related Commands
Command
Description
req-qos
Requests a particular QoS using RSVP to be used in reaching a specified dial peer in VoIP.
Cisco IOS Voice Command Reference
VR-16
Cisco IOS Voice Commands: A acct-template
acct-template To select a group of voice attributes to collect in accounting records, use the acct-template command in gateway accounting AAA or gateway accounting file configuration mode. To disable collection of a group of voice attributes, use the no form of this command. acct-template {template-name | callhistory-detail} no acct-template {template-name | callhistory-detail}
Syntax Description
template-name
Name of the custom accounting template.
callhistory-detail
Collects all voice vendor-specific attributes (VSAs) for accounting.
This command was added to gateway accounting file configuration mode.
12.4(20)T
This command was integrated into Cisco IOS Release 12.4(20)T.
Usage Guidelines
Use this command to collect only the voice attributes that are defined in an accounting template. The accounting template is a text file that you create by selecting specific attributes that are applicable to your billing needs. Use the call accounting-template voice command to define your accounting template before using the acct-template command. The show call accounting-template voice master command displays all the voice attributes that can be filtered by accounting templates. Use the callhistory-detail keyword to send all voice VSAs to the accounting server. For a description of supported voice VSAs, see the “VSAs Supported by Cisco Voice Products” section in the RADIUS VSA Voice Implementation Guide. When you send only those VSAs defined in your accounting template, the default call-history records that are created by the service provider are automatically suppressed.
Examples
The example below uses the acct-template command to specify temp-global, a custom template. gw-accounting aaa acct-template temp-global
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A acct-template
Related Commands
Command
Description
call accounting-template voice
Defines a customized accounting template.
gw-accounting
Enables the method of collecting accounting data.
show call accounting-template voice Displays attributes defined in accounting templates.
Cisco IOS Voice Command Reference
VR-18
Cisco IOS Voice Commands: A activation-key
activation-key To define an activation key that can be dialed by phone users to activate Call Back on Busy on an analog phone, use the activation-key command in STC application feature callback configuration mode. To return the code to its default, use the no form of this command. activation-key string no activation-key
This command was integrated into Cisco IOS Release 12.4(22)T.
Usage Guidelines
Character string that can be dialed on a telephone keypad (0-9, *, #). Length of string is one to five characters. Default: #1.
This command changes the value of the callback activation key for Call Back on Busy from the default (#1) to the specified value. To display information about the Call Back configuration, use the show stcapp feature codes command.
Examples
The following example shows how to change the value of the callback activation key sequence from the default (#1) to a new value (*22). Router(config)# stcapp feature callback Router(config-stcapp-callback)# activation-key *22 Router(config-stcapp-callback)#
The following partial output from the show stcapp feature codes command displays values for the call back feature: Router# show stcapp feature codes . . . stcapp feature callback key *1 timeout 30
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A activation-key
Related Commands
Command
Description
ringing-timeout
Defines the timeout period for Callback on Busy.
show stcapp feature codes
Displays all feature codes for FACs, FSDs, and call back.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A address-family (tgrep)
address-family (tgrep) To set the global address family to be used on all dial peers, use the address-family command in TGREP configuration mode. To change back to the default address family, use the no form of this command. address family {e164 | decimal | penta-decimal} no address family {e164 | decimal | penta-decimal}
Syntax Description
e164
E.164 address family.
decimal
Digital address family.
penta-decimal
Pentadecimal address family.
Command Default
E.164 address family
Command Modes
TGREP configuration
Command History
Release
Modification
12.3(1)
This command was introduced.
Usage Guidelines
The E. 164 address family is used if the telephony network is a public telephony network. Decimal and pentadecimal options can be used to advertise private dial plans. For example, if a company wants to use TRIP in within its enterprise telephony network using five-digit extensions, then the gateway would advertise the beginning digits of the private numbers as a decimal address family. These calls cannot be sent out of the company’s private telephony network because they are not E.164-compliant. The pentadecimal family allows numbers 0 through 9 and alphabetic characters A through E and can be used in countries where letters are also carried in the called number.
Examples
The following example shows that the address family for itad 1234 is set for E.164 addresses: Router(config)# tgrep local-itad 1234 Router(config-tgrep)# address family e164
Related Commands
Command
Description
tgrep local-itad
Enters TGREP configuration mode and defines an ITAD.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A address-hiding
address-hiding To hide signaling and media peer addresses from endpoints other than the gateway use the address-hidding command in voice-service configuration mode. To allow the peer address known to all endpoints, use the no form of this command. address-hiding no address-hiding
Syntax Description
There are no keywords or aruguments.
Command Default
Signaling and media addresses are visable to all endpoints.
Command Modes
Voice-service configuration mode
Command History
Release
Modification
12.4(9)T
This command was introduced.
Usage Guidelines
Note
Examples
All SIP methods/messages should terminate at IP-to-IP gateway and re-originate with IP-to-IP gateway address, address hiding makes the peer address known only to the IP-to-IP gateway. Hiding address in flow-through mode is required for SIP-to-SIP in an IP-to-IP gateway network.
Distinctive ringing headers include ringing information and server address where the ring tone can be optained. These headers will be forwarded as is to the peer side even if address hiding is enabled.
The following example show address-hiding being configured for all VoIP calls: Router(config)# voice service voip Router(config-voi-serv) address-hiding
Related Commands
Command
Description
voice service
Enters voice-service configuration mode.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A advertise (annex g)
advertise (annex g) To control the types of descriptors that the border element (BE) advertises to its neighbors, use the advertise command in Annex G configuration mode. To reset this command to the default value, use the no form of this command. advertise [static | dynamic | all] no advertise
Syntax Description
static
(Optional) Only the descriptors provisioned on this BE is advertised. This is the default.
dynamic
(Optional) Only dynamically learned descriptors is advertised.
all
(Optional) Both static and dynamic descriptors is advertised.
Defaults
Static
Command Modes
Annex G configuration
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400 is not included in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850 universal gateway.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
Examples
The following example configures a BE that advertises both static and dynamic descriptors to its neighbors: Router(config)# call-router h323-annexg be20 Router(config-annexg)# advertise all
Related Commands
Command
Description
call-router
Enables the Annex G border element configuration commands.
show call history
Displays the routes stored in cache for the BE.
show call-router status Displays the Annex G BE status.
Cisco IOS Voice Command Reference
VR-23
Cisco IOS Voice Commands: A advertise (tgrep)
advertise (tgrep) To turn on reporting for a specified address family, use the advertise command in TGREP configuration mode. To turn off reporting for a specified address family, use the no form of this command. advertise {e164 | decimal | penta-decimal }[csr][ac][tc][trunk-group | carrier] advertise {trunk-group | carrier}[csr][ac][tc] no advertise {e164 | decimal | penta-decimal | trunk-group | carrier}
Syntax Description
e164
E.164 address family.
decimal
Decimal address family
penta-decimal
Penta-decimal address family
trunk-group
Trunk group address family
carrier
Carrier code address family
csr
Call success rate
ac
Available circuits
tc
Total circuits
Command Default
No attributes for address families are advertised.
Command Modes
TGREP configuration
Command History
Release
Modification
12.3(1)
This command was introduced.
Usage Guidelines
If you specify e164, decimal or penta-decimal for the address family, you can stipulate whether the related carrier or trunk-group parameters are advertised. If you stipulate carrier or trunk-group for the address family, you can stipulate that the related address family prefix is advertised. If you stipulate carrier or trunk-group for the address family, you cannot stipulate carrier or trunk-group attributes for advertising. When the no version of this command is used, it turns off the advertisement of that particular address family altogether.
Examples
The following example shows that the E.164 address family with call success rate, available circuits, total circuits, and trunk group attributes is being advertised for ITAD 1234: Router(config)# tgrep local-itad 1234 Router(config-tgrep)# advertise e164 csr ac tc trunk-group
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Cisco IOS Voice Commands: A advertise (tgrep)
Related Commands
Command
Description
tgrep local-itad
Enters TGREP configuration mode and defines an ITAD.
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Cisco IOS Voice Commands: A alarm-trigger
alarm-trigger To configure a T1 or E1 controller to send an alarm to the public switched telephone network (PSTN) or switch if specified T1 or E1 DS0 groups are out of service, use the alarm-trigger command in controller configuration mode. To configure a T1 or E1 controller not to send an alarm, use the no form of this command. alarm-trigger blue ds0-group-list no alarm-trigger
Syntax Description
blue
Specifies the alarm type to be sent is “blue,” also known as an Alarm Indication Signal (AIS).
ds0-group-list
Specifies the DS0 group or groups to be monitored for permanent trunk connection status or busyout status.
Command Default
No alarm is sent
Command Modes
Controller configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600, Cisco 3600, and Cisco MC3810.
Usage Guidelines
Any monitored time slot can be used for either permanent trunk connections or switched connections. Permanent virtual circuits (PVCs) and switched virtual circuits (SVCs) can be combined on a T1 or E1 controller and monitored for alarm conditioning. An alarm is sent only if all of the time slots configured for alarm conditioning on a T1 or E1 controller are out of service. If one monitored time slot remains in service or returns to service, no alarm is sent.
Examples
The following example configures T1 0 to send a blue (AIS) alarm if DS0 groups 0 and 1 are out of service: controller t1 0 alarm-trigger blue 0,1 exit
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Cisco IOS Voice Commands: A alarm-trigger
Related Commands
Command
Description
busyout monitor
Configures a voice port to monitor an interface for events that would trigger a voice-port busyout.
connection trunk
Creates a permanent trunk connection (private line or tie-line) between a voice port and a PBX.
voice class permanent
Creates a voice class for a Cisco or FRF-11 permanent trunk.
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Cisco IOS Voice Commands: A alias static
alias static To create a static entry in the local alias table, use the alias static command in gatekeeper configuration mode. To remove a static entry, use the no form of this command. alias static ip-signaling-addr [port] gkid gatekeeper-name [ras ip-ras-addr port] [terminal | mcu | gateway {h320 | h323-proxy | voip}] [e164 e164-address] [h323id h323-id] no alias static ip-signaling-addr [port] gkid gatekeeper-name [ras ip-ras-addr port] [terminal | mcu | gateway {h320 | h323-proxy | voip}] [e164 e164-address] [h323id h323-id]
Syntax Description
ip-signaling-addr
IP address of the H.323 node, used as the address to signal when establishing a call.
port
(Optional) Port number other than the endpoint Call Signaling well-known port number (1720).
gkid gatekeeper-name Name of the local gatekeeper of whose zone this node is a member. ras ip-ras-addr
(Optional) Node remote access server (RAS) signaling address. If omitted, the ip-signaling-addr parameter is used in conjunction with the RAS well-known port.
port
(Optional) Port number other than the RAS well-known port number (1719).
terminal
(Optional) Indicates that the alias refers to a terminal.
mcu
(Optional) Indicates that the alias refers to a multiple control unit (MCU).
gateway
(Optional) Indicates that the alias refers to a gateway.
h320
(Optional) Indicates that the alias refers to an H.320 node.
h323-proxy
(Optional) Indicates that the alias refers to an H.323 proxy.
voip
(Optional) Indicates that the alias refers to VoIP.
e164 e164-address
(Optional) Specifies the node E.164 address. This keyword and argument can be used more than once to specify as many E.164 addresses as needed. Note that there is a maximum number of 128 characters that can be entered for this address. To avoid exceeding this limit, you can enter multiple alias static commands with the same call signaling address and different aliases.
h323id h323-id
(Optional) Specifies the node H.323 alias. This keyword and argument can be used more than once to specify as many H.323 identification (ID) aliases as needed. Note that there is a maximum number of 256 characters that can be entered for this address. To avoid exceeding this limit, you can enter multiple alias static commands with the same call signaling address and different aliases.
Command Default
No static aliases exist.
Command Modes
Gatekeeper configuration
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Cisco IOS Voice Commands: A alias static
Command History
Usage Guidelines
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2500 series and Cisco 3600 series.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
The local alias table can be used to load static entries by performing as many of the commands as necessary. Aliases for the same IP address can be added in different commands, if required. Typically, static aliases are needed to access endpoints that do not belong to a zone (that is, they are not registered with any gatekeeper) or whose gatekeeper is inaccessible.
Examples
The following example creates a static terminal alias in the local zone: zone local gk.zone1.com zone1.com alias static 192.168.8.5 gkid gk.zone1.com terminal e164 14085551212 h323id terminal1
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Cisco IOS Voice Commands: A allow-connections
allow-connections To allow connections between specific types of endpoints in a VoIP network, use the allow-connections command in voice service configuration mode. To refuse specific types of connections, use the no form of this command. allow-connections from-type to to-type no allow-connections from-type to to-type
Syntax Description
Command Default
from-type
Originating endpoint type. The following choices are valid: •
h323—H.323.
•
sip—Session Interface Protocol (SIP).
to
Indicates that the argument that follows is the connection target.
to-type
Terminating endpoint type. The following choices are valid: •
H.323-to-H.323 connections are enabled by default and cannot be changed, and POTS-to-any and any-to-POTS connections are disabled. Cisco IOS Release 12.3(7)T and Later Releases
H.323-to-H.323 connections are disabled by default and can be changed, and POTS-to-any and any-to-POTS connections are enabled. H.323-to-SIP Connections
H.323-to-SIP and SIP-to-H.323 connections are disabled by default, and POTS-to-any and any-to-POTS connections are enabled. SIP-to-SIP Connections
SIP-to-SIP connections are disabled by default, and POTS-to-any and any-to-POTS connections are enabled.
Command Modes
Voice service configuration
Command History
Cisco IOS Release
Modification
12.2(13)T3
This command was introduced.
12.3(7)T
The default was changed.
12.3(11)T
The sip endpoint option was introduced for use with Cisco CallManager Express.
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Cisco IOS Voice Commands: A allow-connections
Usage Guidelines
Cisco IOS Release
Modification
12.2(13)T3
This command was introduced.
12.4(4)T
The sip endpoint option was implemented for use in IP-to-IP gateway networks.
12.2(33)SB
This command was integrated into Cisco IOS Release 12.2(33)SB.
12.4(22)T
Support for IPv6 was added.
Cisco IOS XE Release 2.5
This command was integrated into Cisco IOS XE Release 2.5.
This command is used to allow connections between specific types of endpoints in a Cisco multiservice IP-to-IP gateway. The command is enabled by default and cannot be changed. Connections to or from POTS endpoints are not allowed. Only H.323-to-H.323 connections are allowed. Cisco IOS Release 12.3(7)T and Later Releases
This command is used with Cisco Unified Communications Manager Express 3.1 or later systems and with the Cisco Multiservice IP-to-IP Gateway feature. In Cisco Unified Communications Manager Express, the allow-connections command enables the VoIP-to-VoIP connections used for hairpin call routing or routing to an H.450 tandem gateway.
Examples
The following example specifies that connections between H.323 and SIP endpoints are allowed: Router(config-voi-serv)# allow-connections h323 to sip
The following example specifies that connections between H.323 endpoints are allowed: Router(config-voi-serv)# allow-connections h323 to h323
The following example specifies that connections between SIP endpoints are allowed: Router(config-voi-serv)# allow-connections sip to sip
Related Commands
Command
Description
voice service
Enters voice service configuration mode.
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Cisco IOS Voice Commands: A allow subscribe
allow subscribe To allow internal watchers to monitor external presentities, use the allow subscribe command in presence configuration mode. To disable external watching, use the no form of this command. allow subscribe no allow subscribe
Syntax Description
This command has no arguments or keywords.
Command Default
Only internal presentities can be watched when presence is enabled.
Command Modes
Presence configuration (config-presence)
Command History
Release
Modification
12.4(11)XJ
This command was introduced.
12.4(15)T
This command was integrated into Cisco IOS Release 12.4(15)T.
Usage Guidelines
This command allows internal watchers to receive Busy Lamp Field (BLF) status notification for external directory numbers on a remote router connected through a SIP trunk. An external directory number must be enabled as a presentity with the allow watch command. The router sends SUBSCRIBE requests through the SIP trunk to an external presence server on behalf of the internal watcher and returns presence status to the watcher. To permit the external directory numbers to be watched, you must enable the watcher all command on the remote router.
Examples
The following example shows how to enable internal watchers to monitor external presentities: Router(config)# presence Router(config-presence)# allow subscribe
Related Commands
Command
Description
allow watch
Allows a line on a phone registered to Cisco Unified CME to be watched in a presence service.
blf-speed-dial
Enables BLF monitoring for a speed-dial number on a phone registered to Cisco Unified CME.
presence
Enables presence service on the router and enters presence configuration mode.
presence call-list
Enables BLF monitoring for call lists and directories on phones registered to Cisco Unified CME.
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Cisco IOS Voice Commands: A allow subscribe
Command
Description
presence enable
Allows incoming presence requests from SIP trunks.
server
Specifies the IP address of a presence server for sending presence requests from internal watchers to external presence entities.
show presence global
Displays configuration information about the presence service.
show presence subscription
Displays information about active presence subscriptions.
watcher all
Allows an external watcher to monitor an internal presentity.
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Cisco IOS Voice Commands: A alt-dial
alt-dial To configure an alternate dial-out string for dial peers, use the alt-dial command in dial peer configuration mode. To delete the alternate dial-out string, use the no form of this command. alt-dial string no alt-dial string
Syntax Description
string
Command Default
No alternate dial-out string is configured
Command Modes
Dial Peer configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810.
Usage Guidelines
The alternate dial-out string.
This command applies to plain old telephone service (POTS), Voice over Frame Relay (VoFR), and Voice ATM (VoATM) dial peers. The alt-dial command is used for the on-net-to-off-net alternative dialing function. The string replaces the destination-pattern string for dialing out.
Examples
The following example configures an alternate dial-out string of 95550188: alt-dial 95550188
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Cisco IOS Voice Commands: A anat
anat To enable Alternative Network Address Types (ANAT) on a Session Initiation Protocol (SIP) trunk, use the anat command in voice service voip-sip configuration mode or dial peer configuration mode. To ANAT on SIP trunks, use the no form of this command. anat no anat
Syntax Description
This command has no arguments or keywords.
Command Default
ANAT is enabled on SIP trunks.
Command Modes
Voice service voip-sip configuration Dial peer configuration
Command History
Release
Modification
12.4(22)T
This command was introduced.
Usage Guidelines
Both the Cisco IOS SIP gateway and the Cisco Unified Border Element are required to support Session Description Protocol (SDP) ANAT semantics for SIP IPv6 sessions. SDP ANAT semantics are intended to address scenarios that involve different network address families (for example, different IP versions). Media lines grouped using ANAT semantics provide alternative network addresses of different families for a single logical media stream. The entity creating a session description with an ANAT group must be ready to receive or send media over any of the grouped “m” lines. By default, ANAT is enabled on SIP trunks. However, if the SIP gateway is configured in IPv4-only or IPv6-only mode, the gateway will not use ANAT semantics in its SDP offer.
Examples
The following example enables ANAT on a SIP trunk: router(conf-serv-sip)# anat
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Cisco IOS Voice Commands: A ani mapping
ani mapping To preprogram the Numbering Plan Area (NPA), or area code, into a single Multi Frequency (MF) digit, use the ani mapping command in voice-port configuration mode. To disable Automatic Number Identification (ANI) mapping, use the no form of this command. ani mapping npd-value npa-number no ani mapping
Syntax Description
npd-value
Value of the Numbering Plan Digit (NPD). Range is 0 to 3. There is no default.
npa-number
Number (area code) of the NPA. Range is 100 to 999. There is no default value.
Command Default
No default behavior or values
Command Modes
Voice-port configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
Usage Guidelines
The ani mapping command table translates the NPA into a single MF digit. The number of NPDs programmed is determined by local policy as well as by the number of NPAs that the public service answering point (PSAP) serves. Repeat this command until all NPDs are configured or until the NPD maximum range is reached.
Examples
The following example shows the voice port preprogramming the NPA into a single MF digit: voice-port 1/1/0 timing digit 100 timing inter-digit 100 ani mapping 1 408 signal cama KP-NPD-NXX-XXXX-ST ! voice-port 1/1/1 timing digit 100 timing inter-digit 100 ani mapping 1 408 signal cama KP-NPD-NXX-XXXX-ST
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Cisco IOS Voice Commands: A ani mapping
Related Commands
Command
Description
signal
Specifies the type of signaling for a CAMA port.
voice–port
Enters voice-port configuration mode.
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Cisco IOS Voice Commands: A answer-address
answer-address To specify the full E.164 telephone number to be used to identify the dial peer of an incoming call, use the answer-address command in dial peer configuration mode. To disable the configured telephone number, use the no form of this command. answer-address [+]string[T] no answer-address
Syntax Description
+
(Optional) Character that indicates an E.164 standard number.
string
Series of digits that specify a pattern for the E.164 or private dialing plan telephone number. Valid entries are the digits 0 through 9, the letters A through D, and the following special characters: •
The asterisk (*) and pound sign (#) that appear on standard touch-tone dial pads.
•
Comma (,), which inserts a pause between digits.
•
Period (.), which matches any entered digit (this character is used as a wildcard).
•
Percent sign (%), which indicates that the preceding digit occurred zero or more times; similar to the wildcard usage.
•
Plus sign (+), which indicates that the preceding digit occurred one or more times.
Note
T
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The plus sign used as part of a digit string is different from the plus sign that can be used in front of a digit string to indicate that the string is an E.164 standard number.
•
Circumflex (^), which indicates a match to the beginning of the string.
•
Dollar sign ($), which matches the null string at the end of the input string.
•
Backslash symbol (\), which is followed by a single character, and matches that character. Can be used with a single character with no other significance (matching that character).
•
Question mark (?), which indicates that the preceding digit occurred zero or one time.
•
Brackets ( [ ] ), which indicate a range. A range is a sequence of characters enclosed in the brackets; only numeric characters from 0 to 9 are allowed in the range.
•
Parentheses ( ( ) ), which indicate a pattern and are the same as the regular expression rule.
(Optional) Control character that indicates that the destination-pattern value is a variable-length dial string. Using this control character enables the router to wait until all digits are received before routing the call.
Cisco IOS Voice Commands: A answer-address
Command Default
The default value is enabled with a null string
Command Modes
Dial peer voice configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on Cisco 3600 series routers.
Usage Guidelines
Use the answer-address command to identify the origin (or dial peer) of incoming calls from the IP network. Cisco IOS software identifies the dial peers of a call in one of two ways: by identifying either the interface through which the call is received or the telephone number configured with the answer-address command. In the absence of a configured telephone number, the peer associated with the interface is associated with the incoming call. For calls that come in from a plain old telephone service (POTS) interface, the answer-address command is not used to select an incoming dial peer. The incoming POTS dial peer is selected on the basis of the port configured for that dial peer. There are certain areas in the world (for example, certain European countries) where valid telephone numbers can vary in length. Use the optional control character T to indicate that a particular answer-address value is a variable-length dial string. In this case, the system does not match the dialed numbers until the interdigit timeout value has expired.
Note
Examples
Cisco IOS software does not check the validity of the E.164 telephone number; it accepts any series of digits as a valid number.
The following example shows the E.164 telephone number 555-0104 as the dial peer of an incoming call being configured: dial-peer voice 10 pots answer-address +5550104
Related Commands
Command
Description
destination-pattern
Specifies either the prefix or the full E.164 telephone number to be used for a dial peer.
port (dial peer)
Associates a dial peer with a specific port.
prefix
Specifies the prefix of the dialed digits for a dial peer.
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Cisco IOS Voice Commands: A application (dial peer)
application (dial peer) To enable a specific application on a dial peer, use the application command in dial peer configuration mode. To remove the application from the dial peer, use the no form of this command. application application-name [out-bound] no application application-name [out-bound]
Syntax Description
application-name
Name of the predefined application that you wish to enable on the dial peer. See the “Usage Guidelines” section for valid application names.
out-bound
(Optional) Outbound calls are handed off to the named application. This keyword is used for store-and-forward fax applications and VoiceXML applications.
Command Default
No default behavior or values
Command Modes
Dial peer voice configuration
Command History
Release
Modification
11.3(6)NA2
This command was introduced on the Cisco 2500 series, Cisco 3600 series, and Cisco AS5300.
12.0(5)T
The SGCPAPP application was supported initially on the Cisco AS5300.
12.0(7)XK
Support for the SGCPAPP application was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620).
12.1(2)T
The SGCPAPP application was integrated into Cisco IOS Release 12.1(2)T.
12.1(3)T
The MGCPAPP application was implemented on the Cisco AS5300.
12.1(3)XI
The out-bound keyword was added for store-and-forward fax on the Cisco AS5300.
12.1(5)T
The out-bound keyword was integrated into Cisco IOS Release 12.1(5)T, and the command was implemented on the Cisco AS5800.
12.2(2)T
This command was implemented on the Cisco 7200 series.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(2)XN
Support for enhanced MGCP voice gateway interoperability was added to Cisco CallManager Version 3.1 for the Cisco 2600 series, Cisco 3600 series, and Cisco VG200.
12.2(4)T
This command was implemented on the Cisco 1750.
12.2(4)XM
This command was implemented on the Cisco 1751.
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Cisco IOS Voice Commands: A application (dial peer)
Usage Guidelines
Note
Release
Modification
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the following platforms: The Cisco 3725 and Cisco 3745. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command was integrated into Cisco CallManager Version 3.2 and implemented on the Cisco 1760 and Cisco IAD2420 series routers. This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 in this release.
12.2(13)T
The application-name argument was removed from the no form of this command.
12.2(15)T
Malicious Caller Identification (MCID) was added as a valid application-name argument.
12.2(15)ZJ
The session application referred to by the default value of the application-name argument was updated to include support for Open Settlement Protocol (OSP), call transfer, and call forwarding. The version of the session application referred to by default in Cisco IOS Release 12.2(13)T and earlier releases was renamed default.c.old.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
12.3(14)T
This command is obsolete in Cisco IOS Release 12.3(14)T. For Cisco IOS Release 12.3(14)T and later releases, use the application command in global configuration mode to configure applications on a dial peer.
Use this command when configuring interactive voice response (IVR) or any of the IVR-related features to associate a predefined session application with an incoming POTS dial peer and an outgoing Multimedia Mail over IP (MMoIP) dial peer. Calls that use the incoming POTS dial peer and the outgoing MMoIP dial peer are handed off to the specified predefined session application.
In Cisco IOS Release 12.2(15)T and later releases, the application name default refers to the application new version of the default session application that supports OSP, call transfer, and call forwarding. The default session application in Cisco IOS Release 12.2(13)T and earlier releases has been renamed default.old.c and can still be configure for specific dial peers through the application command or globally configured for all inbound dial peers through the call application global command. For Media Gateway Control Protocol (MGCP) and Simple Gateway Control Protocol (SGCP) networks, enter the application name in uppercase characters. For example, for MGCP networks, you would enter MGCPAPP for the application-name argument. The application can be applied only to POTS dial peers. Note that SGCP dial peers do not use dial-peer hunting.
Note
In Cisco IOS Release 12.2, you cannot mix SGCP and non-SGCP endpoints in the same T1 controller, nor can you mix SGCP and non-SGCP endpoints in the same DS0 group.
Note
MGCP scripting is not supported on the Cisco 1750 router or on Cisco 7200 series routers.
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Cisco IOS Voice Commands: A application (dial peer)
For H.323 networks, the application is defined by a Tool Command Language/interactive voice response (Tcl/IVR) filename and location. Incoming calls that use POTS dial peers and outgoing calls that use MMoIP dial peers are handed off to this application. For Session Initiation Protocol (SIP) networks, use this command to associate a predefined session application. The default Tcl application (from the Cisco IOS image) for SIP is session and can be applied to both VoIP and POTS dial peers.
Examples
The following example defines an application and applies it to an outbound MMoIP dial peer for the fax on-ramp operation: call application voice fax_on_vfc_onramp http://santa/username/clid_4digits_npw_3.tcl dial-peer voice 3 mmoip application fax_on_vfc_onramp out-bound destination-pattern 57108.. session target mailto:[email protected]
The following example applies the MGCP application to a dial peer: dial-peer voice 1 pots application MGCPAPP
The following example applies a predefined application to an incoming POTS dial peer: dial-peer voice 100 pots application c4
The following example applies a predefined application to an outbound MMoIP dial peer for the on-ramp operation: dial-peer voice 3 mmoip application fax_on_vfc_onramp_ap out-bound destination-pattern 57108.. session target mailto:[email protected]
The following example applies the predefined SIP application to a dial peer: dial-peer voice 10 pots application session
For Cisco IOS Release 12.2(15)T, MCID was added as a valid application-name argument. The following is a sample configuration using the MCID application name: call application voice mcid http://santa/username/app_mcid_dtmf.2.0.0.28.tcl dial-peer voice 3 pots application mcid incoming called-number 222.... direct-inward-dial port 1:D
Related Commands
Command
Description
application
Enables a specific application on a dial peer.
call application voice
Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
mgcp
Starts the MGCP daemon.
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A application (global)
application (global) To enter application configuration mode to configure applications, use the application command in global configuration mode. application
Syntax Description
This command has no keywords or arguments.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.3(14)T
This command was introduced to replace the application command in dial peer configuration mode.
Usage Guidelines
Use this command to enter application configuration mode. Related commands are used in application configuration mode to configure standalone applications (services) and linkable functions (packages).
Examples
The following example shows how to enter application configuration mode and configure a debit card service: Enter application configuration mode to configure applications and services: Router(config)# application
Load the debit card script: Router(config-app)# service debitcard tftp://server-1/tftpboot/scripts/app_debitcard.2.0.2.8.tcl
Configure language parameters for the debit card service: Router(config-app-param)# paramspace english language en paramspace english index 1 paramspace english prefix en paramspace english location tftp://server-1/tftpboot/scripts/au/en/
Related Commands
Command
Description
call application voice
Defines the name of a voice application and specify the location of the Tcl or VoiceXML document to load for this application.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A arq reject-resource-low
arq reject-resource-low To configure the gatekeeper to send an Admission Reject (ARJ) message to the requesting gateway if destination resources are low, use the arq reject-resource-low command in gatekeeper configuration mode. To disable the gatekeeper from checking resources, use the no form of this command. arq reject-resource-low no arq reject-resource-low
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.3(1)
This command was introduced.
Examples
The following example shows that the gatekeeper is configured to send an ARJ message to the requesting gateway if destination resources are low: gatekeeper arq reject-resource-low
Related Commands
Command
Description
lrq reject-resource-low
Configures a gatekeeper to notify a sending gatekeeper on receipt of an LRQ message that no terminating endpoints are available.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A arq reject-unknown-prefix
arq reject-unknown-prefix To enable the gatekeeper to reject admission requests (ARQs) for zone prefixes that are not configured, use the arq reject-unknown-prefix command in gatekeeper configuration mode. To reenable the gatekeeper to accept and process all incoming ARQs, use the no form of this command. arq reject-unknown-prefix no arq reject-unknown-prefix
Syntax Description
This command has no arguments or keywords
Command Default
The gatekeeper accepts and processes all incoming ARQs.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
11.3(6)Q,
This command was introduced.
11.3(7)NA
This command was introduced.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
Usage Guidelines
Use the arq reject-unknown-prefix command to configure the gatekeeper to reject any incoming ARQs for a destination E.164 address that does not match any of the configured zone prefixes. When an endpoint or gateway initiates an H.323 call, it sends an ARQ to its gatekeeper. The gatekeeper uses the configured list of zone prefixes to determine where to direct the call. If the called address does not match any of the known zone prefixes, the gatekeeper attempts to hairpin the call out through a local gateway. If you do not want your gateway to do this, then use the arq reject-unknown-prefix command. (The term hairpin is used in telephony. It means to send a call back in the direction from which it came. For example, if a call cannot be routed over IP to a gateway that is closer to the target phone, the call is typically sent back out through the local zone, back the way it came.) This command is typically used to either restrict local gateway calls to a known set of prefixes or deliberately fail such calls so that an alternate choice on a gateway’s rotary dial peer is selected.
Examples
Consider a gatekeeper configured as follows: zone zone zone zone
Cisco IOS Voice Commands: A arq reject-unknown-prefix
In this example configuration, the gatekeeper manages a zone containing gateways to the 408 area code, and it knows about a peer gatekeeper that has gateways to the 415 area code. Using the zone prefix command, the gatekeeper is then configured with the appropriate prefixes so that calls to those area codes hop off in the optimal zone. If the arq request-unknown-prefix command is not configured, the gatekeeper handles calls in the following way: •
A call to the 408 area code is routed out through a local gateway.
•
A call to the 415 area code is routed to the gk415 zone, where it hops off on a local gateway.
•
A call to the 212 area code is routed to a local gateway in the gk408 zone.
If the arq reject-unknown-prefix command is configured, the gatekeeper handles calls in the following way:
Related Commands
•
A call to the 408 area code is routed out through a local gateway.
•
A call to the 415 area code is routed to the gk415 zone, where it hops off on a local gateway.
•
A call to the 212 area code is rejected because the destination address does not match any configured prefix.
Command
Description
zone prefix
Adds a prefix to the gatekeeper zone list.
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Cisco IOS Voice Commands: A as
as To define an application server for backhaul, use the as command in IUA configuration mode. To disable the backhaul ability from an application server, use the no form of this command. as as-name {localip1 [localip2]} [local-sctp-port] [fail-over-timer] [sctp-startup-rtx] [sctp-streams] [sctp-t1init] no as name
Syntax Description
as-name
Defines the protocol name (only ISDN is supported).
localip1
Defines the local IP address(es) for all the ASPs in a particular AS.
localip2
(Optional) Defines the local IP address(es) for all the ASPs in a particular application server .
local-sctp-port
(Optional) Defines a specific local Simple Control Transmission Protocol (SCTP) port rather than an ISDN Q.921 User Adaptation Layer (IUA) well-known port.
fail-over-timer
(Optional) Configures the failover timer for a particular application server .
sctp-startup-rtx
(Optional) Configures the SCTP maximum startup retransmission timer.
sctp-streams
(Optional) Configures the number of SCTP streams for a particular application server .
sctp-t1init
(Optional) Configures the SCTP T1 initiation timer.
Command Default
No application server is defined.
Command Modes
IUA configuration
Command History
Release
Modification
12.2(4)T
This command was introduced.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T and support was added for the Cisco AS5300 platform.
12.2(13)T1
This command was implemented on the Cisco AS5850.
12.2(15)T
This command was integrated into Cisco IOS Release 12.2(15)T and implemented on the Cisco 2420, Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series; Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 network access server (NAS) platforms.
Usage Guidelines
A maximum of two local IP addresses can be specified. (Note that SCTP has built-in support for multihomed machines.)
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Cisco IOS Voice Commands: A as
Note
All of the ASPs in an application server must be removed before an application server can be unconfigured. The default value of the SCTP streams is determined by the hardware that you have installed. The value of the failover timer is found in the show iua as all command output. The number of streams to assign to a given association is implementation dependent. During the initialization of the IUA association, you need to specify the total number of streams that can be used. Each D channel is associated with a specific stream within the association. With multiple trunk group support, every interface can potentially be a separate D channel. At startup, the IUA code checks for all the possible T1, E1, or T3 interfaces and sets the total number of inbound and outbound streams supported accordingly. In most cases, there is only a need for one association between the gateway (GW) and the Media Gateway Controller (MGC). For the rare case that you are configuring multiple AS associations to various MGCs, the overhead from the unused streams would have minimal impact. The NFAS D channels are configured for one or more interfaces, where each interface is assigned a unique stream ID. The total number of streams for the association needs to include an additional stream for the SCTP management messages. So during startup, the IUA code adds one to the total number of interfaces (streams) found. You have the option to manually configure the number of streams per association. In the backhaul scenario, if the number of D channel links is limited to one, allowing the number of streams to be configurable avoids the unnecessary allocation of streams in an association that is never used. For multiple associations between a GW and multiple MGCs, the configuration utility is useful in providing only the necessary number of streams per association. The overhead from the streams allocated but not used in the association is negligible. If the number of streams is manually configured through the CLI, the IUA code cannot distinguish between a startup event, which automatically sets the streams to the number of interfaces, or if the value is set manually during runtime. If you are configuring the number of SCTP streams manually, you must add one plus the number of interfaces using the sctp-streams keyword. Otherwise, IUA needs to always add one for the management stream, and the total number of streams increments by one after every reload. When you set the SCTP stream with the CLI, you cannot change the inbound and outbound stream support once the association is established with SCTP. The value takes effect when you first remove the IUA AS configuration and then configure it back as the same application server or a new one. The other option is to reload the router.
Examples
An application server and the application server process (ASP) should be configured first to allow a National ISDN-2 with Cisco extensions (NI2+) to be bound to this transport layer protocol. The application server is a logical representation of the SCTP local endpoint. The local endpoint can have more than one IP address but must use the same port number. The following is an example of an application server configuration on a gateway. The configuration shows that an application server named as5400-3 is configured to use two local IP addresses and a port number of 2577: Router(config-iua)# as as5400-3 10.1.2.34 10.1.2.35 2577
The following output shows that the application server (as1) is defined for backhaul: AS as1 10.21.0.2 9900
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Cisco IOS Voice Commands: A as
Related Commands
Command
Description
asp
Defines an ASP for backhaul.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A asp
asp To define an application server process (ASP) for backhaul, use the asp command in IUA configuration mode. To disable the ASP, use the no form of this command. asp asp-name as as-name {remoteip1 [remoteip2]} [remote-sctp-port] [ip-precedence [sctp-keepalives] [sctp-max-associations] [sctp-path-retransmissions] [sctp-t3-timeout] no asp asp-name
Syntax Description
asp-name
Names the current ASP.
as
The application server to which the ASP belongs.
as-name
Name of the application server to which the ASP belongs.
remoteip1
(Optional) Designates the remote IP address for this Simple Control Transmission Protocol (SCTP) association.
remoteip2
Designates the remote IP address for this SCTP association.
remote-sctp-port
Connects to a remote SCTP port rather than the IUA well-known port.
ip-precedence
(Optional) Sets IP Precedence bits for protocol data units (PDUs).
sctp-keepalives
•
IP precedence is expressed in the type of service (ToS) field of the show ip sctp association parameters output. The default type of service (ToS) value is 0.
•
Valid precedence values range from 0 to 7. You can also use the default IP precedence value for this address by choosing the default option.
(Optional) Modifies the keepalive behavior of an IP address in a particular ASP. •
sctp-max-associations
(Optional) Sets the SCTP maximum association retransmissions for a particular ASP. Valid values range from 2 to 20. The default is 5.
sctp-pathretransmissions
(Optional) Sets the SCTP path retransmissions for a particular ASP. Valid values range from 2 to 10. The default is 3.
sctp-t3-timeout
(Optional) Sets the SCTP T3 retransmission timeout for a particular ASP. The default value is 900 ms.
Command Default
No ASP is defined.
Command Modes
IUA configuration
Cisco IOS Voice Command Reference
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Valid keepalive interval values range from 1000 to 60000. The default value is 500 ms (see the show ip sctp association parameters output under heartbeats).
Cisco IOS Voice Commands: A asp
Command History
Usage Guidelines
Note
Release
Modification
12.2(4)T
This command was introduced.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and support was added for the Cisco AS5300.
12.2(11)T1
This command was implemented on the Cisco AS5850.
12.2(15)T
This command was integrated into Cisco IOS Release 12.2(15)T and implemented on the Cisco 2420, Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series; and Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 network access server (NAS) platforms.
This command establishes SCTP associations. There can be only a maximum of three ASPs configured per AS. IP precedence is expressed in the ToS field of show ip sctp association parameters output. The default ToS value is 0.
All of the ASPs in an application server must be removed before an application serever can be unconfigured. You can configure the precedence value in IUA in the range of 0 to 7 for a given IP address. Within IUA, the upper three bits representing the IP precedence in the ToS byte (used in the IP header) is set based on the user input before passing down the value to SCTP. In turn, SCTP passes the ToS byte value to IP. The default value is 0 for “normal” IP precedence handling. The asp-name argument specifies the name of this ASP. The ip-precedence keyword sets the precedence and ToS field. The remote-ip-address argument specifies the IP address of the remote end-point (the address of MGC, for example). The number argument can be any IP precedence bits in the range 1 to 255. The no form of the command results in precedence bits not being explicitly set by SCTP. In the case of a hot-standby Cisco PGW2200 pair, from the gateway (GW) perspective there is usually one ASP active and another in the INACTIVE state. The ASP_UP message is used to bring the ASP state on the GW to the INACTIVE state, followed by the ASPTM message, ASP_ACTIVE to ready the IUA link for data exchange. (Eventually the QPTM Establish Request message actually initiates the start of the D channel for the given interface.) In the event that the GW detects a failure on the active ASP, it can send a NTFY message to the standby ASP to request that it become active.
Examples
An ASP can be viewed as a local representation of an SCTP association because it specifies a remote endpoint that is in communication with an AS local endpoint. An ASP is defined for a given AS. For example, the following configuration defines a remote signaling controller asp-name at two IP addresses for AS as1. The remote SCTP port number is 2577: Router(config-iua)# as as1 10.4.8.69, 10.4.9.69 2477 Router(config-iua)# asp asp1 as as1 10.4.8.68 10.4.9.68 2577
Multiple ASPs can be defined for a single AS for the purpose of redundancy, but only one ASP can be active. The ASPs are inactive and only become active after fail-over. In the Cisco Media Gateway Controller (MGC) solution, a signaling controller is always the client that initiates the association with a gateway. During the initiation phase, you can request outbound and inbound stream numbers, but the gateway only allows a number that is at least one digit higher than the number of interfaces (T1/E1) allowed for the platform.
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Cisco IOS Voice Commands: A asp
The following example specifies the IP precedence level on the specified IP address. This example uses IP precedence level 7, which is the maximum level allowed: Router(config-iua)# asp asp1 as ip-precedence 10.1.2.345 7
The following example specifies the IP address to enable and disable keepalives: Router(config-iua)# asp asp1 as sctp-keepalive 10.1.2.34
The following example specifies the keepalive interval in milliseconds. In this example, the maximum value of 60000 ms is used: Router(config-iua)# asp asp1 as sctp-keepalive 10.10.10.10 60000
The following example specifies the IP address for the SCTP maximum association and the maximum association value. In this example, a maximum value of 20 is used: Router(config-iua)# asp asp1 as sctp-max-association 10.10.10.10 20
The following example specifies the IP address for the SCTP path retransmission and the maximum path retransmission value. In this example, a maximum value of 20 is used: Router(config-iua)# asp asp1 as sctp-path-retransmissions 10.10.10.10 10
The following example specifies the IP address for SCTP T3 timeout and specifies the T3 timeout value in milliseconds. In this example, the maximum value of 60000 is used: Router(config-iua)# asp asp1 as sctp-t3-timeout 10.10.10.10 60000
Related Commands
Command
Description
as
Defines an application server for backhaul.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A asserted-id
asserted-id To set the privacy level and enable either P-Asserted-Identity (PAI) or P-Preferred-Identity (PPI) privacy headers in outgoing SIP requests or response messages, use the asserted-id command in voice service voip-sip configuration mode or on a dial peer. To remove the privacy level of either PAI or PPI, use the no form of this command. asserted-id [pai | ppi] no asserted-id
Syntax Description
pai
(Optional) Enables PAI privacy headers in outgoing SIP requests or response messages.
ppi
(Optional) Enables PPI privacy headers in outgoing SIP requests or response messages.
Command Default
The command is disabled.
Command Modes
Voice service voip-sip configuration Dial-peer configuration
Command History
Release
Modification
12.4(15)T
This command was introduced.
Usage Guidelines
Enter SIP configuration mode from voice-service configuration mode, as shown in the example. If you configure the asserted-id pai command, the gateway builds a P-Asserted-Identity header into the common SIP stack. The asserted-id pai command has a priority over the Remote-Party-ID (RPID) header and removes this header from any outbound message, even if the router is configured to use the RPID header. If you configure the asserted-id ppi command, the gateway builds a P-Preferred-Identity header into the common SIP stack. The asserted-id ppi command has a priority over the Remote-Party-ID (RPID) header and removes this header from any outbound message, even if the router is configured to use the RPID header.
Examples
The following example shows how to set the P-Asserted Identity in the privacy header: router> enable router# configure terminal router(config)# voice service voip router(conf-voi-serv)# sip router(conf-serv-sip)# asserted-id pai
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Cisco IOS Voice Commands: A asserted-id
Related Commands
Command
Description
calling-info pstn-to-sip Specifies calling information treatment for PSTN-to-SIP calls. privacy
Cisco IOS Voice Command Reference
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Sets privacy in support of RFC 3323.
Cisco IOS Voice Commands: A associate application sccp
associate application sccp To associate Skinny Client Control Protocol (SCCP) to the digital signal processor (DSP) farm profile, use the associate application sccp command in DSP farm profile configuration mode. To remove the protocol, use the no form of this command. associate application sccp no associate application sccp
Syntax Description
This command has no arguments or keywords.
Command Default
No SCCP is associated with the DSP farm profile.
Command Modes
DSP farm profile configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.2(33)SB
This command was integrated into Cisco IOS Release 12.2(33)SB.
12.4(22)T
Support for IPv6 was added.
Usage Guidelines
This command enables SCCP as the application control protocol used to interface with Cisco Unified CallManager.
Examples
The following example associates SCCP to the DSP farm profile: Router(config-dspfarm-profile)# associate application sccp
Related Commands
Command
Description
codec (dspfarm-profile) Specifies the codecs supported by a DSP farm profile. description (dspfarm-profile)
Includes a specific description about the DSP farm profile.
dspfarm profile
Enters DSP farm profile configuration mode and defines a profile for DSP farm services.
maximum sessions (dspfarm-profile)
Specifies the maximum number of sessions that need to be supported by the profile.
shutdown (dspfarm-profile)
Allocates DSP farm resources and associates with the application.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A associate ccm
associate ccm To associate a Cisco Unified CallManager with a Cisco CallManager group and establish its priority within the group, use the associate ccm command in the SCCP Cisco CallManager configuration mode. To disassociate a Cisco Unified CallManager from a Cisco CallManager group, use the no form of this command. associate ccm identifier-number priority priority-number no associate ccm identifier-number priority priority-number
Syntax Description
identifier-number
Number that identifies the Cisco Unified CallManager. Range is 1 to 65535. There is no default value.
priority priority-number
Priority of the Cisco Unified CallManager within the Cisco CallManager group. Range is 1 to 4. There is no default value. The highest priority is 1.
Command Default
No default behavior or values
Command Modes
SCCP Cisco CallManager configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
Examples
The following example associates Cisco Unified CallManager 125 with Cisco CallManager group 999 and sets the priority of the Cisco Unified CallManager within the group to 2: Router(config)# sccp ccm group 999 Router(config-sccp-ccm)# associate ccm 125 priority 2
Related Commands
Command
Description
connect interval
Specifies the amount of time that a DSP farm profile waits before attempting to connect to a Cisco Unified CallManager when the current Cisco Unified CallManager fails to connect.
connect retries
Specifies the number of times that a DSP farm attempts to connect to a Cisco Unified CallManager when the current Cisco Unified CallManager connections fails.
sccp ccm group
Creates a Cisco CallManger group and enters SCCP Cisco CallManager configuration mode.
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Cisco IOS Voice Commands: A associate profile
associate profile To associate a digital signal processor (DSP) farm profile with a Cisco CallManager group, use the associate profile command in SCCP Cisco CallManager configuration mode. To disassociate a DSP farm profile from a Cisco Unified CallManager, use the no form of this command. associate profile profile-identifier register device-name no associate profile profile-identifier register device-name
Syntax Description
profile-identifier
Number that identifies the DSP farm profile. Range is 1 to 65535. There is no default value.
register device-name
User-specified device name in Cisco Unified CallManager. A maximum number of 15 characters can be entered for the device name.
Command Default
This command is not enabled.
Command Modes
SCCP Cisco CallManager configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.4(22)T
Support for IPv6 was added.
Usage Guidelines
Note
Examples
The device name must match the name configured in Cisco UnifiedCallManager; otherwise the profile is not registered to Cisco Unified CallManager.
Each profile can be associated to only one Cisco CallManager group.
The following example associates DSP farm profile abgz12345 to Cisco CallManager group 999: Router(config)# sccp ccm group 999 Router(conif-sccp-ccm)# associate profile 1 register abgz12345
Related Commands
Command
Description
bind interface
Binds an interface to a Cisco CallManager group.
dspfarm profile
Enters DSP farm profile configuration mode and defines a profile for DSP farm services.
sccp ccm group
Creates a Cisco CallManager group and enters SCCP Cisco CallManager configuration mode.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A associate registered-number
associate registered-number To associate the preloaded route and outbound proxy details with the registered number, use the associate registered-number command in voice service VoIP SIP configuration mode. To remove the association, use the no form of this command. associate registered-number number no associate registered-number
Syntax Description
number
Command Default
The preloaded route and outbound proxy details are not associated with the registered number by default.
Command Modes
Voice service VoIP SIP configuration (conf-serv-sip)
Command History
Release
Modification
15.1(2)T
This command was introduced.
Examples
Registered number. The number must be between 4 and 32.
The following example shows how to associate a registered number in the SIP configuration mode: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# associate registered-number 5
Related Commands
Command
Description
voice-class sip associate registered-number
Associates preloaded route and outbound proxy details with the registered number in the dial-peer configuration level.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A asymmetric payload
asymmetric payload To configure Session Initiation Protocol (SIP) asymmetric payload support, use the asymmetric payload command in SIP configuration mode. To disable asymmetric payload support, use the no form of this command. asymmetric payload {dtmf | dynamic-codecs | full | system} no asymmetric payload
Syntax Description
dtmf
(Optional) Specifies that the asymmetric payload support is dual-tone multi-frequency (DTMF) only.
dynamic-codecs
(Optional) Specifies that the asymmetric payload support is for dynamic codec payloads only.
full
(Optional) Specifies that the asymmetric payload support is for both DTMF and dynamic codec payloads.
system
(Optional) Specifies that the asymmetric payload uses the global value.
Command Default
This command is disabled.
Command Modes
Voice service SIP configuration (conf-serv-sip)
Command History
Release
Modification
12.4(15)T
This command was introduced.
Cisco IOS XE Release 3.1S
This command was integrated into Cisco IOS Release IOS XE 3.1.
Usage Guidelines
Enter SIP configuration mode from voice-service configuration mode, as shown in the example. For the Cisco UBE the SIP asymmetric payload-type is supported for audio/video codecs, DTMF, and NSE. Hence, dtmf and dynamic-codecs keywords are internally mapped to the full keyword to provide asymmetric payload-type support for audio/video codecs , DTMF, and NSE.
Examples
The following example shows how to set up a full asymmetric payload globally on a SIP network for both DTMF and dynamic codecs: Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# asymmetric payload full
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Cisco IOS Voice Commands: A asymmetric payload
Related Commands
Command
Description
sip
Enters SIP configuration mode from voice-service VoIP configuration mode.
voice-class sip asymmetric payload
Configures SIP asymmetric payload support on a dial peer.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A atm scramble-enable
atm scramble-enable To enable scrambling on E1 links, use the atm scramble-enable command in interface configuration mode. To disable scrambling, use the no form of this command. atm scramble-enable no atm scramble-enable
Syntax Description
This command has no arguments or keywords.
Command Default
By default, payload scrambling is set off
Command Modes
Interface configuration
Command History
Release
Modification
12.0(5)XK
This command was introduced for ATM interface configuration on the Cisco MC3810.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Usage Guidelines
Enable scrambling on E1 links only. On T1 links, the default binary 8-zero substitution (B8ZS) line encoding normally ensures sufficient reliability. Scrambling improves data reliability on E1 links by randomizing the ATM cell payload frames to avoid continuous nonvariable bit patterns and to improve the efficiency of the ATM cell delineation algorithms. The scrambling setting must match that of the far end.
Examples
The following example shows how to set the ATM0 E1 link to scramble payload: interface atm0 atm scramble-enable
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A atm video aesa
atm video aesa To set the unique ATM end-station address (AESA) for an ATM video interface that is using switched virtual circuit (SVC) mode, use the atm video aesa command in ATM interface configuration mode. To remove any configured address for the interface, use the no form of this command. atm video aesa [default | esi-address] no atm video aesa
Syntax Description
default
(Optional) Automatically creates a network service access point (NSAP) address for the interface, based on a prefix from the ATM switch (26 hexadecimal characters), the MAC address (12 hexadecimal characters) as the end station identifier (ESI), and a selector byte (two hexadecimal characters).
esi-address
(Optional) Defines the 12 hexadecimal characters used as the ESI. The ATM switch provides the prefix (26 hexadecimal characters), and the video selector byte provides the remaining two hexadecimal characters.
Command Default
default
Command Modes
ATM Interface configuration
Command History
Release
Modification
12.0(5)XK
This command was introduced.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Usage Guidelines
You cannot specify the ATM interface NSAP address in its entirety. The system creates either all of the address or part of it, depending on how you use this command.
Examples
The following example shows the ATM interface NSAP address set automatically: interface atm0 atm video aesa default
The following example shows the ATM interface NSAP address set to a specific ESI value: interface atm0/1 atm video aesa 444444444444
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Cisco IOS Voice Commands: A atm video aesa
Related Commands
Command
Description
show atm video-voice address
Displays the NSAP address for the ATM interface.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A attribute acct-session-id overloaded
attribute acct-session-id overloaded To overload the acct-session-id attribute with call detail records, use the attribute acct-session-id overloaded command in gateway accounting AAA configuration mode. To disable overloading the acct-session-id attribute with call detail records, use the no form of this command. attribute acct-session-id overloaded no attribute acct-session-id overloaded
Syntax Description
This command has no arguments or keywords.
Command Default
The acct-session-id attribute is not overloaded with call detail records.
Command Modes
Gateway accounting AAA configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
Usage Guidelines
Examples
•
The attribute acct-session-id overloaded command replaces the gw-accounting h323 command.
•
The acct-session-id attribute is RADIUS attribute 44. For more information on this attribute, see the document RADIUS Attribute 44 (Accounting Session ID) in Access Requests.
•
Attributes that cannot be mapped to standard RADIUS attributes are packed into the acct-session-id attribute field as ASCII strings separated by the forward slash (“/”) character.
•
The Accounting Session ID (acct-session-id) attribute contains the RADIUS account session ID, which is a unique identifier that links accounting records associated with the same login session for a user. This unique identifier makes it easy to match start and stop records in a log file.
•
Accounting Session ID numbers restart at 1 each time the router is power-cycled or the software is reloaded.
The following example shows the acct-session-id attribute being overloaded with call detail records: gw-accounting aaa attribute acct-session-id overloaded
Related Commands
Command
Description
call accounting-template voice
Defines and loads the template file at the location defined by the URL.
gw-accounting aaa
Enables VoIP gateway accounting.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: A attribute h323-remote-id resolved
attribute h323-remote-id resolved To resolve the h323-remote-id attribute, use the attribute h323-remote-id resolved command in gateway accounting AAA configuration mode. To keep the h323-remote-id attribute unresolved, use the no form of this command. attribute h323-remote-id resolved no attribute h323-remote-id resolved
Syntax Description
This command has no arguments or keywords.
Command Default
The h323-remote-id attribute is not resolved.
Command Modes
gateway accounting aaa configuration
Command History
Release
Modification
12.2(11)T
This command was introduced on the Cisco 3660, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
Usage Guidelines
In Cisco IOS Release 12.2(11)T, the attribute h323-remote-id resolved command replaces the gw-accounting h323 resolve command, and the h323-remote-id attribute has been added as a Cisco vendor-specific attribute (VSA). This attribute is a string that indicates the Domain Name System (DNS) name or locally defined host name of the remote gateway. You can obtain the value of the h323-remote-id attribute by doing a DNS lookup of the h323-remote-address attribute. The h323-remote-address attribute indicates the IP address of the remote gateway.
Examples
The following example sets the h323-remote-id attribute to resolved: gw-accounting aaa attribute h323-remote-id resolved
Related Commands
Command
Description
gw-accounting aaa
Enables VoIP gateway accounting.
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Cisco IOS Voice Commands: A audio
audio To enable the incoming and outgoing IP-IP call gain/loss feature for audio volume control on the incoming dial peer and the outgoing dial peer, enter the audio command in dial-peer configuration mode. To disable this feature, use the no form of this command. audio {incoming | outgoing} level adjustment value no audio {incoming | outgoing} level adjustment value
Syntax Description
incoming
Enables the incoming IP-IP call volume control on either the incoming dial peer or the outgoing dial peer.
outgoing
Enables the outgoing IP-IP call volume control on either the incoming dial peer or the outgoing dial peer.
value
Range is -27 to 16.
Command Default
This command is disabled by default, and there is no volume control available.
Command Modes
Dial-peer configuration (config-dialpeer)
Command History
Release
Modification
15.0(1)M
This command was introduced.
Usage Guidelines
This feature enables the adjustment of the audio volume within a Cisco Unified Border Element (Cisco UBE) call. As with codec repacketization, dissimilar networks that have different built-in loss/gain characteristics may experience connectivity problems. By adding the ability to control the loss/gain within the Cisco UBE, you can more easily connect your networks. The DSP requires one level for each stream, so the value for audio incoming level-adjustment and the value for audio outgoing level-adjustment will be added together. If the combined values are outside of the limit the DSP can perform, the value sent to the DSP will be either the minimum (-27) or maximum (+16) supported by the DSP.
Caution
For gain/loss control, be aware that adding gain in a network with echo can generate feedback loud enough to cause hearing damage. Always exercise extreme caution when configuring gain into your network. To configure IP-IP Call Gain/Loss Control on a voice gateway, you must configure the incoming and outgoing VoIP dial peers.
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Cisco IOS Voice Commands: A audio
Examples
The following example shows how to configure audio incoming level to 5 and the audio outgoing level to -5: Router(config-dialpeer)# audio incoming level-adjustment 5 Router(config-dialpeer)# audio outgoing level-adjustment -5
Related Commands
Command
Description
show dial peer voice
Displays the codec setting for dial peers.
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Cisco IOS Voice Commands: A audio-prompt load
audio-prompt load To initiate loading the selected audio file (.au), which contains the announcement prompt for the caller, from Flash memory into RAM, use the audio-prompt load command in privileged EXEC mode. This command does not have a no form. audio-prompt load name
Syntax Description
name
Command Default
No default behavior or values
Command Modes
Privileged EXEC
Command History
Release
Modification
11.3(6)NA2
This command was introduced.
Location of the audio file that you want to have loaded from memory, flash memory, an FTP server, an HTTP server, or an HTTPS (HTTP over Secure Socket Layer (SSL)) server.
Note
Usage Guidelines
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1750 and Cisco 1751. Support for other Cisco platforms is not included in this release.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T. This command is supported on the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 in this release.
12.4(15)T
The name argument was modified to accept an HTTPS server URL.
The first time the interactive voice response (IVR) application plays a prompt, it reads it from the URL (or the specified location for the .au file, such as Flash or FTP) into RAM. Then it plays the script from RAM. An example of the sequence of events follows: •
When the first caller is asked to enter the account and personal identification numbers (PINs), the enter_account.au and enter_pin.au files are loaded into RAM from Flash memory.
•
When the next call comes in, these prompts are played from the RAM copy.
•
If all callers enter valid account numbers and PINs, the auth_failed.au file is not loaded from Flash memory into RAM.
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With Cisco IOS Release 11.3(6)NA2, the URL pointer refers to the directory where Flash memory is stored.
Cisco IOS Voice Commands: A audio-prompt load
The router loads the audio file only when the script initially plays that prompt after the router restarts. If the audio file is changed, you must run this privileged EXEC command to reread the file. This generates an error message if the file is not accessible or if there is a format error.
Examples
The following example shows how to load the enter_pin.au audio file from Flash memory into RAM: audio-prompt load flash:enter_pin.au
The following example shows how to load the hello.au audio file from an HTTPS server into RAM: audio-prompt load https://http-server1/audio/hello.au
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Cisco IOS Voice Commands: A authenticate redirecting-number
authenticate redirecting-number To enable a Cisco IOS voice gateway to authenticate and pass Session Initiation Protocol (SIP) credentials based on the redirecting number when available instead of the calling number of a forwarded call, use the authenticate redirecting-number command in voice service SIP configuration mode. To return a Cisco IOS voice gateway to the default setting so that the gateway uses only the calling number for SIP credentials, use the no form of this command. authenticate redirecting-number no authenticate redirecting-number
Syntax Description
This command has no arguments or keywords.
Command Default
The Cisco IOS voice gateway uses only the calling number of a forwarded call for SIP credentials even when the redirecting number information is available for that call.
Command Modes
Voice service SIP configuration (conf-serv-sip)
Command History
Release
Modification
12.4(24)T
This command was introduced.
Usage Guidelines
When an INVITE message sent out by the gateway is challenged, it must respond with the appropriate SIP credentials before the call is established. The default global behavior for the gateway is to authenticate and pass SIP credentials based on the calling number and all dial peers on a gateway default to the global setting. However, for forwarded calls, it is sometimes more appropriate to use the redirecting number and this can be specified at either the global or dial peer level (configuring behavior for a specific dial peer supersedes the global setting). Use the authenticate redirecting-number command in voice service SIP configuration mode to globally enable a Cisco IOS voice gateway to authenticate and pass SIP credentials based on the redirecting number when available. Use the no form of this command to configure the gateway to authenticate and pass SIP credentials based only on the calling number of forwarded calls unless otherwise configured at the dial peer level: •
Use the voice-class sip authenticate redirecting-number command in dial peer voice configuration mode to supersede global settings and force a specific dial peer on the gateway to authenticate and pass SIP credentials based on the redirecting number when available.
•
Use the no form of the voice-class sip authenticate redirecting-number command in dial peer voice configuration mode to supersede global settings and force a specific dial peer on the gateway to authenticate and pass SIP credentials based only on the calling number regardless of the global setting.
The redirecting number is present only in the headers of forwarded calls. When this command is disabled or the redirecting number is not available (nonforwarded calls), the gateway uses the calling number for SIP credentials.
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Cisco IOS Voice Commands: A authenticate redirecting-number
Examples
The following example shows how to globally enable a Cisco IOS voice gateway to authenticate and pass the redirecting number of a forwarded call when a SIP INVITE message is challenged: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# authenticate redirecting-number
Related Commands
Command
Description
voice-class sip authenticate Supersedes global settings and enables a dial peer on a Cisco IOS voice redirecting-number gateway to authenticate and pass SIP credentials based on the redirecting number of forwarded calls.
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Cisco IOS Voice Commands: A authentication (dial peer)
authentication (dial peer) To enable SIP digest authentication on an individual dial peer, use the authentication command in dial peer voice configuration mode. To disable SIP digest authentication, use the no form of this command. authentication username username password [0 | 7] password [realm realm [challenge]] no authentication username username [password [0 | 7] password [realm realm [challenge]]]
Syntax Description
username
Specifies the username for the user who is providing authentication.
username
A string representing the username for the user who is providing authentication. A username must be at least four characters.
password
Specifies password settings for authentication.
0
(Optional) Specifies encryption type as cleartext (no encryption). This is the default.
7
(Optional) Specifies encryption type as encrypted.
password
A string representing the password for authentication. If no encryption type is specified, the password will be cleartext format. The string must be between 4 and 128 characters.
realm
(Optional) Specifies the domain where the credentials are applicable.
realm
(Optional) A string representing the domain where the credentials are applicable.
Command Default
SIP digest authentication is disabled.
Command Modes
Dial peer voice configuration (config-dial-peer)
Command History
Release
Modification
12.3(8)T
This command was introduced.
15.1(3)T
This command was modified. The challenge keyword was added.
Usage Guidelines
The following configuration rules are applicable when enabling digest authentication: •
Only one username can be configured per dial peer. Any existing username configuration must be removed before configuring a different username.
•
A maximum of five password or realm arguments can be configured for any one username. The username and password arguments are used to authenticate a user. An authenticating server/proxy issuing a 407/401 challenge response includes a realm in the challenge response and the user provides credentials that are valid for that realm. Because it is assumed that a maximum of five proxy servers in the signaling path can try to authenticate a given request from a user-agent client (UAC) to a user-agent server (UAS), a user can configure up to five password and realm combinations for a configured username.
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Cisco IOS Voice Commands: A authentication (dial peer)
Note
The user provides the password in plain text but it is encrypted and saved for 401 challenge response. If the password is not saved in encrypted form, a junk password is sent and the authentication fails.
•
The realm specification is optional. If omitted, the password configured for that username applies to all realms that attempt to authenticate.
•
Only one password can be configured at a time for all configured realms. If a new password is configured, it overwrites any previously configured password. This means that only one global password (one without a specified realm) can be configured. If you configure a new password without configuring a corresponding realm, the new password overwrites the previous one.
Examples
•
If a realm is configured for a previously configured username and password, that realm specification is added to that existing username and password configuration. However, once a realm is added to a username and password configuration, that username and password combination is valid only for that realm. A configured realm cannot be removed from a username and password configuration without first removing the entire configuration for that username and password—you can then reconfigure that username and password combination with or without a different realm.
•
In an entry with both a password and realm, you can change either the password or realm.
The following example shows how to enable the digest authentication: Router> enable Router# configure terminal Router(config)# dial-peer voice 1 pots Router(config-dial-peer)# authentication username MyUser password 7 MyPassword realm MyRealm.example.com
The following example shows how to remove a previously configured digest authentication: Router> enable Router# configure terminal Router(config)# dial-peer voice 1 pots Router(config-dial-peer)# no authentication username MyUser password MyPassword
Related Commands
Command
Description
authentication (SIP UA)
Enables SIP digest authentication globally.
credentials (SIP UA)
Configures a Cisco UBE to send a SIP registration message when in the UP state.
localhost
Configures global settings for substituting a DNS local hostname in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages.
registrar
Enables Cisco IOS SIP gateways to register E.164 numbers on behalf of FXS, EFXS, and SCCP phones with an external SIP proxy or SIP registrar.
voice-class sip localhost
Configures settings for substituting a DNS local hostname in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages on an individual dial peer, overriding the global setting.
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Cisco IOS Voice Commands: A authentication (SIP UA)
authentication (SIP UA) To enable SIP digest authentication, use the authentication command in SIP UA configuration mode. To disable SIP digest authentication, use the no form of this command. authentication username username password [0 | 7] password [realm realm] no authentication username username [password [0 | 7] password [realm realm]]
Syntax Description
username username
A string representing the username for the user who is providing authentication (must be at least four characters).
password
Specifies password settings for authentication.
0
(Optional) Specifies encryption type as cleartext (no encryption). This is the default.
7
(Optional) Specifies encryption type as encrypted.
password
A string representing the password for authentication. If no encryption type is specified, the password will be cleartext format. The string must be between 4 and 128 characters.
realm realm
(Optional) A string representing the domain where the credentials are applicable.
Command Default
SIP digest authentication is disabled.
Command Modes
SIP UA configuration (config-sip-ua)
Command History
Release
Modification
12.3(8)T
This command was introduced.
Usage Guidelines
The following configuration rules are applicable when enabling digest access authentication: •
Only one username can be configured globally in SIP UA configuration mode. Any existing username configuration must be removed before configuring a different username.
•
A maximum of five password or realm arguments are allowed for a given username argument. The username and password arguments are used to authenticate a user. An authenticating server/proxy issuing a 407/401 challenge response includes a realm in the challenge response and the user provides credentials that are valid for that realm. Because it is assumed that a maximum of five proxy servers in the signaling path can try to authenticate a given request from a user-agent client (UAC) to a user-agent server (UAS), a user can configure up to five password and realm combinations for a configured username.
•
The realm specification is optional. If omitted, the password configured for that username applies to all realms that attempt to authenticate.
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Cisco IOS Voice Commands: A authentication (SIP UA)
•
Only one password can be configured at a time for all configured realms. If a new password is configured, it overwrites any previously configured password. This means that only one global password (one without a specified realm) can be configured. If you configure a new password without configuring a corresponding realm, the new password overwrites the previous one.
Examples
•
If a realm is configured for a previously configured username and password, that realm specification is added to that existing username and password configuration. However, once a realm is added to a username and password configuration, that username and password combination is valid only for that realm. A configured realm cannot be removed from a username and password configuration without first removing the entire configuration for that username and password—you can then reconfigure that username and password combination with or without a different realm.
•
In an entry with both a password and realm, you can change either the password or realm.
The following example shows how to enable digest access authentication: Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# authentication username MyUser password MyPassword realm example.com
The following example shows how to remove a previously configured digest access authentication: Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# no authentication username MyUser password MyPassword realm example.com
Related Commands
Command
Description
authentication (dial peer)
Enables SIP digest authentication on an individual dial peer.
credentials (SIP UA)
Configures a Cisco UBE to send a SIP registration message when in the UP state.
localhost
Configures global settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages.
registrar
Enables Cisco IOS SIP gateways to register E.164 numbers on behalf of FXS, EFXS, and SCCP phones with an external SIP proxy or SIP registrar.
voice-class sip localhost
Configures settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages on an individual dial peer, overriding the global setting.
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Cisco IOS Voice Commands: A authentication method
authentication method To set an authentication method at login for calls that come into a dial peer, use the authentication method command in voice class AAA configuration mode. To disable the authentication method set at login, use the no form of this command. authentication method MethListName no authentication method MethListName
Syntax Description
MethListName
Command Default
When this command is not used to specify a login authentication method, the system uses the aaa authentication login h323 command as the default.
Command Modes
Voice class AAA configuration
Command History
Release
Modification
12.2(11)T
This command was introduced on the Cisco 3660, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
Usage Guidelines
Authentication method list name.
This command is used to direct authentication requests to a RADIUS server based on dialed number information service (DNIS) or trunk grouping. This command is used for directing dial-peer-based authentication requests. The method list must be defined during initial authentication setup.
Examples
In the example below, “dp” is the method list name used for authentication. The method list name is defined during initial authentication setup. voice class aaa 1 authentication method dp
Related Commands
Command
Description
aaa authentication login Sets AAA authentication at login. voice class aaa
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Enables dial-peer-based VoIP AAA configurations.
Cisco IOS Voice Commands: A authorization method
authorization method To set an authorization method at login for calls that are into a dial peer, use the authorization method command in voice class AAA configuration mode. To disable the authorization method set at login, use the no form of this command. authorization method MethListName no authorization method MethListName
Syntax Description
MethListName
Command Default
When this command is not used to specifiy a login authorization method, the system uses the aaa authorization exec h323 command as the default.
Command Modes
Voice class AAA configuration
Command History
Release
Modification
12.2(11)T
This command was introduced on the Cisco 3660, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
Usage Guidelines
Defines an authorization method list name.
This command is used to direct authentication requests to a RADIUS server based on dialed number information service (DNIS) or trunk grouping. This command is used for directing dial-peer-based authentication requests. The method list must be defined during initial authentication setup.
Examples
The following example set an authorization method of “dp”: voice class aaa 1 authorization method dp
Related Commands
Command
Description
aaa authorization exec Runs authorization to determine if the user is allowed to run an EXEC shell. voice class aaa
Enables dial-peer-based VoIP AAA configurations.
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Cisco IOS Voice Commands: A auto-config
auto-config To enable auto-configuration or to enter auto-config application configuration mode for the Skinny Client Control Protocol (SCCP) application, use the auto-config command in global configuration mode. To disable auto-configuration, use the no form of this command. auto-config [application sccp] no auto-config
Syntax Description
application sccp
Command Default
Auto-configuration is disabled.
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)XY
This command was introduced on the Communication Media Module for the SCCP application.
12.3(14)T
This command was integrated into Cisco IOS Release 12.3(14)T.
Examples
(Optional) Enters auto-config application configuration mode for the SCCP application.
The following example shows the auto-config command used to enter auto-configuration application configuration mode for the SCCP application and the no shutdown command used to enable the SCCP application for download: Router(config)# auto-config application sccp Router(auto-config-app)# no shutdown
Related Commands
Command
Description
shutdown (auto-config application)
Disables an auto-configuration application for download.
show auto-config
Displays the current status of auto-configuration applications.
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Cisco IOS Voice Commands: A auto-cut-through
auto-cut-through To enable call completion when a PBX does not provide an M-lead response, use the auto-cut-through command in voice-port configuration mode. To disable the auto-cut-through operation, use the no form of this command. auto-cut-through no auto-cut-through
Syntax Description
This command has no arguments or keywords.
Command Default
Auto-cut-through is enabled.
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810.
12.0(7)XK
This command was first supported on the Cisco 2600 and Cisco 3600 series.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
The auto-cut-through command applies to ear and mouth (E&M) voice ports only.
Examples
The following example shows enabling of call completion on a router when a PBX does not provide an M-lead response: voice-port 1/0/0 auto-cut-through
Related Commands
Command
Description
show voice port
Displays voice port configuration information.
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Cisco IOS Voice Commands: A auto-cut-through
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Cisco IOS Voice Commands: B This chapter contains commands to configure and maintain Cisco IOS voice applications. The commands are presented in alphabetical order. Some commands required for configuring voice may be found in other Cisco IOS command references. Use the command reference master index or search online to find these commands. For detailed information on how to configure these applications and features, refer to the Cisco IOS Voice Configuration Library.
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Cisco IOS Voice Commands: B backhaul-session-manager
backhaul-session-manager To enter backhaul session manager configuration mode, use the backhaul-session-manager command in global configuration mode. backhaul-session-manager
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(2)T
This command was implemented on the Cisco 7200.
12.2(4)T
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.2(2)XB
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850 platform.
12.2(8)T
This command was implemented on Cisco IAD2420. Support for the Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Usage Guidelines
Use the backhaul-session-manager command to switch to backhaul session manager configuration mode from global configuration mode. Use the exit command to exit backhaul session manager configuration mode and return to global configuration mode.
Examples
The following example enters backhaul session manager configuration mode: Router(config)# backhaul-session-manager Router(config-bsm)#
Related Commands
Command
Description
clear backhaul-session-manager group
Resets the statistics or traffic counters for a specified session group.
clear rudpv1 statistics
Clears the RUDP statistics and failure counters.
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Cisco IOS Voice Commands: B backhaul-session-manager
Command
Description
group
Creates a session group and associates it with a specified session set.
group auto-reset
Configures the maximum auto-reset value.
group cumulative-ack
Configures maximum cumulative acknowledgments.
group out-of-sequence
Configures maximum out-of-sequence segments that are received before an EACK is sent.
group receive
Configures maximum receive segments.
group retransmit
Configures maximum retransmits.
group timer cumulative-ack
Configures cumulative acknowledgment timeout.
group timer keepalive
Configures keepalive (or null segment) timeout.
group timer retransmit
Configures retransmission timeout.
group timer transfer
Configures state transfer timeout.
isdn bind-l3
Configures the ISDN serial interface for backhaul.
session group
Associates a transport session with a specified session group.
set
Creates a fault-tolerant or non-fault-tolerant session set with the client or server option.
show backhaul-session-manager group
Displays status, statistics, or configuration of a specified or all session groups.
show backhaul-session-manager session
Displays status, statistics, or configuration of sessions.
show backhaul-session-manager set
Displays session groups associated with a specific or all session sets.
show rudpv1
Displays RUDP statistics.
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Cisco IOS Voice Commands: B bandwidth (dial peer)
bandwidth (dial peer) To set the maximum bandwidth on a POTS dial peer for an H.320 call, use the bandwidth command in dial peer configuration mode. To remove the bandwidth setting, use the no form of this command. bandwidth maximum value [maximum value] no bandwidth
Syntax Description
maximum value
Sets the maximum bandwidth for an H.320 call on a POTS dial peer. The range is 64 to 1024, entered in increments of 64 kilobits per second (kbps). The default is 64.
minimum value
(Optional) Sets the minimum bandwidth. Acceptable values are 64 kbps or minimum value=maximum value.
Command Default
No maximum bandwidth is set.
Command Modes
Dial peer configuration
Command History
Release
Modification
12.4(11)T
This command was introduced.
Usage Guidelines
Use this command to set the maximum and minimum bandwidth for an H.320 POTS dial-peer. Only the maximum bandwidth is required. The value must be entered in increments of 64 kbps. The minimum bandwidth setting is optional, and the value must be either 64 kbps or equal to the maximum value setting.
Examples
The following example shows configuration for POTS dial peer 200 with a maximum bandwidth of 1024 kbps: dial-peer voice 200 pots bandwidth maximum 1024
The following example shows configuration for POTS dial peer 11 with a maximum bandwidth of 640 and a minimum of 64: dial-peer voice 11 pots bandwidth maximum 640 minimum 64
Related Commands
Command
Description
bandwidth
Specifies the maximum aggregate bandwidth for H.323 traffic and verifies the available bandwidth of the destination gatekeeper.
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Cisco IOS Voice Commands: B bandwidth
bandwidth To specify the maximum aggregate bandwidth for H.323 traffic and verify the available bandwidth of the destination gatekeeper, use the bandwidth command in gatekeeper configuration mode. To disable maximum aggregate bandwidth, use the no form of this command. bandwidth {interzone | total | session} {default | zone zone-name} bandwidth-size no bandwidth {interzone | total | session} {default | zone zone-name}
Syntax Description
interzone
Total amount of bandwidth for H.323 traffic from the zone to any other zone.
total
Total amount of bandwidth for H.323 traffic allowed in the zone.
session
Maximum bandwidth allowed for a session in the zone.
default
Default value for all zones.
zone
A particular zone.
zone-name
Name of the particular zone.
bandwidth-size
Maximum bandwidth, in kbps. For interzone and total, range : 1 to 10000000. For session, range:1 to 5000.
Command Default
Maximum aggregate bandwidth is unlimited by default.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2500, Cisco 3600 series and the Cisco AS5300.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T. The bandwidth command replaced the zone bw command.
12.1(5)XM
The bandwidth command was recognized without using the zone gatekeeper command.
12.2(2)T
The changes in Cisco IOS Release 12.1(5)XM were integrated into Cisco IOS Release 12.2(2)T.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
Usage Guidelines
This command, in conjunction with the bandwidth remote command, replaces the zone gatekeeper command. To specify maximum bandwidth for traffic between one zone and any other zone, use the default keyword with the interzone keyword. To specify maximum bandwidth for traffic within one zone or for traffic between that zone and another zone (interzone or intrazone), use the default keyword with the total keyword.
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Cisco IOS Voice Commands: B bandwidth
To specify maximum bandwidth for a single session within a specific zone, use the zone keyword with the session keyword. To specify maximum bandwidth for a single session within any zone, use the default keyword with the session keyword.
Examples
The following example configures the default maximum bandwidth for traffic between one zone and another zone to 5000 kbps: gatekeeper bandwidth interzone default 5000
The following example configures the default maximum bandwidth for all zones to 5000 kbps: gatekeeper bandwidth total default 5000
The following example configures the default maximum bandwidth for a single session within any zone to 2000 kbps: gatekeeper bandwidth session default 2000
The following example configures the default maximum bandwidth for a single session with a specific zone to 1000 kbps: gatekeeper bandwidth session zone example 1000
Related Commands
Command
Description
bandwidth check-destination
Enables the gatekeeper to verify available bandwidth resources at the destination endpoint.
bandwidth remote
Specifies the total bandwidth for H.323 traffic between this gatekeeper and any other gatekeeper.
h323 interface
Defines on which port the proxy listens.
h323 t120
Enables the T.120 capabilities on the router and specifies bypass or proxy mode.
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Cisco IOS Voice Commands: B bandwidth check-destination
bandwidth check-destination To enable the gatekeeper to verify available bandwidth resources at the destination endpoint, use the bandwidth check-destination command in gatekeeper configuration mode. To disable resource verification, use the no form of this command. bandwidth check-destination no bandwidth check-destination
Syntax Description
This command has no arguments or keywords.
Command Default
Resource verification is disabled by default.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.3(1)
This command was introduced.
Examples
The following example activates bandwidth resource verification at the destination: gatekeeper bandwidth check-destination
Related Commands
Command
Description
bandwidth
Specifies the maximum aggregate bandwidth for H.323 traffic from a zone to another zone, within a zone, or for a session in a zone.
bandwidth remote
Specifies the total bandwidth for H.323 traffic between this gatekeeper and any other gatekeeper.
h323 interface
Defines the port on which port the proxy listens.
h323 t120
Enables the T.120 capabilities on your router and specifies bypass or proxy mode.
Cisco IOS Voice Command Reference
VR-83
Cisco IOS Voice Commands: B bandwidth remote
bandwidth remote To specify the total bandwidth for H.323 traffic between this gatekeeper and any other gatekeeper, use the bandwidth remote command in gatekeeper configuration mode. To disable total bandwidth specified, use the no form of this command. bandwidth remote bandwidth-size no bandwidth remote
Syntax Description
bandwidth-size
Command Default
Total bandwidth is unlimited by default.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series.
12.2(2)T
This command was integrated into Cisco IOS Release 12.2(2)T.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
Maximum bandwidth, in kbps. Range: 1 to 10000000.
Usage Guidelines
This command, with the bandwidth command, replaces the zone gatekeeper command.
Examples
The following example configures the remote maximum bandwidth to 100,000 kbps: gatekeeper bandwidth remote 100000
Related Commands
Command
Description
bandwidth
Specifies the maximum aggregate bandwidth for H.323 traffic from a zone to another zone, within a zone, or for a session in a zone.
bandwidth check-destination
Enables the gatekeeper to verify available bandwidth resources at the destination endpoint.
h323 interface
Defines which port the proxy listens on.
h323 t120
Enables the T.120 capabilities on your router and specifies bypass or proxy mode.
Cisco IOS Voice Command Reference
VR-84
Cisco IOS Voice Commands: B battery-reversal
battery-reversal To specify battery polarity reversal on a Foreign Exchange Office (FXO) or Foreign Exchange Station (FXS) port, use the battery-reversal command in voice-port configuration mode. To disable battery reversal, use the no form of this command. battery-reversal [answer] no battery-reversal [answer]
Syntax Description
answer
Command Default
Battery reversal is enabled
Command Modes
Voice-port configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco 2600 series and Cisco 3600 series and on the Cisco MC3810.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.2(2)T
The answer keyword was added.
Usage Guidelines
(Optional) Configures an FXO port to support answer supervision by detection of battery reversal.
The battery-reversal command applies to FXO and FXS voice ports. On Cisco 2600 and 3600 series routers, only analog voice ports in VIC-2FXO-M1 and VIC-2FXO-M2 voice interface cards are able to detect battery reversal; analog voice ports in VIC-2FXO and VIC-2FXO-EU voice interface cards do not detect battery reversal. On digital voice ports, battery reversal is supported only on E1 Mercury Exchange Limited Channel Associated Signaling (MEL CAS); it is not supported in T1 channel associated signaling (CAS) or E1 CAS. FXS ports normally reverse battery upon call connection. If an FXS port is connected to an FXO port that does not support battery reversal detection, you can use the no battery-reversal command on the FXS port to prevent unexpected behavior. FXO ports in loopstart mode normally disconnect calls when they detect a second battery reversal (back to normal). You can use the no battery-reversal command on FXO ports to disable this action. The battery-reversal command restores voice ports to their default battery-reversal operation. If an FXO voice port is connected to the PSTN and supports battery reversal, use the battery-reversal command with the answer keyword to configure answer supervision. This configures the FXO voice port to detect when a call is answered in order to provide correct billing information. If the voice port, PSTN, or PBX does not support battery reversal, do not use the battery-reversal command because it prevents outgoing calls from being connected. Use the supervisory answer dualtone command instead.
Cisco IOS Voice Command Reference
VR-85
Cisco IOS Voice Commands: B battery-reversal
If an FXO port or its peer FXS port does not support battery reversal, avoid configuring battery-reversal or battery-reversal answer on the FXO port. On FXO ports that do not support battery reversal, the battery-reversal command can cause unpredictable behavior, and the battery-reversal answer command prevents calls from being answered. To ensure that battery reversal answer is disabled on FXO ports that do not support battery reversal, use the no battery-reversal command.
Examples
The following example disables battery reversal on voice port 1/0/0 on a router: voice-port 1/0/0 no battery-reversal
The following example enables battery reversal to provide answer supervision on voice port 1/0/0 on a router: voice-port 1/0/0 battery-reversal answer
Related Commands
Command
Description
show voice port
Displays voice port configuration information.
supervisory answer dualtone Enables answer supervision on an FXO voice port on which battery reversal is not supported.
Cisco IOS Voice Command Reference
VR-86
Cisco IOS Voice Commands: B bearer-capability clear-channel
bearer-capability clear-channel To specify the information transfer capability of the bearer capability information element (IE) in the outgoing ISDN SETUP message for Session Initiation Protocol (SIP) early-media calls that negotiate the clear-channel codec, use the bearer-capability clear-channel command in SIP configuration mode. To reset the information transfer capability of the bearer capability IE to speech (default), use the no form of this command. bearer-capability clear-channel {speech | udi | rdi | audio | tones | video} no bearer-capability clear-channel
Syntax Description
speech
Specifies speech as the information transfer capability (default).
udi
Specifies untrestricted digital information (UDI).
rdi
Specifies restricted digital information (RDI).
audio
Specifies 3.1 kHz audio.
tones
Specifies UDI with tones and announcements.
video
Specifies video as the information transfer capability.
Command Default
The default information transfer capability setting for the bearer-capability IE is speech.
Command Modes
SIP configuration (conf-serv-sip)
Command History
Release
Modification
12.4(15)T
This command was introduced.
Usage Guidelines
Note
When a Cisco voice gateway receives a SIP early-media call and negotiates the clear-channel codec, the default for the information transfer capability octet (octet 3) of the bearer capability IE in the outgoing ISDN SETUP message is set to speech. Use the bearer-capability clear-channel command to change the information transfer capability of the bearer capability IE to a different value.
Changing the information transfer capability of the bearer capability IE affects only SIP early-media calls. The information transfer capability value is always speech for SIP delayed-media calls, even when the clear-channel codec is negotiated. You can display the current information transfer capability setting for the bearer capability IE using the show running-config command. To show only voice service configuration information, limit the display output to the section on voice service (see the “Examples” section).
Note
When the information transfer capability is set to the default value (speech), the output of the show running-config command does not include the bearer-capability information line.
Cisco IOS Voice Command Reference
VR-87
Cisco IOS Voice Commands: B bearer-capability clear-channel
Examples
The following examples show how to configure the information transfer capability of the bearer capability IE to UDI to allow for 64 kb/s data transfer over ISDN and how to display the current setting. Use the following commands to change the information transfer capability setting in the bearer capability IE to udi: voice service voip sip bearer-capability clear-channel udi
Use the following command to display the current information transfer capability setting: Router# show running-config | section voice service voice service voip h323 sip bearer-capability clear-channel udi
Cisco IOS Voice Command Reference
VR-88
Cisco IOS Voice Commands: B billing b-channel
billing b-channel To enable the H.323 gateway to access B-channel information for all H.323 calls, use the billing b-channel command in H.323 voice service configuration mode. To return to the default setting, use the no form of this command. billing b-channel no billing b-channel
Syntax Description
This command has no arguments or keywords.
Command Default
B-channel information is disabled.
Command Modes
H.323 voice service configuration
Command History
Release
Modification
12.3(7)T
This command was introduced.
Usage Guidelines
This command enables the H.323 application to receive B-channel information of incoming ISDN calls. The B-channel information appears in H.323 ARQ / LRQ messages and can be used during call transfer or to route a call.
Examples
The following example adds B-channel information to the H.323 gateway: Router(config)# voice service voip Router(conf-voi-serv)# h323 Router(conf-serv-h323)# billing b-channel
Related Commands
Command
Description
h323
Enables H.323 voice service configuration commands.
voice service
Enters voice-service configuration mode and specifies the voice encapsulation type.
Cisco IOS Voice Command Reference
VR-89
Cisco IOS Voice Commands: B bind
bind To bind the source address for signaling and media packets to the IPv4 or IPv6 address of a specific interface, use the bind command in SIP configuration mode. To disable binding, use the no form of this command. bind {control | media | all} source-interface interface-id [ipv4-address ipv4-address | ipv6-address ipv6-address] no bind
Binds SIP signaling and media packets. The source address (the address that shows where the SIP request came from) of the signaling and media packets is set to the IPv4 or IPv6 address of the specified interface.
source-interface
Specifies an interface as the source address of SIP packets.
interface-id
Specifies one of the following interfaces: Async: ATM interface
•
BVI: Bridge-Group Virtual Interface
•
CTunnel: CTunnel interface
•
Dialer: Dialer interface
•
Ethernet: IEEE 802.3
•
FastEthernet: Fast Ethernet
•
Lex: Lex interface
•
Loopback: Loopback interface
•
Multilink: Multilink-group interface
•
Null: Null interface
•
Serial: Serial interface (Frame Relay)
•
Tunnel: Tunnel interface
•
Vif: PGM Multicast Host interface
•
Virtual-Template: Virtual template interface
•
Virtual-TokenRing: Virtual token ring
ipv4-address ipv4-address
(Optional) Configures the IPv4 address. Several IPv4 addresses can be configured under one interface.
ipv6-address ipv6-address
(Optional) Configures the IPv6 address under an IPv4 interface. Several IPv6 addresses can be configured under one IPv4 interface.
Binding is disabled.
Cisco IOS Voice Command Reference
VR-90
•
Cisco IOS Voice Commands: B bind
Command Modes
SIP configuration (conf-serv-sip)
Command History
Release
Modification
12.2(2)XB
This command was introduced on the Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco AS5300, Cisco AS5350, and Cisco AS5400.
12.2(2)XB2
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. This command does not support the Cisco AS5300, Cisco AS5350, Cisco AS5850, and Cisco AS5400 in this release.
12.3(4)T
The media keyword was added.
12.4(22)T
Support for IPv6 was added.
Cisco IOS XE Release 2.5
This command was integrated into Cisco IOS XE Release 2.5
Usage Guidelines
Async, Ethernet, FastEthernet, Loopback, and Serial (including Frame Relay) are interfaces within the SIP application. If the bind command is not enabled, the IPv4 layer still provides the best local address.
Examples
The following example sets up binding on a SIP network: Router(config)# voice serv voip Router(config-voi-serv)# sip Router(config-serv-sip)# bind control source-interface FastEthernet 0
Related Commands
Command
Description
sip
Enters SIP configuration mode from voice service VoIP configuration mode.
Cisco IOS Voice Command Reference
VR-91
Cisco IOS Voice Commands: B bind interface
bind interface To bind an interface to a Cisco CallManager group, use the bind interface command in SCCP Cisco CallManager configuration mode. To unbind the selected interface, use the no form of this command. bind interface interface-type interface-number no bind interface interface-type interface-number
Syntax Description
interface-type
Type of selected interface.
interface-number
Number of the selected interface.
Command Default
Interfaces are not associated with any Cisco CallManager group.
Command Modes
SCCP Cisco CallManager configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
Usage Guidelines
Note
Examples
The selected interface is used for all calls that belong to the profiles that are associated to this Cisco CallManager group. If the interface is not selected, it selects the best interface’s IP address in the gateway. Interfaces are selected according to user requirements. If there is only one group interface, configuration is not needed.
Only one interface can be selected. A given interface can be bound to more than one Cisco CallManager group.
The following example binds the interface to a specific Cisco CallManager group: Router(config-sccp-ccm)# bind interface fastethernet 2:1
Related Commands
Command
Description
associate profile
Associates a DSP farm profile with a Cisco CallManager group.
sccp ccm group
Creates a Cisco CallManger group and enters SCCP Cisco CallManager configuration mode.
Cisco IOS Voice Command Reference
VR-92
Cisco IOS Voice Commands: B block
block To configure global settings to drop (not pass) specific incoming Session Initiation Protocol (SIP) provisional response messages on a Cisco IOS voice gateway or Cisco Unified Border Element (Cisco UBE), use the block command in voice service SIP configuration mode. To disable a global configuration to drop incoming SIP provisional response messages, use the no form of this command. block {180 | 181 | 183} [sdp {absent | present}] no block {180 | 181 | 183}
Syntax Description
180
Specifies that incoming SIP 180 Ringing messages should be dropped (not passed to the other leg).
181
Specifies that incoming SIP 181 Call is Being Forwarded messages should be dropped (not passed to the other leg).
183
Specifies that incoming SIP 183 Session in Progress messages should be dropped (not passed to the other leg).
sdp
(Optional) Specifies that either the presence or absence of Session Description Protocol (SDP) information in the received response determines when the dropping of specified incoming SIP messages takes place.
absent
Configures the SDP option so that specified incoming SIP messages are dropped only if SDP is absent from the received provisional response.
present
Configures the SDP option so that specified incoming SIP messages are dropped only if SDP is present in the received provisional response.
Command Default
Incoming SIP 180, 181, and 183 provisional responses are forwarded.
Command Modes
Voice service SIP configuration (conf-serv-sip)
Command History
Release
Modification
12.4(22)YB
This command was introduced. Only SIP 180 and SIP 183 messages are supported on Cisco UBEs.
15.0(1)M
This command was integrated into Cisco IOS Release 15.0(1)M.
15.0(1)XA
This command was modified. Support was added for SIP 181 messages on the Cisco IOS SIP gateway, SIP-SIP Cisco UBEs, and the SIP trunk of Cisco Unified Communications Manager Express (Cisco Unified CME).
15.1(1)T
This command was integrated into Cisco IOS Release 15.1(1)T.
Usage Guidelines
Use the block command in voice service SIP configuration mode to globally configure Cisco IOS voice gateways and Cisco UBEs to drop specified SIP provisional response messages. Additionally, you can use the sdp keyword to further control when the specified SIP message is dropped based on either the absence or presence of SDP information.
Cisco IOS Voice Command Reference
VR-93
Cisco IOS Voice Commands: B block
To configure settings for an individual dial peer, use the voice-class sip block command in dial peer voice configuration mode. To disable global configurations for dropping specified incoming SIP messages on a Cisco IOS voice gateway or Cisco UBE, use the no block command in voice service SIP configuration mode. Note
Examples
This command is supported only on outbound dial peers—it is nonoperational if configured on inbound dial peers. You should configure this command on the outbound SIP leg that sends out the initial INVITE message. Additionally, this feature applies only to SIP-to-SIP calls and will have no effect on H.323-to-SIP calls.
The following example shows how to globally configure dropping of incoming SIP provisional response messages: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# block 181
The following example shows how to globally configure dropping of incoming SIP with SDP provisional response messages: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# block 183 sdp present
The following example shows how to globally configure dropping of incoming SIP without SDP provisional response messages: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# block 180 sdp absent
The following example shows how to globally configure passing all specified incoming SIP provisional response messages (except for those on individual dial peers that are configured to override the global configuration): Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# no block 181
Related Commands
Command
Description
map resp-code
Configures global settings on a Cisco UBE for mapping specific incoming SIP provisional response messages to a different SIP response message.
voice-class sip block
Configures an individual dial peer on a Cisco IOS voice gateway or Cisco UBE to drop specified SIP provisional response messages.
voice-class sip map resp-code
Configures a specific dial peer on a Cisco UBE to map specific incoming SIP provisional response messages to a different SIP response message.
Cisco IOS Voice Command Reference
VR-94
Cisco IOS Voice Commands: B block-caller
block-caller To configure call blocking on caller ID, use the block-caller command in dial peer voice configuration mode. To disable call blocking on caller ID, use the no form of this command. block-caller number no block-caller number
Syntax Description
number
Command Default
Call blocking is disabled; the router does not block any calls for any listed directory numbers (LDNs) based on caller ID numbers
Command Modes
Dial peer voice configuration
Command History
Release
Modification
12.1(2)XF
This command was introduced on the Cisco 800 series routers.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
Specifies the telephone number to block. You can use a period (.) as a digit wildcard. For example, the command block-caller 5.51234 blocks all numbers beginning with the digit 5, followed by any digit, and then sequentially followed by the digits 5, 1, 2, 3, and 4.
This command is available on Cisco 800 series routers that have plain old telephone service (POTS) ports. For each dial peer, you can enter up to ten caller ID numbers to block. The routers do not accept additional caller ID numbers if ten numbers are already present. In that case, a number must be removed before another caller ID number can be added for blocking. If you do not specify the block-caller command for a local directory, all voice calls to that local directory are accepted. If you specify the block-caller command for a local directory, the router verifies that the incoming calling-party number does not match any caller ID numbers in that local directory before processing or accepting the voice call. Each specified caller ID number and incoming calling-party number is compared from right to left, up to the number of digits in the specified caller ID number or incoming calling-party number, whichever has fewer digits. This command is effective only if you subscribe to caller ID service. If you enable call blocking on caller ID without subscribing to the caller ID service, the routers do not perform the verification process on calling-party numbers and do not block any calls.
Examples
The following example configures a router to block calls from a caller whose caller ID number is 408-555-0134. dial-peer voice 1 pots block-caller 4085550134
Cisco IOS Voice Command Reference
VR-95
Cisco IOS Voice Commands: B block-caller
Related Commands
Command
Description
caller-id
Identifies incoming calls with caller ID.
debug pots csm csm
Activates events from which an application can determine and display the status and progress of calls to and from POTS ports.
isdn i-number
Configures several terminal devices to use one subscriber line.
pots call-waiting
Enables local call waiting on a router.
registered-caller ring
Configures the Nariwake service registered caller ring cadence.
Cisco IOS Voice Command Reference
VR-96
Cisco IOS Voice Commands: B bootup e-lead off
bootup e-lead off To prevent an analog ear and mouth (E&M) voice port from keying the attached radio on router boot up, use the bootup e-lead off command in voice-port configuration mode. To allow the analog E&M voice port to key the attached radio on boot up, use the no form of this command. bootup e-lead off no bootup e-lead off
Syntax Description
This command has no arguments or keywords.
Command Default
The analog E&M voice port keys the attached radio on radio boot up.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.3(4)XD
This command was introduced.
12.3(7)T
This command was integrated into Cisco IOS Release 12.3(7)T.
Usage Guidelines
This command configures the E-lead behavior on boot up for both voice ports on the voice interface card (VIC).
Examples
The following example configures the analog E&M voice port to not key the attached radio on router boot up: voice-port 1/0/0 bootup e-lead off
Cisco IOS Voice Command Reference
VR-97
Cisco IOS Voice Commands: B busyout forced
busyout forced To force a voice port into the busyout state, use the busyout forced command in voice-port configuration mode. To remove the voice port from the busyout state, use the no form of this command. busyout forced no busyout forced
Syntax Description
This command has no arguments or keywords.
Command Default
The voice-port is not in the busyout state.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco MC3810.
12.0(7)XK
This command was implemented on the Cisco 2600s series and Cisco 3600 series. On the Cisco MC3810, the voice-port busyout command was eliminated in favor of this command.
12.1(2)T
The command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
If a voice port is in the forced busyout state, only the no busyout forced command can restore the voice port to service. To avoid conflicting command-line interface (CLI) commands, do not use the busyout forced command and the ds0 busyout command on the same controller.
Examples
The following example forces analog voice port 3/1/1 on a Cisco 3600 router into the busyout state: voice-port 3/1/1 busyout forced
The following example forces digital voice port 0/0:12 on a Cisco 3600 router into the busyout state: voice-port 0/0:12 busyout forced
Related Commands
Command
Description
busyout-monitor interface
Configures a voice port to monitor a serial interface for events that would trigger a voice-port busyout.
busyout seize
Changes the busyout seize procedure for a voice port.
show voice busyout
Displays information about the voice busyout state.
Cisco IOS Voice Command Reference
VR-98
Cisco IOS Voice Commands: B busyout monitor
busyout monitor To place a voice port into the busyout monitor state, enter the busyout monitor command in voice-port configuration mode. To remove the busyout monitor state from the voice port, use the no form of this command. busyout monitor {serial interface-number | ethernet interface-number | keepalive} [in-service] no busyout monitor {serial interface-number | ethernet interface-number | keepalive}
Syntax Description
serial
Specifies monitoring of a serial interface. More than one interface can be entered for a voice port.
ethernet
Specifies monitoring of an Ethernet interface. More than one interface can be entered for a voice port.
interface-number
The interface to be monitored for the voice port busyout function.
keepalive
In case of keepalive failures, the selected voice port or ports is busied out.
in-service
(Optional) Configures the voice port to be busied out when any monitored interface comes into service (its state changes to up). If the keyword is not entered, the voice port is busied out when all monitored interfaces go out of service (their state changes to down).
Command Default
The voice port does not monitor any interfaces.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco MC3810.
12.0(5)XE
This command was implemented on the Cisco 7200 series.
12.0(5)XK
This command was implemented on the Cisco 2600 series and Cisco 3600 series.
12.0(7)T
This command was implemented on the Cisco 2600 series and Cisco 3600 series and integrated into Cisco IOS Release 12.0(7)T.
12.0(7)XK
The ability to monitor an Ethernet port was introduced and the in-service keyword was added. The serial keyword was first supported on the Cisco 2600 series and Cisco 3600 series.
12.1(1)T
The implementation of this command on the Cisco 7200 series was integrated into Cisco IOS Release 12.1(1)T.
12.1(2)T
The serial and ethernet keywords were added, the in-service keyword was integrated into Cisco IOS Release 12.1(2)T, and the interface-number argument was added to the serial and ethernet keywords.
12.1(3)T
The interface keyword was removed.
12.4(6)T
The keepalive keyword was added.
Cisco IOS Voice Command Reference
VR-99
Cisco IOS Voice Commands: B busyout monitor
Usage Guidelines
When you place a voice port in the busyout monitor state, the voice port monitors the specified interface and enters the busyout state when the interface is down. This down state forces the rerouting of calls. The busyout monitor command monitors only the up or down status of an interface—not end-to-end TCP/IP connectivity. When an interface is operational, a busied-out voice port returns to its normal state. This feature can monitor LAN, WAN, and virtual subinterfaces. A voice port can monitor multiple interfaces at the same time. To configure a voice port to monitor multiple interfaces, reenter the busyout monitor command for each additional interface to be monitored. If you specify more than one monitored interface for a voice port, all the monitored interfaces must be down to trigger busyout on the voice port. You can combine in-service and out-of-service monitoring on a voice port. The following rule describes the action if monitored interfaces change state. A voice port is busied out if either of the following occurs:
Examples
•
Any interface monitored for coming into service comes up.
•
All interfaces monitored for going out of service go down.
The following example shows configuration of analog voice port 1/2 to busy out if serial port 0 or 1 comes into service: voice-port 1/2 busyout monitor serial 0 in-service busyout monitor serial 1 in-service
The following example shows configuration of digital voice port 1/2/2 on a Cisco 3600 series router to busy out if serial port 0 goes out of service: voice-port 1/2/2 busyout monitor serial 0
The following example shows configuration of the voice port to monitor two serial interfaces and an Ethernet interface. When all these interfaces are down, the voice port is busied out. When at least one interface is operating, the voice port is put back into a normal state. voice-port 3/0:0 busyout monitor ethernet 0/0 busyout monitor serial 1/0 busyout monitor serial 2/0
The following example shows configuration of the voice port to be busied out in case of a keepalive failure: voice-port 10 busyout monitor keepalive
Related Commands
Command
Description
busyout forced
Forces a voice port into the busyout state.
busyout monitor probe
Configures a voice port to enter busyout state if an SAA probe signal returned from a remote interface crosses a delay or loss threshold.
Cisco IOS Voice Command Reference
VR-100
Cisco IOS Voice Commands: B busyout monitor
busyout seize
Changes the busyout seize procedure for a voice port.
show voice busyout
Displays information about the voice busyout state.
voice-port busyout
Places all voice ports associated with a serial or ATM interface into a busyout state.
Cisco IOS Voice Command Reference
VR-101
Cisco IOS Voice Commands: B busyout monitor action
busyout monitor action To place a voice port into graceful or shutdown busyout state when triggered by the busyout monitor, use the busyout monitor action command in voice-port configuration mode. To remove the voice port from the busyout state, use the no form of this command. busyout monitor action {graceful | shutdown | alarm blue} no busyout monitor action {graceful | shutdown | alarm blue}
Syntax Description
Command Default
graceful
Graceful busyout state.
shutdown
D-channel shutdown busyout state.
alarm blue
Shutdown state with a blue alarm, also known as an alarm-indication signal (AIS).
Default voice busyout behavior without this command is a forced busyout. Default voice busyout behavior for PRI depends on whether or not the ISDN switch type supports service messages: •
If the switch type supports service messages, default voice busyout behavior is to transmit B-channel out-of-service (OOS) messages and to keep the D channel active. D-Channel service-messages are supported on the following ISDN switch-types: NI, 4ESS (User Side only), 5ESS (User Side only), DMS100.
•
If the switch type does not support service messages, default voice busyout behavior is to bring down the D channel.
•
For switch-types not specified above, the D-channel is taken down when the busyout monitor action graceful is configured.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.2(13)T
The busyout monitor action graceful command was introduced on the following platforms: Cisco 2600 series, Cisco 2600XM, Cisco 2691, Cisco 3640, Cisco 3660, Cisco 3725, and Cisco VG200.
12.3(6)
The busyout monitor action shutdown command was introduced on the following platforms: Cisco 1700 series, Cisco IAD2420 series, Cisco 2600 series, Cisco 2600XM series, Cisco 2691, Cisco 3600 series, Cisco 3700 series, Cisco 4224, Cisco 7200 series, Cisco 7301, Cisco 7400 series, Cisco MC3810, Cisco WS-X4604-GWY, and Cisco VG200.
12.3(7)T
The busyout monitor action shutdown command was integrated into Cisco IOS Release 12.3(7)T and support was added for the Cisco IAD2430 series.
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Cisco IOS Voice Commands: B busyout monitor action
Usage Guidelines
Release
Modification
12.4(6)T
The busyout monitor action graceful and busyout monitor action shutdown commands were introduced to replace the busyout action graceful and busyout action shutdown commands.
12.4(9)T
The busyout monitor action command was introduced to combine the busyout monitor action graceful and busyout monitor action shutdown commands. The shutdown alarm blue keywords were added.
Use this command to control busyout behavior that is triggered by the busyout monitor command. This command with the graceful keyword busies out the voice port immediately or, if there is an active call on this voice port, waits until the call is over. This command with the shutdown keyword has the following attributes:
Examples
•
Before Cisco IOS Release 12.2(8)T, when voice busyout is triggered on a PRI voice port, the D channel is deactivated until the busyout trigger is cleared. Some ISDN switch types, however, support in-service and OOS Q.931 messages that permit B channels to be taken out of service while still keeping the D channel active. Starting in Cisco IOS Release 12.3(8)T for these ISDN switch types, OOS messages are sent and the D channel is kept active when a voice busyout is triggered.
•
This keyword is available only for PRI voice ports.
•
For switch-types not specified above, the D-channel is be taken down when the busyout monitor action graceful command is configured.
The following example shows analog voice-port busyout state set to graceful: voice-port 2/0:15 busyout monitor action graceful
The following example shows E1 PRI voice-port busyout state set to shutdown: voice-port 1/1:15 (E1 PRI) busyout monitor gatekeeper busyout monitor action shutdown
The following example shows T1 PRI voice-port busyout state set to shutdown: voice-port 0/1:23 (T1 PRI) busyout monitor gatekeeper busyout monitor action shutdown
Related Commands
Command
Description
busyout forced
Forces a voice port into busyout state.
busyout monitor
Configures a voice port to monitor an interface for events that would trigger voice-port busyout.
busyout monitor backhaul
Configures a voice port to enter busyout-monitor state with backhaul-L3 connectivity monitoring during a WAN failure.
busyout monitor gatekeeper
Configures a voice port to enter busyout state if connectivity to the gatekeeper is lost.
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Cisco IOS Voice Commands: B busyout monitor action
Command
Description
busyout monitor probe Configures a voice port to enter busyout state if an SAA probe signal returned from a remote, IP-addressable interface crosses a specified delay or loss threshold. busyout seize
Changes the busyout seize procedure for a voice port.
show voice busyout
Displays information about voice-busyout state.
voice-port
Enters voice-port configuration mode and identifies the voice port to be configured.
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Cisco IOS Voice Commands: B busyout monitor backhaul
busyout monitor backhaul To configure a voice port to enter busyout-monitor state with backhaul-L3 connectivity monitoring during a wide-area-network (WAN) failure, use the busyout monitor backhaul command in voice-port configuration mode. To disable busyout-monitor state, use the no form of this command. busyout monitor backhaul no busyout monitor backhaul
Syntax Description
This command has no arguments or keywords.
Command Default
If this command is not used, the voice port is not configured to enter busyout state during a WAN failure.
Command Modes
Voice-port configuration
Command History
Release
Modification
12.4(9)T
This command was introduced.
Usage Guidelines
Use this command to implement backhaul-L3 connectivity monitoring.
Examples
The following example configures a voice port to enter busyout-monitor state with backhaul-L3 connectivity monitoring during a WAN failure: Router(config-voiceport)# busyout monitor backhaul
Related Commands
Command
Description
busyout monitor action
Places a voice port into busyout state.
busyout monitor
Configures a voice port to enter busyout-monitor state.
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Cisco IOS Voice Commands: B busyout monitor gatekeeper
busyout monitor gatekeeper To configure a voice port to enter the busyout state if connectivity to the gatekeeper is lost, use the busyout monitor gatekeeper command in voice-port configuration mode. To configure the monitor to trigger a busyout when any voice port assigned to a specific voice class loses connectivity to the gatekeeper, use the busyout monitor gatekeeper command in voice-class configuration mode. To disable the busyout monitoring state for the gatekeeper, use the no form of this command. busyout monitor gatekeeper no busyout monitor gatekeeper
Syntax Description
This command has no arguments or keywords.
Command Default
If this command is not used, the voice port or voice class is not configured to enter a busyout state if connectivity to the gatekeeper is lost.
This command was introduced on the following platforms: Cisco 2600 series, Cisco 2600XM, Cisco 2691, Cisco 3640, Cisco 3660, Cisco 3725 and Cisco VG200.
12.4(6)T
This command was extended to include functionality in voice-class configuration mode.
Usage Guidelines
Use this command to monitor the connection between the gateway and gatekeeper.
Examples
The following example shows the busyout monitor state set to busyout the port according to the state of the gatekeeper: voice-port 1/1/1 busyout monitor gatekeeper
The following example enters voice-class (busyout) configuration mode and creates a voice class named 33. The monitor is set to busyout when any voice port in voice class 33 loses connectivity to the gatekeeper: voice-class busyout 33 busyout monitor gatekeeper
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Cisco IOS Voice Commands: B busyout monitor gatekeeper
Related Commands
Command
Description
busyout monitor action graceful
Places a voice port into the graceful busyout state when triggered by the busyout monitor.
busyout monitor action graceful
Shuts down the voice port immediately, but if there is an active call it waits until the call is over.
busyout forced
Forces a voice port into the busyout state.
busyout monitor
Configures a voice port to monitor an interface for events that would trigger a voice-port busyout.
busyout monitor probe Configures a voice port to enter the busyout state if an SAA probe signal returned from a remote, IP-addressable interface crosses a specified delay or loss threshold. busyout seize
Changes the busyout seize procedure for a voice port.
show voice busyout
Displays information about the voice busyout state.
voice-port
Enters voice-port configuration mode and identifies the voice port to be configured.
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Cisco IOS Voice Commands: B busyout monitor probe
busyout monitor probe To configure a voice port to enter the busyout state if a Service Assurance Agent (SAA) probe signal is returned from a remote IP-addressable interface after the expiration of a specified delay or loss threshold, use the busyout monitor probe command in voice-port configuration mode or voice class busyout mode. To configure a voice port not to monitor SAA probe signals, use the no form of this command. busyout monitor probe [icmp-ping] ip-address [codec codec-type | size bytes] [icpif number | loss percent delay milliseconds ] [grace-period seconds] size no busyout monitor probe ip-address
Syntax Description
icmp-ping
(Optional) Configures voice-port parameters to use ICMP pings to monitor IP destinations.
ip-address
The IP address of a target interface for the SAA probe signal.
codec
(Optional) Configures the profile of the SAA probe signal to mimic the packet size and interval of a specific codec type.
codec-type
(Optional) The codec type for the SAA probe signal. Available options are as follows: g711a—G.711 a-law
•
g711u—G.711 mu-law (the default)
•
g729—G.729
•
g729a—G.729 Annex A
•
g729b—G.729 Annex B
size bytes
(Optional) Size (in bytes) of the ping packet. Default is 32.
icpif
(Optional) Configures the busyout monitor probe to use an Impairment/Calculated Planning Impairment Factor (ICPIF) loss/delay busyout threshold, in accordance with ITU-T G.113. The ICPIF numbers represent predefined combinations of loss and delay.
number
(Optional) The ICPIF threshold for initiating a busyout condition. Range is from 0 to 30. Low numbers are equivalent to low loss and delay thresholds.
loss
(Optional) Configures the percentage-of-packets-lost threshold for initiating a busyout condition.
percent
(Optional) The loss value (expressed as a percentage) for initiating a busyout condition. Range is from 1 to 100.
delay
(Optional) Configures the average packet delay threshold for initiating a busyout condition.
milliseconds
(Optional) The delay threshold, in milliseconds, for initiating a busyout condition. Range is from 1 to 2,147,483,647.
grace-period
(Optional) Configures a time limit that the system waits before initiating a busyout condition after the loss of SAA probe connectivity.
seconds
(Optional) Number of seconds for the duration of the grace period. Range is from 30 to 300.
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•
Cisco IOS Voice Commands: B busyout monitor probe
Command Default
If the busyout monitor probe command is not entered, the voice port does not monitor SAA probe signals. If the busyout monitor probe command is entered with no optional keywords or arguments, the default codec type is G.711 a-law, the default loss and delay thresholds are the threshold values that are configured with the call fallback threshold delay-loss command, and the loss of SAA connectivity causes an immediate forced busyout condition.
Command Modes
Voice-port configuration and voice class busyout
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 and Cisco 3600 series and on the Cisco MC3810.
12.3(15)
This command was integrated into Cisco IOS Release 12.3(15) and the grace-period keyword and seconds argument were added.
12.4(1)
This command was integrated into Cisco IOS Release 12.4(1).
12.4(2)T
This command was integrated into Cisco IOS Release 12.4(2)T.
Usage Guidelines
Caution
A voice port can monitor multiple interfaces at the same time. To configure a voice port to monitor multiple interfaces, enter the busyout monitor probe command for each additional interface to be monitored.
The busyout monitor probe command is effective only if the call fallback function is enabled on the source router, and the SAA responder is enabled on the target router. To enable the call fallback function, you must enter the call fallback active command for the busyout monitor probe command to work. The SAA probe is transmitted periodically with a period determined by the call fallback function. Low thresholds of ICPIF, loss, and delay result in early busyout when the link deteriorates, thereby raising the voice minimum quality level. High thresholds prevent busyout until loss and delay are long, allowing transmission of lower-quality voice.
Caution
If thresholds are set too low, the link can alternate between in-service and out-of-service states, causing repeated interruptions of traffic. Before the introduction of the grace-period keyword to the busyout monitor probe command, the loss of SAA probe connectivity was sufficient to immediately enforce busyout, causing service and connectivity problems in some networks because busyout conditions could occur frequently and abruptly. To improve busyout monitoring via SAA probes, the grace-period setting allows for an additional timer that must expire before a busyout condition is enforced. That is, the SAA probes and the period of grace must both expire before a busyout condition is invoked. If the SAA IP connectivity is restored within the period of grace, the busyout condition does not occur.
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Cisco IOS Voice Commands: B busyout monitor probe
Note
To disable the grace-period option, you must first enter the no busyout monitor probe command and then re-enter the busyout monitor probe command without the grace-period option. The grace-period keyword is not available in Cisco IOS Release 12.3T.
Examples
The following example shows how to configure analog voice port 1/1/0 to use an SAA probe with a G.711a-law profile to probe the link to two remote interfaces that have IP addresses and to busy out the voice port if SAA probe connectivity is lost for at least 5 seconds. Both links have a loss exceeding 25 percent or a packet delay of more than 1.5 seconds. voice-port 1/1/0 busyout monitor probe 209.165.202.128 codec g711a loss 25 delay 1500 grace-period 45 busyout monitor probe 209.165.202.129 codec g711a loss 25 delay 1500 grace-period 45
Related Commands
Command
Description
busyout monitor
Places a voice port into the busyout monitor state.
call fallback active
Enables the ICMP-ping or SAA (formerly RTR) probe mechanism for use with the dial-peer monitor probe or voice-port busyout monitor probe commands.
call fallback threshold delay-loss
Forces a voice port into the busyout state.
show voice busyout
Displays information about the voice busyout state.
voice class busyout
Creates a voice class for local voice busyout functions.
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Cisco IOS Voice Commands: B busyout seize
busyout seize To change the busyout action for a Foreign Exchange Office (FXO) or Foreign Exchange Station (FXS) voice port, use the busyout seize command in voice-port configuration mode. To restore the default busyout action, use the no form of this command. busyout seize {ignore | repeat} no busyout seize
Syntax Description
ignore
Specifies the type of ignore procedure, depending on the type of voice port signaling. See Table 1 for more information.
repeat
Specifies the type of repeat procedure, depending on the type of voice port signaling. See Table 1 for more information.
Command Default
See Table 1 for the default actions for different voice ports and signaling types
Command Modes
Voice-port configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco MC3810.
12.0(7)XK
This command was implemented on the Cisco 2600 and Cisco 3600 series.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
The busyout seize command is valid for both analog and digital voice ports. On digital voice ports, the busyout actions are valid whether the busyout results from a voice-port busyout event or from the ds0-busyout command. The voice port returns to an idle state when the event that triggered the busyout disappears. Table 1 describes the busyout actions for the busyout seize settings on each voice port type. The busyout action for E and M voice ports is to seize the far end by setting lead busy.
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Cisco IOS Voice Commands: B busyout seize
Table 1
Busyout Seize Actions for Voice Ports
Voice Port Signaling Type
Procedure Setting (busyout-option command)
FXS loop start
Default
Removes the power from the loop. For analog voice ports, this is equivalent to removing the ground from the tip lead. For digital voice ports, the port generates the bit pattern equivalent to removing the ground from the tip lead, or it busies out if the bit pattern exists.
If the far end then opens the loop, FXS removes the ground from the tip lead.
5.
FXS waits for several seconds before returning to Step 1.
Closes the loop and stays at this state. 1.
Leaves the loop open.
2.
Ignores the ringing current on the ring level.
1.
Closes the loop.
2.
After the detected far end starts the power denial procedure, FXO opens the loop.
3.
After the detected far end has completed the power denial procedure, FXO waits for several seconds before returning to Step 1.
Grounds the tip lead. 1.
Leaves the loop open.
2.
Ignores the running current on the ring lead, or the ground current on the tip lead.
1.
Grounds the ring lead.
2.
Removes the ground from the ring lead and closes the loop after the detected far end grounds the tip lead.
3.
When the detected far end removes the ground from tip lead, FXO opens the loop.
4.
FXO waits for several seconds before returning to Step 1.
The following example shows configuration of analog voice port 1/1 to perform the ignore actions when busied out: voice-port 1/1 busyout seize ignore
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Busyout Actions
Cisco IOS Voice Commands: B busyout seize
The following example shows configuration of digital voice port 0:2 to perform the repeat actions when busied out: voice-port 0:2 busyout seize repeat
Related Commands
Command
Description
busyout forced
Forces a voice port into the busyout state.
busyout-monitor interface
Configures a voice port to monitor an interface for events that would trigger a voice port busyout.
ds0 busyout
Forces a DS0 time slot on a controller into the busyout state.
show voice busyout
Displays information about the voice busyout state.
voice-port busyout
Places all voice ports associated with a serial or ATM interface into a busyout state.
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Cisco IOS Voice Commands: B busyout seize
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Cisco IOS Voice Commands: C This chapter contains commands to configure and maintain Cisco IOS voice applications. The commands are presented in alphabetical order. Some commands required for configuring voice may be found in other Cisco IOS command references. Use the command reference master index or search online to find these commands. For detailed information on how to configure these applications and features, refer to the Cisco IOS Voice Configuration Library.
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Cisco IOS Voice Commands: C cac master
cac master To configure the call admission control (CAC) operation as master, enter the cac master command in voice-service configuration mode. To restore CAC operation to slave, use the no form of this command. cac master no cac master
Syntax Description
This command has no arguments or keywords.
Defaults
CAC operation is slave
Command Modes
Voice-service configuration
Command History
Release
Modification
12.1(1)XA
This command was introduced on the Cisco MC3810.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.2(2)T
This command was implemented on the Cisco 7200 series.
Usage Guidelines
You should configure the router at opposite ends of an ATM adaptation layer 2 (AAL2) trunk for the opposite CAC operation—master at one end and slave at the other end. A router configured as a master always performs CAC during fax and modem upspeed. A router configured as a slave sends a request for CAC to the CAC master.
Examples
The following example shows configuration of the CAC operation of a router as master: voice service voatm session protocol aal2 cac master
The following example shows configuration of these entities being returned to slave status: voice service voatm session protocol aal2 no cac master
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Cisco IOS Voice Commands: C cac_off
cac_off To disable connection admission control (CAC), use the cac_off command in interface-ATM-VC configuration mode. To enable CAC, use the no form of this command. cac_off no cac_off
Syntax Description
This command has no keywords or arguments.
Command Default
Call admission control is enabled.
Command Modes
Interface-ATM-VC configuration
Command History
Release
Modification
12.3(4)XD
This command was introduced.
12.3(7)T
This command was integrated into Cisco IOS Release 12.3(7)T.
Usage Guidelines
Connection admission control (CAC) is a set of actions taken by each ATM switch during connection setup to determine whether the requested quality of service (QoS) will violate the QoS guarantees for established connections. CAC reserves bandwidth for voice calls, however, the bandwidth required when the lossless compression codec (LLCC) is used is dynamic and usually less than what is generally reserved by CAC. Disabling CAC can help in better utilization of bandwidth when LLCC is used.
Examples
The following example disables call admission control on a PVC: interface ATM0/IMA1.1 point-to-point pvc test1 15/135 cac_off
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Cisco IOS Voice Commands: C cache (neighbor BE)
cache (neighbor BE) To configure the local border element (BE) to cache the descriptors received from its neighbors, use the cache command in neighbor BE configuration mode. To disable caching, use the no form of this command. cache no cache
Syntax Description
This command has no arguments or keywords.
Defaults
Caching is not enabled
Command Modes
Neighbor BE configuration
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400 is not included in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
Usage Guidelines
Use this command to configure the local BE to cache the descriptors received from its neighbor. If caching is enabled, the neighbors are queried at the specified interval for their descriptors.
Examples
The following example shows the border element enabled to cache the descriptors from its neighbors. Router(config-annexg-neigh)# id neighbor-id Router(config-annexg-neigh)# cache
Related Commands
Command
Description
id
Configures the local ID of the neighboring BE.
port
Configures the neighbor’s port number that is used for exchanging Annex G messages.
query-interval
Configures the interval at which the local BE queries the neighboring BE.
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Cisco IOS Voice Commands: C cache reload time (global application configuration mode)
cache reload time (global application configuration mode) To configure the router to reload scripts from cache on a regular interval, use the cache reload time command in global application configuration mode. To set the value to the default, use the no form of this command. cache reload time bg-minutes no cache reload time
Syntax Description
bg-minutes
Number of minutes after which the background process is awakened. This background process checks the time elapsed since the script was last used and whether the script is current: •
If the script has not been used in the last “unload time,” it unloads the script and quits. The unload time is not configurable.
•
If the script has been used, the background process loads the script from the URL. It compares the scripts, and if they do not match, it begins using the new script for new calls.
Command Default
30 minutes
Command Modes
Global application configuration
Command History
Release
Modification
12.3(14)T
The call application cache reload time command was moved to global application configuration mode and changed to cache reload time.
Examples
The following example displays the cache reload time command configured to specify 15 minutes before a background process is awakened: Enter application configuration mode to configure applications and services: application
Enter global application configuration mode: global
Configure the cache reload time: cache reload time 15
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Cisco IOS Voice Commands: C cache reload time (global application configuration mode)
Related Commands
Command
Description
call application cache reload time
Configures the router to reload the MGCP scripts from cache on a regular interval.
show call application voice Displays all Tcl or MGCP scripts that are loaded.
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Cisco IOS Voice Commands: C cadence
cadence To define the tone-on and tone-off durations for a call-progress tone, use the cadence command in call-progress dualtone configuration mode. To restore the default cadence, use the no form of this command. cadence {cycle-1-on-time cycle-1-off-time [cycle-2-on-time cycle-2-off-time] [cycle-3-on-time cycle-3-off-time] [cycle-4-on-time cycle-4-off-time]} | continuous no cadence
Syntax Description
cycle-1-on-time
Tone-on duration for the first cycle of the cadence pattern, in milliseconds (ms). Range is from 0 to 1000. The default is 0.
cycle-1-off-time
Tone-off duration for the first cycle of the cadence pattern, in milliseconds. Range is from 0 to 1000. The default is 0.
cycle-2-on-time
(Optional) Tone-on duration for the second cycle of the cadence pattern, in milliseconds. Range is from 0 to 1000. The default is 0.
cycle-2-off-time
(Optional) Tone-off duration for the second cycle of the cadence pattern, in milliseconds. Range is from 0 to 1000. The default is 0.
cycle-3-on-time
(Optional) Tone-on duration for the third cycle of the cadence pattern, in milliseconds. Range is from 0 to 1000. The default is 0.
cycle-3-off-time
(Optional) Tone-off duration for the third cycle of the cadence pattern, in milliseconds. Range is from 0 to 1000. The default is 0.
cycle-4-on-time
(Optional) Tone-on duration for the fourth cycle of the cadence pattern, in milliseconds. Range is from 0 to 1000. The default is 0.
cycle-4-off-time
(Optional) Tone-off duration for the fourth cycle of the cadence pattern, in milliseconds. Range is from 0 to 1000. The default is 0.
continuous
Continuous call-progress tone is detected.
Command Default
Continuous
Command Modes
Call-progress dualtone configuration
Command History
Release
Modification
12.1(5)XM
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and the Cisco MC3810.
12.2(2)T
This command was implemented on the Cisco 1750 and integrated into Cisco IOS Release 12.2(2)T.
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Cisco IOS Voice Commands: C cadence
Usage Guidelines
This command specifies the cadence for a class of custom call-progress tones. You must define each cadence that you want a voice port to detect. Reenter the command for each additional cadence to be detected. You must associate the class of custom call-progress tones with a voice port for this command to affect tone detection.
Examples
The following example defines a cadence for a busy tone in the custom-cptone voice class with the name “country-x.” This example defines 500 ms tone on and 500 ms tone off. voice class custom-cptone country-x dualtone busy cadence 500 500
The following example configures detection of the default frequency and cadence values for the busy tone in the custom-cptone voice class with the name “country-x”. The default frequency is a 300 Hz tone, and the default cadence is continuous. voice class custom-cptone country-x dualtone busy no cadence no frequency
Related Commands
Command
Description
supervisory custom-cptone Associates a class of custom call-progress tones with a voice port. voice class custom-cptone
Creates a voice class for defining custom call-progress tones.
voice class dualtone-detect-params
Modifies the boundaries and limits for custom call-progress tones defined by the voice class custom-cptone command.
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Cisco IOS Voice Commands: C cadence-list
cadence-list To specify a tone cadence pattern to be detected, use the cadence-list command in voice-class configuration mode. To delete a cadence pattern, use the no form of this command. cadence-list cadence-id cycle-1-on-time cycle-1-off-time [cycle-2-on-time cycle-2-off-time] [cycle-3-on-time cycle-3-off-time] [cycle-4-on-time cycle-4-off-time] no cadence-list cadence-id
Syntax Description
cadence-id
A tag to identify this cadence list. The range is from 1 to 10.
cycle-1-on-time
The tone duration for the first cycle of the cadence pattern. Range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-1-off-time
The silence duration for the first cycle of the cadence pattern. Range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-2-on-time
(Optional) The tone duration for the second cycle of the cadence pattern. Range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-2-off-time
(Optional) The silence duration for the second cycle of the cadence pattern. Range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-3-on-time
(Optional) The tone duration for the third cycle of the cadence pattern. Range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-3-off-time
(Optional) The silence duration for the third cycle of the cadence pattern. Range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-4-on-time
(Optional) The tone duration for the fourth cycle of the cadence pattern. Range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
cycle-4-off-time
(Optional) The silence duration for the fourth cycle of the cadence pattern. Range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
Command Default
No cadence pattern is configured.
Command Modes
Voice-class configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series, Cisco 3600 series, the Cisco MC3810.
Usage Guidelines
A cadence list enables the router to match a complex tone pattern from a PBX or public switched telephone network (PSTN). A tone is detected if it matches any configured cadence list. You can create up to ten cadence lists, enabling the router to detect up to ten different tone patterns. If the tone to be detected consists of only one on-off cycle, you can configure this in either of two ways: •
Create a cadence list using only the cycle-1-on-time and cycle-1-off-time variables.
•
Use the cadence-max-off-time and cadence-min-on-time commands.
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Cisco IOS Voice Commands: C cadence-list
You must also configure the times of the cadence-max-off-time and cadence-min-on-time commands to be compatible with the on and off times specified by the cadence-list command. The time of the cadence-max-off-time must be equal to or greater than the longest off-time in the cadence list; the cadence-min-on-time must be equal to or less than the shortest on-time in the cadence list.
Examples
The following example shows configuration of cadence list 1 with three on/off cycles and cadence list 2 with two on/off cycles for voice class 100: voice class dualtone 100 cadence-list 1 100 100 300 300 100 200 cadence-list 2 100 200 100 400
Related Commands
Command
Description
cadence-max-off-time
Specifies the maximum off duration for detection of a tone.
cadence-min-on-time
Specifies the minimum on duration for detection of a tone.
voice class dualtone
Creates a voice class for FXO tone detection parameters.
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Cisco IOS Voice Commands: C cadence-max-off-time
cadence-max-off-time To specify the maximum time that a tone can be off and still detected as part of a cadence, use the cadence-max-off-time command in voice-class configuration mode. To restore the default, use the no form of this command. cadence-max-off-time time no cadence-max-off-time
Syntax Description
time
Command Default
0 (no off time)
Command Modes
Voice-class configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series, Cisco 3600 series and the Cisco MC3810.
The maximum off time of a tone that can be detected, in 10-millisecond increments. Range is from 0 to 5000 (0 milliseconds to 50 seconds). The default is 0.
Usage Guidelines
Specify a time value greater than the off time of the tone to be detected, and use a time value greater than 0 to enable detection of a cadenced tone. With the default (0), the router detects only a continuous tone.
Examples
The following example shows configuration of a maximum off duration of 20 seconds for voice class 100: voice class dualtone 100 cadence-max-off-time 2000
Related Commands
Command
Description
cadence-min-on-time
Specifies the minimum on duration for detection of a tone.
cadence-variation
Specifies the cadence variation time allowed for detection of a tone.
voice class dualtone
Creates a voice class for FXO tone detection parameters.
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Cisco IOS Voice Commands: C cadence-min-on-time
cadence-min-on-time To specify the minimum time that a tone can be on and still detected as part of a cadence, use the cadence-min-on-time command in voice-class configuration mode. To restore the default, use the no form of this command. cadence-min-on-time time no cadence-min-on-time
Syntax Description
time
Command Default
0 (no minimum on time)
Command Modes
Voice-class configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series, Cisco 3600 series and the Cisco MC3810.
The minimum on time of a tone that can be detected, in 10-millisecond increments. Range is from 0 to 100 (0 milliseconds to 1 seconds). The default is 0.
Usage Guidelines
Specify a time value shorter than the on time of the tone to be detected. With the default (0), a tone of any length is detected.
Examples
The following example shows configuration of a minimum on duration of 30 milliseconds (three 10-ms time intervals) for voice class 100: voice class dualtone 100 cadence-min-on-time 3
Related Commands
Command
Description
cadence-max-off-time
Specifies the maximum off duration for detection of a tone.
cadence-variation
Specifies the cadence variation time allowed for detection of a tone.
voice class dualtone
Creates a voice class for FXO tone detection parameters.
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Cisco IOS Voice Commands: C cadence-variation
cadence-variation To specify the cadence variation time allowed for detection of a tone, use the cadence-variation command in voice-class configuration mode. To restore the default cadence variation time, use the no form of this command. cadence-variation time no cadence-variation
Syntax Description
time
Command Default
0 milliseconds
Command Modes
Voice-class configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and the Cisco MC3810.
12.1(5)XM
This command was implemented on the Cisco 2600 series, Cisco 3600 series, and the Cisco MC3810.
12.2(2)T
This command was integrated into Cisco IOS Release 12.2(2)T and implemented on the Cisco 1750 router.
Usage Guidelines
The maximum time by which the tone onset can vary from the specified onset time and still be detected, in 10-millisecond increments. Range is from 0 to 200 (0 milliseconds to 2 seconds). The default is 0.
Specify a time value greater than the cadence variation of the tone to be detected. With the default of 0, only those tones that match the configured cadence are detected. This command creates a detection limit for one parameter within a voice class. You can apply the detection limit to any voice port.
Examples
The following example specifies a cadence variation time of 30 milliseconds for voice class 100: voice class dualtone 100 cadence-variation 3
The following example specifies 80 ms (eight 10-ms time intervals) as the maximum allowable cadence variation in voice class 70: voice class dualtone-detect-params 70 cadence-variation 8
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C cadence-variation
Related Commands
Command
Description
cadence-max-off-time
Specifies the maximum off duration for detection of a tone.
cadence-min-on-time
Specifies the minimum on duration for detection of a tone.
supervisory answer dualtone
Enables answer supervision on a voice port.
supervisory dualtone-detect-params
Assigns the boundary and detection tolerance parameters defined by the voice class dualtone-detect-params command to a voice port.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call accounting-template
call accounting-template To select an accounting template at a specific location, use the call accounting-template command in global configuration or application configuration mode. To deselect a specific accounting template, use the no form of this command. call accounting-template acctTempName url no call accounting-template acctTempName url
Syntax Description
acctTempName
Template name.
url
Location of the template.
Command Default
No default behavior or values
Command Modes
Global configuration Application configuration
Command History
Release
Modification
12.2(11)T
This command was introduced on the following platforms: Cisco 3660, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
12.3(14)T
This command was added to the application configuration mode to replace the call application voice accounting-template command.
Usage Guidelines
For call detail records, the template name must have a .cdr extension. To select call records based on your accounting needs and to specify the location of an accounting template that defines the applicable vendor-specific attributes (VSAs) for generating those selected call records, use the call accounting-template command in global configuration mode. The acctTempName argument refers to a specific accounting template file that you want to send to the RADIUS server. This template file defines only specific VSAs selected by you to control your call records based on your accounting needs.
Examples
The example below shows the accounting template cdr1 selected from a specific TFTP address. call accounting-template temp-ivr tftp://kyer/sample/cdr/cdr1.cdr
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call accounting-template
Related Commands
Command
Description
call application voice accounting-template
Configures T.37 fax accounting with VoIP AAA nonblocking API.
show call accounting-template voice
Selects an accounting template at a specific location.
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Cisco IOS Voice Commands: C call accounting-template voice
call accounting-template voice To select an accounting template at a specific location, use the call accounting-template voice command in global configuration mode. To remove a specific accounting template, use the no form of this command. call accounting-template voice acctTempName url no call accounting-template voice acctTempName url
Syntax Description
acctTempName
Template name.
url
Location of the template.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.2(11)T
This command was introduced on the following platforms: Cisco 3660, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
12.3(14)T
The call accounting-template voice command is replaced by the call accounting-template command in application configuration mode. See the call accounting-template command for more information.
Usage Guidelines
The template name must have a .cdr extension. To select call records based on your accounting needs and to specify the location of an accounting template that defines the applicable vendor-specific attributes (VSAs) for generating those selected call records, use the call accounting-template voice command in global configuration mode. The acctTempName argument refers to a specific accounting template file that you want to send to the RADIUS server. This template file defines only specific VSAs selected by you to control your call records based on your accounting needs.
Examples
The example below shows the accounting template cdr1 selected from a specific TFTP address. call accounting-template voice temp-ivr tftp://kyer/sample/cdr/cdr1.cdr
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Cisco IOS Voice Commands: C call accounting-template voice
Related Commands
Command
Description
call accounting-template voice reload
Reloads the accounting template.
show call accounting-template voice
Selects an accounting template at a specific location.
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Cisco IOS Voice Commands: C call accounting-template voice reload
call accounting-template voice reload To reload the accounting template, use the call accounting-template voice reload command in privileged EXEC mode. call accounting-template voice reload acctTempName
Syntax Description
reload
Reloads the accounting template from the address (for example, a tftp address) where the template is stored.
acctTempName
Name of the accounting template.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.2(11)T
This command was introduced on the following platforms: Cisco 3660, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
Usage Guidelines
Use the call accounting-template voice reload command to reload the template from the URL defined in the call accounting-template voice command. After bootup, if the template file fails to load from the TFTP server, the system tries to automatically reload the file at 5-minute intervals.
Examples
The example below shows how to reload accounting template cdr2: call accounting-template voice reload cdr2
Related Commands
Command
Description
call accounting-template voice
Selects an accounting template at a specific location
gw-accounting aaa
Defines and loads the template file at the location defined by the URL.
show call accounting-template voice
Displays the VSAs that are contained in the accounting template.
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Cisco IOS Voice Commands: C call-agent
call-agent To define the call agent for a Media Gateway Control Protocol (MGCP) profile, use the call-agent command in MGCP profile configuration mode. To return to the default values, use the no form of this command. call-agent {dns-name | ip-address} [port] [service-type type] [version protocol-version] no call-agent
Syntax Description
dns-name
Fully qualified domain name (including host portion) for the call agent. For example, “ca123.example.net”.
ip-address
IP address of the call agent.
port
(Optional) User Datagram Protocol (UDP) port number over which the gateway sends messages to the call agent. Range is from 1025 to 65535. •
The default call-agent UDP port is 2727 for MGCP 1.0, Network-based Call Signaling (NCS) 1.0, and Trunking Gateway Control Protocol (TGCP) 1.0.
•
The default call-agent UDP port is 2427 for MGCP 0.1 and Simple Gateway Control Protocol (SGCP).
service-type type
(Optional) Protocol service type valid values for the type argument are mgcp, ncs, sgcp, and tgcp. The default service type is mgcp.
version protocol-version
(Optional) Version number of the protocol. Valid values follow: •
Service-type MGCP—0.1, 1.0
•
Service-type NCS—1.0
•
Service-type SGCP—1.1, 1.5
•
Service-type TGCP—1.0
The default service type and version are mgcp and 0.1.
Command Default
The default call-agent UDP port is 2727 for MGCP 1.0, Network-based Call Signaling (NCS) 1.0, and Trunking Gateway Control Protocol (TGCP) 1.0. The default call-agent UDP port is 2427 for MGCP 0.1 and Simple Gateway Control Protocol (SGCP). The default service type and version are MGCP 0.1.
Command Modes
MGCP profile configuration
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T.
12.2(11)T
This command was implemented on the Cisco AS5300 and Cisco AS5850.
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Cisco IOS Voice Commands: C call-agent
Usage Guidelines
This command is used when values for a MGCP profile are configured. Call-agent configuration for an MGCP profile (with this command) and global call-agent configuration (with the mgcp call-agent command) are mutually exclusive; the first to be configured on an endpoint blocks configuration of the other on the same endpoint. Identifying call agents by Domain Name System (DNS) name rather than by IP address in the call-agent command provides call-agent redundancy, because a DNS name can have more than one IP address associated with it. If a call agent is identified by a DNS name and a message from the gateway fails to reach the call agent, the max1 lookup and max2 lookup commands enable a search from the DNS lookup table for a backup call agent at a different IP address. The port argument configures the call agent port number (the UDP port over which the gateway sends messages to the call agent). The reverse, or the gateway port number (the UDP port over which the gateway receives messages from the call agent), is configured by specifying a port number in the mgcp command. The service type mgcp supports the Restart In Progress (RSIP) error messages sent by the gateway if the mgcp sgcp restart notify command is enabled. The service type sgcp ignores the RSIP messages.
Examples
The following example defines a call agent for the MGCP profile named “tgcp_trunk”: Router(config)# mgcp profile tgcp_trunk Router(config-mgcp-profile)# call-agent 10.13.93.3 2500 service-type tgcp version 1.0
Related Commands
Command
Description
max1 lookup
Enables DNS lookup of the MGCP call agent address when the suspicion threshold value is reached.
max2 lookup
Enables DNS lookup of the MGCP call agent address when the disconnect threshold value is reached.
mgcp
Starts and allocates resources for the MGCP daemon.
mgcp call-agent
Configures the address of the call agent (media gateway controller).
mgcp profile
Initiates MGCP profile mode to create and configure a named MGCP profile associated with one or more endpoints or to configure the default profile.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application alternate
call application alternate Note
Effective with Cisco IOS Release 12.3(14)T, the call application alternate command is replaced by the service command in global application configuration mode. See the service command for more information. To specify an alternate application to use if the application that is configured in the dial peer fails, use the call application alternate command in global configuration mode. To return to the default behavior, use the no form of this command. call application alternate [application-name] no call application alternate
Syntax Description
application-name
Command Default
The call is rejected if the application in the dial peer fails.
Command Modes
Global configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
12.3(14)T
This command was replaced by the service command in global application configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
(Optional) Name of the specific voice application to use if the application in the dial peer fails. If a specific application name is not entered, the gateway uses the DEFAULT application.
If this command is not configured, calls are rejected when the dial peer that matches the call does not specify a valid voice application. In releases before Cisco IOS Release 12.2(11)T, the default application (DEFAULT) was automatically triggered if no application was configured in the dial peer or if the configured application failed. The default application is no longer automatically executed unless the call application alternate command is configured. The application named DEFAULT is a simple application that outputs dial tone, collects digits, and places a call to the dialed number. This application is included in Cisco IOS software; you do not have to download it or configure it by using the call application voice command.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application alternate
The call application alternate command specifies that if the application that is configured in the dial peer fails, the default voice application is executed. If the name of a specific application is entered, that application is triggered if the application configured in the dial peer fails. If the alternate application also fails, the call is rejected. If an application name is entered, that application must first be configured on the gateway by using the call application voice command.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application alternate Warning: This command has been deprecated. Please use the following: service
The following example configures the DEFAULT application as the alternate: call application alternate
The following example configures the application session as the alternate: call application alternate session
Related Commands
Command
Description
application
Enables a voice application on a dial peer.
call application voice
Defines the name of a voice application and specifies the location of the Tcl or VoiceXML document to load for this application.
service
Loads and configures a specific, standalone application on a dial peer.
show call application voice
Displays information about voice applications.
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Cisco IOS Voice Commands: C call application cache reload time
call application cache reload time Note
Effective with Cisco IOS Release 12.3(14)T, the call application cache reload time command is replaced by the cache reload time command in application configuration global mode. See the cache reload time command for more information. To configure the router to reload the Media Gateway Control Protocol (MGCP) scripts from cache on a regular interval, use the call application cache reload time command in global configuration mode. To set the value to the default, use the no form of this command. call application cache reload time bg-minutes no call application cache reload time
Syntax Description
bg-minutes
Specifies the number of minutes after which the background process is awakened. This background process checks the time elapsed since the script was last used and whether the script is current: •
If the script has not been used in the last “unload time,” it unloads the script and quits. The unload time is not configurable.
•
If the script has been used, the background process loads the script from the URL. It compares the scripts, and if they do not match, it begins using the new script for new calls.
Command Default
30 minutes
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco AS5300.
12.3(14)T
This command was replaced by the cache reload time command in application configuration global mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application cache reload time
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application cache reload 20 Warning: This command has been deprecated. Please use the following: cache reload time
The following example displays the call application cache reload time command configured to specify 30 minutes before a background process is awakened: call application cache reload time 30
Related Commands
Command
Description
cache reload time
Configures the router to reload scripts from cache on a regular interval.
call application voice load
Allows reload of an application that was loaded via the MGCP scripting package.
show call application voice Displays all Tcl or MGCP scripts that are loaded.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application dump event-log
call application dump event-log To flush the event log buffer for application instances to an external file, use the call application dump event-log command in privileged EXEC mode. call application dump event-log
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.3(8)T
This command was introduced.
Usage Guidelines
Note
Examples
This command immediately writes the event log buffer to the external file whose location is defined with the call application event-log dump ftp command in global configuration mode.
The call application dump event-log command and the call application event-log dump ftp command are two different commands.
The following example flushes the application event log buffer: Router# call application dump event-log
Related Commands
Command
Description
call application event-log
Enables event logging for voice application instances.
call application event-log dump ftp
Enables the gateway to write the contents of the application event log buffer to an external file.
call application event-log max-buffer-size
Sets the maximum size of the event log buffer for each application instance.
show call application session-level
Displays event logs and statistics for voice application instances.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application event-log
call application event-log Note
Effective with Cisco IOS Release 12.3(14)T, the call application event-log command is replaced by the event-log command in application configuration monitor mode. See the event-log command for more information. To enable event logging for all voice application instances, use the call application event-log command in global configuration mode. To reset to the default, use the no form of this command. call application event-log no call application event-log
Syntax Description
This command has no arguments or keywords.
Command Default
Event logging for voice applications is disabled.
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the event-log command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
Note
This command enables event logging globally for all voice application instances. To enable or disable event logging for a specific application, use the call application voice event-log command.
To prevent event logging from adversely impacting system resources for production traffic, the gateway uses a throttling mechanism. When free processor memory drops below 20%, the gateway automatically disables all event logging. It resumes event logging when free memory rises above 30%. While throttling is occurring, the gateway does not capture any new event logs even if event logging is enabled. You should monitor free memory and enable event logging only when necessary for isolating faults.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application event-log
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application event-log Warning: This command has been deprecated. Please use the following: event-log
The following example enables event logging for all application instances: call application event-log
Related Commands
Command
Description
call application event-log error-only
Restricts event logging to error events only for application instances.
call application event-log max-buffer-size
Sets the maximum size of the event log buffer for each application instance.
call application interface event-log
Enables event logging for external interfaces used by voice applications.
call application stats
Enables statistics collection for voice applications.
call application voice event-log
Enables event logging for a specific voice application.
call leg event-log
Enables event logging for voice, fax, and modem call legs.
event-log
Enables event logging for applications.
monitor call application event-log
Displays the event log for an active application instance in real-time.
show call application session-level
Displays event logs and statistics for voice application instances.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application event-log dump ftp
call application event-log dump ftp Note
Effective with Cisco IOS Release 12.3(14)T, the call application event-log dump ftp command is replaced by the event-log dump ftp command in application configuration monitor mode. See the event-log dump ftp command for more information. To enable the gateway to write the contents of the application event log buffer to an external file, use the call application event-log dump ftp command in global configuration mode. To reset to the default, use the no form of this command. call application event-log dump ftp server[:port]/file username username password [encryption-type] password no call application event-log dump ftp
Syntax Description
server
Name or IP address of FTP server where file is located.
:port
(Optional) Specific port number on server.
/file
Name and path of file.
username
Username required to access file.
encryption-type
(Optional) The Cisco proprietary algorithm used to encrypt the password. Values are 0 or 7. To disable encryption enter 0; to enable encryption enter 7. If you specify 7, you must enter an encrypted password (a password already encrypted by a Cisco router).
password
Password required to access file.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the event-log dump ftp command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
This command enables the gateway to automatically write the event log buffer to the named file either after an active application instance terminates or when the event log buffer becomes full. The default buffer size is 4 KB. To modify the size of the buffer, use the call application event-log max-buffer-size command. To manually flush the event log buffer, use the call application dump event-log command in privileged EXEC mode.
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Cisco IOS Voice Commands: C call application event-log dump ftp
Note
The call application dump event-log command and the call application event-log dump ftp command are two different commands.
Note
Enabling the gateway to write event logs to FTP could adversely impact gateway memory resources in some scenarios, for example, when: •
The gateway is consuming high processor resources and FTP does not have enough processor resources to flush the logged buffers to the FTP server.
•
The designated FTP server is not powerful enough to perform FTP transfers quickly
•
Bandwidth on the link between the gateway and the FTP server is not large enough
•
The gateway is receiving a high volume of short-duration calls or calls that are failing
You should enable FTP dumping only when necessary and not enable it in situations where it might adversely impact system performance.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application dump event-log Warning: This command has been deprecated. Please use the following: event-log dump ftp
The following example enables the gateway to write application event logs to an external file named app_elogs.log on a server named ftp-server: call application event-log dump ftp ftp-server/:elogs/app-elogs.log username myname password 0 mypass
The following example specifies that application event logs are written to an external file named app_elogs.log on a server with the IP address of 10.10.10.101: call application event-log dump ftp 10.10.10.101/:elogs/app-elogs.log username myname password 0 mypass
Related Commands
Command
Description
call application dump event-log
Flushes the event log buffer for application instances to an external file.
call application event-log
Enables event logging for voice application instances.
call application event-log max-buffer-size
Sets the maximum size of the event log buffer for each application instance.
event-log dump ftp
Enables the gateway to write the contents of the application event log buffer to an external file.
show call application session-level
Displays event logs and statistics for voice application instances.
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Cisco IOS Voice Commands: C call application event-log error-only
call application event-log error-only Note
Effective with Cisco IOS Release 12.3(14)T, the call application event-log error-only command is replaced by the event-log error-only command in application configuration monitor mode. See the event-log error-only command for more information. To restrict event logging to error events only for application instances, use the call application event-log error-only command in global configuration mode. To reset to the default, use the no form of this command. call application event-log error-only no call application event-log error-only
Syntax Description
This command has no arguments or keywords.
Command Default
All application events are logged.
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the event-log error-only command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
This command limits new event logging to error events only; it does not enable logging. You must use this command with either the call application event-log command, which enables event logging for all voice applications, or with the call application voice event-log command, which enables event logging for a specific application. Any events logged before this command is issued are not affected.
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Cisco IOS Voice Commands: C call application event-log error-only
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application event-log error-only Warning: This command has been deprecated. Please use the following: event-log error-only
The following example enables event logging for error events only: call application event-log call application event-log error-only
Related Commands
Command
Description
call application event-log
Enables event logging for voice application instances.
call application history session event-log save-exception-only
Saves in history only the event logs for application instances that have at least one error.
call application voice event-log
Enables event logging for a specific voice application.
event-log error-only
Restricts event logging to error events only for application instances.
show call application app-level
Displays application-level statistics for voice applications.
show call application session-level
Displays event logs and statistics for voice application instances.
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Cisco IOS Voice Commands: C call application event-log max-buffer-size
call application event-log max-buffer-size Note
Effective with Cisco IOS Release 12.3(14)T, the call application event-log max-buffer-size command is replaced by the event-log max-buffer-size command in application configuration monitor mode. See the event-log max-buffer-size command for more information. To set the maximum size of the event log buffer for each application instance, use the call application event-log max-buffer-size command in global configuration mode. To reset to the default, use the no form of this command. call application event-log max-buffer-size kilobytes no call application event-log max-buffer-size
Syntax Description
kilobytes
Command Default
4 kilobytes
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the event-log max-buffer-size command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
Maximum buffer size, in kilobytes. Range is 1 to 50. Default is 4.
If the event log buffer reaches the limit set by this command, the gateway allocates a second buffer of equal size. The contents of both buffers is displayed when you use the show call application session-level command. When the first event log buffer becomes full, the gateway automatically appends its contents to an external FTP location if the call application event-log dump ftp command is used. A maximum of two buffers are allocated for an event log. If both buffers are filled, the first buffer is deleted and another buffer is allocated for new events (buffer wraps around). If the call application event-log dump ftp command is configured and the second buffer becomes full before the first buffer is dumped, event messages are dropped and are not recorded in the buffer.
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Cisco IOS Voice Commands: C call application event-log max-buffer-size
Note
Do not set the maximum buffer size to more than you need for a typical application session. After an active session terminates, the amount of memory used by the buffer is allocated to the history table and is maintained for the length of time set by the call application history session retain-timer command. Also consider that most fatal errors are captured at the end of an event log. To conserve memory resources, write the event log buffer to FTP by using the call application event-log dump ftp command.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application event-log max-buffer-size Warning: This command has been deprecated. Please use the following: event-log max-buffer-size
The following example sets the application event log buffer to 8 kilobytes: call application event-log call application event-log max-buffer-size 8
Related Commands
Command
Description
call application dump event-log
Flushes the event log buffer for application instances to an external file.
call application event-log
Enables event logging for voice application instances.
call application event-log dump ftp
Enables the gateway to write the contents of the application event log buffer to an external file.
event-log max-buffer-size
Sets the maximum size of the event log buffer for each application instance.
show call application session-level
Displays event logs and statistics for voice application instances.
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Cisco IOS Voice Commands: C call application global
call application global Note
Effective with Cisco IOS Release 12.3(14)T, the call application global command is replaced by the global command in application configuration mode. See the global command for more information. To configure an application to use for incoming calls whose incoming dial peer does not have an explicit application configured, use the call application global command in global configuration mode. To remove the application, use the no form of this command. call application global application-name no call application global application-name
Syntax Description
application-name
Command Default
The default application is default for all dial peers.
Command Modes
Global configuration
Command History
Release
Modification
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
12.3(14)T
This command was replaced by the global command in application configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
Character string that defines the name of the application.
The application defined in the dial peer always takes precedence over the global application configured with the call application global command. The application configured with this command executes only when a dial peer has no application configured. The application you configure with this command can be an application other than the default session application, but it must be included with the Cisco IOS software or be loaded onto the gateway with the call application voice command before using this command. If the application does not exist in Cisco IOS software or has not been loaded onto the gateway, this command will have no effect.
Note
In Cisco IOS Release 12.3(4)T and later releases, the application-name default refers to the applicationCall Admission Control Based on CPU Utilization that supports Open Settlement Protocol (OSP), call transfer, and call forwarding. The default session application in Cisco IOS Release 12.2(13)T
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Cisco IOS Voice Commands: C call application global
and earlier releases has been renamed default.old.c and can still be configured for specific dial peers through the application command or globally configured for all inbound dial peers through the call application global command.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application global Warning: This command has been deprecated. Please use the following: global
In the following example, the clid_authen_collect application is configured as the global application for all inbound dial peers that do not have a specific application configured: call application global clid_authen_collect
Related Commands
Command
Description
application
Enables a specific IVR application on a dial peer.
call application voice
Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
global
Enters application configuration mode.
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Cisco IOS Voice Commands: C call application history session event-log save-exception-only
call application history session event-log save-exception-only Note
Effective with Cisco IOS Release 12.3(14)T, the call application history session event-log save-exception-only command is replaced by the history session event-log save-exception-only command in application configuration monitor mode. See the history session event-log save-exception-only command for more information. To save in history only the event logs for application sessions that have at least one error, use the call application history session event-log save-exception-only command in global configuration mode. To reset to the default, use the no form of this command. call application history session event-log save-exception-only no call application history session event-log save-exception-only
Syntax Description
This command has no arguments or keywords.
Command Default
All event logs for sessions are saved to history.
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the history session event-log save-exception-only command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
Note
Application event logs move from active to history after an instance terminates. If you use this command, the voice gateway saves event logs only for instances that had one or more errors. Event logs for normal instances that do not contain any errors are not saved to history.
This command does not affect records saved to an FTP server by using the call application dump event-log command.
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Cisco IOS Voice Commands: C call application history session event-log save-exception-only
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application history session event-log save-exception-only Warning: This command has been deprecated. Please use the following: history session event-log save-exception-only
The following example saves an event log in history only if the instance had an error: call application history session event-log save-exception-only
Related Commands
Command
Description
call application event-log
Enables event logging for voice application instances.
call application event-log error-only
Restricts event logging to error events only for application instances.
call application event-log max-buffer-size
Sets the maximum size of the event log buffer for each application instance.
call application history session max-records
Sets the maximum number of application instance records saved in history.
call application history session retain-timer
Sets the maximum number of minutes for which application instance records are saved in history.
history session event-log save-exception-only
Saves in history only the event logs for application sessions that have at least one error.
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Cisco IOS Voice Commands: C call application history session max-records
call application history session max-records Note
Effective with Cisco IOS Release 12.3(14)T, the call application history session max-records command is replaced by the history session max-records command in application configuration monitor mode. See the history session max-records command for more information. To set the maximum number of application instance records saved in history, use the call application history session max-records command in global configuration mode. To reset to the default, use the no form of this command. call application history session max-records number no call application history session max-records
Syntax Description
number
Command Default
360
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the history session max-records command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Maximum number of records to save in history. Range is 0 to 2000. Default is 360.
Usage Guidelines
This command affects the number of records that display when you use the show call application history session-level command.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application history session max-records Warning: This command has been deprecated. Please use the following: history session max-records
The following example sets the maximum record limit to 500: call application history session max-records 500
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Cisco IOS Voice Commands: C call application history session max-records
Related Commands
Command
Description
call application event-log
Enables event logging for voice application instances.
call application history session event-log save-exception-only
Saves in history only the event logs for application instances that have at least one error.
call application history session retain-timer
Sets the maximum number of minutes that application instance records are saved in history.
history session max-records
Sets the maximum number of application instance records saved in history.
show call application session-level
Displays event logs and statistics for voice application instances.
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Cisco IOS Voice Commands: C call application history session retain-timer
call application history session retain-timer Note
Effective with Cisco IOS Release 12.3(14)T, the call application history session retain-timer command is replaced by the history session retain-timer command in application configuration monitor mode. See the history session retain-timer command for more information. To set the maximum number of minutes for which application instance records are saved in history, use the call application history session retain-timer command in global configuration mode. To reset to the default, use the no form of this command. call application history session retain-timer minutes no call application history session retain-timer
Syntax Description
minutes
Command Default
15
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the history session retain-timer command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Maximum time, in minutes, for which history records are saved. Range is 0 to 4294,967,295. Default is 15.
Usage Guidelines
This command affects the number of records that display when you use the show call application history session-level command.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application history session retain-timer Warning: This command has been deprecated. Please use the following: history session retain-timer
The following example sets the maximum time to save history records to 1 hour: call application history session retain-timer 60
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Cisco IOS Voice Commands: C call application history session retain-timer
Related Commands
Command
Description
call application event-log
Enables event logging for voice application instances.
call application history session event-log save-exception-only
Saves in history only the event logs for application instances that have at least one error.
call application history session max-records
Sets the maximum number of application instance records saved in history.
history session retain-timer
Sets the maximum number of minutes for which application instance records are saved in history.
show call application session-level
Displays event logs and statistics for voice application instances.
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Cisco IOS Voice Commands: C call application interface dump event-log
call application interface dump event-log To flush the event log buffer for application interfaces to an external file, use the call application interface dump event-log command in privileged EXEC mode. call application interface dump event-log
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.3(8)T
This command was introduced.
Usage Guidelines
Note
Examples
This command immediately writes the event log buffer to the external file whose location is defined with the call application interface event-log dump ftp command in global configuration mode.
The call application interface dump event-log command and the call application interface event-log dump ftp command are two different commands.
The following example writes the event log buffer to the external file named int_elogs: Router(config)# call application interface event-log dump ftp ftp-server/int_elogs.log username myname password 0 mypass Router(config)# exit Router# call application interface dump event-log
Related Commands
Command
Description
call application interface event-log
Enables event logging for external interfaces used by voice applications.
call application interface event-log dump ftp
Enables the voice gateway to write the contents of the interface event log buffer to an external file.
Sets the maximum size of the event log buffer for each application interface.
show call application interface
Displays event logs and statistics for application interfaces.
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Cisco IOS Voice Commands: C call application interface event-log
call application interface event-log Note
Effective with Cisco IOS Release 12.3(14)T, the call application interface event-log command is replaced by the interface event-log command in application configuration monitor mode. See the interface event-log command for more information. To enable event logging for interfaces that provide services to voice applications, use the call application interface event-log command in global configuration mode. To reset to the default, use the no form of this command. call application interface event-log [{aaa | asr | flash | http | ram | rtsp | smtp | tftp | tts} [server server] [disable]] no call application interface event-log [{aaa | asr | flash | http | ram | rtsp | smtp | tftp | tts} [server server] [disable]]
Syntax Description
aaa
Authentication, authorization, and accounting (AAA) interface type.
Hypertext Transfer Protocol (HTTP) interface type.
ram
Memory of the Cisco gateway.
rtsp
Real Time Streaming Protocol (RTSP) interface type.
smtp
Simple Mail Transfer Protocol (SMTP) interface type.
tftp
Trivial File Transfer Protocol (TFTP) interface type.
tts
Text-to-speech (TTS) interface type.
server server
(Optional) Server name or IP address.
disable
(Optional) Disables event logging for the specified interface type or server.
Command Default
Event logging for application interfaces is disabled.
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the interface event-log command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
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Cisco IOS Voice Commands: C call application interface event-log
Usage Guidelines
Note
Examples
This command enables event logging globally for all interface types and servers unless you select a specific interface type or server. Specifying an interface type takes precedence over the global command for a specific interface type. Specifying an individual server takes precedence over the interface type.
To prevent event logging from adversely impacting system resources for production traffic, the gateway uses a throttling mechanism. When free processor memory drops below 20%, the gateway automatically disables all event logging. It resumes event logging when free memory rises above 30%. While throttling is occurring, the gateway does not capture any new event logs even if event logging is enabled. You should monitor free memory and enable event logging only when necessary for isolating faults.
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application interface event-log Warning: This command has been deprecated. Please use the following: interface event-log
The following example enables event logging for all interfaces: call application interface event-log
The following example enables event logging for HTTP interfaces only: call application interface event-log http
The following example enables event logging for all interfaces except HTTP: call application interface event-log call application interface event-log http disable
The following example enables event logging for all HTTP servers except the server with the IP address of 10.10.1.1: call application interface event-log http call application interface event-log http server http://10.10.1.1 disable
Related Commands
Command
Description
call application interface event-log dump ftp
Enables the gateway to write the contents of the interface event log buffer to an external file.
call application interface event-log error-only
Restricts event logging to error events only for application interfaces.
Effective with Cisco IOS Release 12.3(14)T, the call application interface event-log dump ftp command is replaced by the interface event-log dump ftp command in application configuration monitor mode. See the interface event-log dump ftp command for more information. To enable the gateway to write the contents of the interface event log buffer to an external file, use the call application interface event-log dump ftp command in global configuration mode. To reset to the default, use the no form of this command. call application interface event-log dump ftp server[:port]/file username username password [encryption-type] password no call application interface event-log dump ftp
Syntax Description
server
Name or IP address of FTP server where the file is located.
:port
(Optional) Specific port number on server.
/file
Name and path of file.
username username
Username required to access file.
encryption-type
(Optional) Cisco proprietary algorithm used to encrypt the password. Values are 0 or 7. To disable encryption enter 0; to enable encryption enter 7. If you specify 7, you must enter an encrypted password (a password already encrypted by a Cisco router).
password password
Password required to access file.
Command Default
Interface event log buffer is not written to an external file.
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the interface event-log dump ftp command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
This command enables the gateway to automatically write the interface event log buffer to the named file when the buffer becomes full. The default buffer size is 4 KB. To modify the size of the buffer, use the call application interface event-log max-buffer-size command. To manually flush the event log buffer, use the call application interface dump event-log command in privileged EXEC mode.
The call application interface dump event-log command and the call application interface event-log dump ftp command are two different commands.
•
Enabling the gateway to write event logs to FTP can adversely impact gateway-memory resources in scenarios such as the following: – The gateway is consuming high processor resources and FTP does not have enough processor
resources to flush the logged buffers to the FTP server. – The designated FTP server is not powerful enough to perform FTP transfers quickly – Bandwidth on the link between the gateway and the FTP server is not large enough – The gateway is receiving a high volume of short-duration calls or calls that are failing
You should enable FTP dumping only when necessary and not enable it in situations where it might adversely impact system performance.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application interface event-log dump ftp Warning: This command has been deprecated. Please use the following: interface event-log dump ftp
The following example specifies that interface event log are written to an external file named int_elogs.log on a server named ftp-server: call application interface event-log dump ftp ftp-server/elogs/int_elogs.log username myname password 0 mypass
The following example specifies that application event logs are written to an external file named int_elogs.log on a server with the IP address of 10.10.10.101: call application interface event-log dump ftp 10.10.10.101/elogs/int_elogs.log username myname password 0 mypass
Related Commands
Command
Description
call application interface dump event-log
Flushes the event log buffer for application interfaces to an external file.
call application interface event-log
Enables event logging for external interfaces used by voice applications.
Effective with Cisco IOS Release 12.3(14)T, the call application interface event-log error-only command is replaced by the interface event-log error only command in application configuration monitor mode. See the interface event-log error only command for more information. To restrict event logging to error events only for application interfaces, use the call application interface event-log error-only command in global configuration mode. To reset to the default, use the no form of this command. call application interface event-log error-only no call application interface event-log error-only
Syntax Description
This command has no arguments or keywords.
Command Default
All events are logged.
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the interface event-log error only command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
This command limits the severity level of the events that are logged; it does not enable logging. You must use this command with the call application interface event-log command, which enables event logging for all application interfaces.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application interface event-log error-only Warning: This command has been deprecated. Please use the following: interface event-log error only
The following example enables event logging for error events only: call application interface event-log error-only
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Cisco IOS Voice Commands: C call application interface event-log error-only
Related Commands
Command
Description
call application interface event-log
Enables event logging for external interfaces used by voice applications.
Effective with Cisco IOS Release 12.3(14)T, the call application interface event-log max-buffer-size command is replaced by the interface event-log max-buffer-size command in application configuration monitor mode. See the interface event-log max-buffer-size command for more information. To set the maximum size of the event log buffer for each application interface, use the call application interface event-log max-buffer-size command in global configuration mode. To reset to the default, use the no form of this command. call application interface event-log max-buffer-size kilobytes no call application interface event-log max-buffer-size
Syntax Description
kilobytes
Command Default
4 kilobytes
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the interface event-log max-buffer-size command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
Maximum buffer size, in kilobytes. Range is 1 to 10. Default is 4.
If the event log buffer reaches the limit set by this command, the gateway allocates a second buffer of equal size. The contents of both buffers is displayed when you use the show call application interface command. When the first event log buffer becomes full, the gateway automatically appends its contents to an external FTP location if the call application interface event-log dump ftp command is used. A maximum of two buffers are allocated for an event log. If both buffers are filled, the first buffer is deleted and another buffer is allocated for new events (buffer wraps around). If the call application interface event-log dump ftp command is configured and the second buffer becomes full before the first buffer is dumped, event messages are dropped and are not recorded in the buffer.
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Cisco IOS Voice Commands: C call application interface event-log max-buffer-size
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application interface event-log max-buffer-size Warning: This command has been deprecated. Please use the following: interface event-log max-buffer-size
The following example sets the maximum buffer size to 8 kilobytes: call application interface event-log max-buffer-size 8
Related Commands
Command
Description
call application interface dump event-log
Flushes the event log buffer for application interfaces to an external file.
call application interface event-log dump ftp
Enables the gateway to write the contents of the interface event log buffer to an external file.
call application interface max-server-records
Sets the maximum number of application interface records that are saved.
interface event-log max-buffer-size
Sets the maximum size of the event log buffer for each application interface.
show call application interface
Displays event logs and statistics for application interfaces.
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Cisco IOS Voice Commands: C call application interface max-server-records
Effective with Cisco IOS Release 12.3(14)T, the call application interface max-server-records command is replaced by the interface max-server-records command in application configuration monitor mode. See the interface max-server-records command for more information. To set the maximum number of application interface records that are saved, use the call application interface max-server-records command in global configuration mode. To reset to the default, use the no form of this command. call application interface max-server-records number no call application interface max-server-records
Syntax Description
number
Command Default
10
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the interface max-server-records command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Maximum number of records to save. Range is 1 to 100. Default is 10.
Usage Guidelines
Only the specified number of records from the most recently accessed servers are kept.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application interface max-server-records Warning: This command has been deprecated. Please use the following: interface max-server-records
The following example sets the maximum saved records to 50: call application interface max-server-records 50
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Cisco IOS Voice Commands: C call application interface max-server-records
Related Commands
Command
Description
call application interface event-log
Enables event logging for external interfaces used by voice applications.
Sets the maximum size of the event log buffer for each application interface.
interface max-server-records
Sets the maximum number of application interface records that are saved.
show call application interface
Displays event logs and statistics for application interfaces.
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Cisco IOS Voice Commands: C call application interface stats
call application interface stats Note
Effective with Cisco IOS Release 12.3(14)T, the call application interface stats command is replaced by the interface stats command in application configuration monitor mode. See the interface stats command for more information. To enable statistics collection for application interfaces, use the call application interface stats command in global configuration mode. To reset to the default, use the no form of this command. call application interface stats no call application interface stats
Syntax Description
This command has no arguments or keywords.
Command Default
Statistics collection is disabled.
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the interface stats command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
To display the interface statistics enabled by this command, use the show call application interface command. To reset the interface counters to zero, use the clear call application interface command.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application interface stats Warning: This command has been deprecated. Please use the following: interface stats
The following example enables statistics collection for application interfaces: call application interface stats
Related Commands
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Cisco IOS Voice Commands: C call application interface stats
Command
Description
call application interface event-log
Enables event logging for external interfaces used by voice applications.
clear call application interface
Clears application interface statistics or event logs.
interface stats
Enables statistics collection for application interfaces.
show call application interface
Displays event logs and statistics for application interfaces.
stats
Enables statistics collection for voice applications.
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Cisco IOS Voice Commands: C call application session start (global)
call application session start (global) Note
Effective with Cisco IOS Release 12.3(14)T, the call application session start (global) command is replaced by the the session start command in application configuration mode. See the session start command for more information. To start a new instance (session) of a Tcl IVR 2.0 application, use the call application session start command in global configuration mode. To stop the session and remove the configuration, use the no form of this command. call application session start instance-name application-name no call application session start instance-name
Syntax Description
instance-name
Alphanumeric label that uniquely identifies this application instance.
application-name
Name of the Tcl application. This is the name of the application that was assigned with the call application voice command.
Command Default
This command has no default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.3(4)T
This command was introduced.
12.3(14)T
The call application session start (global configuration) command was replaced by the session start command in application configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
This command starts a new session, or instance, of a Tcl IVR 2.0 application. It cannot start a session for a VoiceXML application because Cisco IOS software cannot start a VoiceXML application without an active call leg. You can start an application instance only after the Tcl application is loaded onto the gateway with the call application voice command. If this command is used, the session restarts if the gateway reboots. The no call application session start command stops the Tcl session and removes the configuration from the gateway. You can stop an application session without removing the configuration by using the call application session stop command.
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Cisco IOS Voice Commands: C call application session start (global)
VoiceXML sessions cannot be stopped with the no call application session start command because VoiceXML sessions cannot be started with Cisco IOS commands. If the application session stops running, it does not restart unless the gateway reboots. A Tcl script might intentionally stop running by executing a “call close” command for example, or it might fail because of a script error. You can start multiple instances of the same application by using different instance names.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application session start Warning: This command has been deprecated. Please use the following: session start
The following example starts a session named my_instance for the application named demo: call application session start my_instance demo
The following example starts another session for the application named demo: call application session start my_instance2 demo
Related Commands
Command
Description
call application session start (privileged EXEC)
Starts a new instance (session) of a Tcl application from privileged EXEC mode.
call application session stop
Stops a voice application session that is running.
debug voip ivr
Displays debug messages for VoIP IVR interactions.
session start
Starts a new instance (session) of a Tcl IVR 2.0 application.
show call application services Displays a one-line summary of all registered services. registry show call application sessions Displays summary or detailed information about voice application sessions.
call application session start (privileged EXEC) To start a new instance (session) of a Tcl IVR 2.0 application, use the call application session start command in privileged EXEC mode. call application session start instance-name [application-name]
Syntax Description
instance-name
Alphanumeric label that uniquely identifies this application instance.
application-name
(Optional) Name of the Tcl application. This is the name of the application that was assigned with the call application voice command. Note
This argument is optional if the application instance was previously started and stopped.
Command Default
None
Command Modes
Privileged EXEC
Command History
Release
Modification
12.3(4)T
This command was introduced.
Usage Guidelines
This command starts a new session, or instance, of a Tcl IVR 2.0 application. It cannot start a session for a VoiceXML application because Cisco IOS software cannot start a VoiceXML application without an active call leg. You can start an application instance only after the Tcl application is loaded onto the gateway with the call application voice command. Using this command does not restart the session if the gateway reboots. To automatically restart the session if the gateway reboots, use the call application session start command in global configuration mode. To stop an application session once it starts running, use the call application session stop command. If the application session stops running, it does not restart unless the gateway reboots and the call application session start command is used in global configuration mode. A Tcl script might intentionally stop running by executing a “call close” command for example, or it might fail due to a script error. You can start multiple instances of the same application by using different instance names.
Examples
The following example restarts an application session called my_instance: call application session start my_instance
Starts a new instance (session) of a Tcl application in global configuration mode.
call application session stop
Stops a voice application session that is running.
show call application services Displays a one-line summary of all registered services. registry show call application sessions Displays summary or detailed information about voice application sessions.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application session stop
call application session stop To stop a voice application session that is running, use the call application session stop command in privileged EXEC mode. call application session stop {callid call-id | handle handle | id session-id | name instance-name}
Syntax Description
callid call-id
Call-leg ID that can be displayed in the output from the debug voip ivr script command if the Tcl script uses puts commands.
handle handle
Handle of a session from the Tcl mod_handle infotag.
id session-id
Session ID that can be displayed with the show call application sessions command.
name instance-name
Instance name that was configured with the call application session start command.
Command Default
None
Command Modes
Privileged EXEC
Command History
Release
Modification
12.3(4)T
This command was introduced.
Usage Guidelines
This command stops a Tcl IVR 2.0 or VoiceXML application session that is identified by one of four different methods: call ID, handle, session ID, or instance name. To see a list of currently running applications, use the show call application sessions command. A Tcl session that is stopped with this command receives a session terminate event. The session is expected to close all call legs and stop. If a session does not close itself after a 10-second timer, it is forcibly stopped and all call legs that it controls disconnect. Using this command to stop a VoiceXML session immediately stops the document interpretation and disconnects the call leg. No VoiceXML events are thrown. If you stop a Tcl session that is configured to start with the call application session start command in global configuration mode, you must remove the session by using the no call application session start command before you can restart it. To see a list of stopped sessions, use the show call application sessions command. Only stopped sessions that are configured to start with the call application session start command in global configuration mode are displayed. If a session was started with the call application session start command in privileged EXEC mode, it is not tracked by the system and it is not shown as stopped in the output of the show call application sessions command.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application session stop
Examples
The following example stops an application session called my_instance: call application session stop name my_instance
Starts a new instance (session) of a Tcl application from global configuration mode.
show call application services registry
Displays a one-line summary of all registered services.
show call application sessions
Displays summary or detailed information about voice application sessions.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application stats
call application stats Note
Effective with Cisco IOS Release 12.3(14)T, the call application stats command is replaced by the stats command in application configuration monitor mode. See the stats command for more information. To enable statistics collection for voice applications, use the call application stats command in global configuration mode. To reset to the default, use the no form of this command. call application stats no call application stats
Syntax Description
This command has no arguments or keywords.
Command Default
Statistics collection is disabled.
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
This command was replaced by the stats command in application configuration monitor mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
To display the application statistics, use the show call application session-level, show call application app-level, or show call application gateway-level command. To reset the application counters in history to zero, use the clear call application stats command.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application stats Warning: This command has been deprecated. Please use the following: stats
The following example enables statistics collection for voice applications: call application stats
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application stats
Related Commands
Command
Description
call application event-log
Enables event logging for voice application instances.
call application interface stats
Enables statistics collection for application interfaces.
clear call application stats Clears application-level statistics in history and subtracts the statistics from the gateway-level statistics. show call application app-level
Displays application-level statistics for voice applications.
show call application gateway-level
Displays gateway-level statistics for voice application instances.
show call application session-level
Displays event logs and statistics for voice application instances.
stats
Enables statistics collection for voice applications.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice
call application voice Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice command is replaced by the commands shown in Table 2. See these commands for more information. To define the name of a voice application and specify the location of the Tool Command Language (Tcl) or VoiceXML document to load for this application, use the call application voice command in global configuration mode. To remove the defined application and all configured parameters associated with it, use the no form of this command. call application voice application-name {location | av-pair} no call application voice application-name
Syntax Description
application-name
Character string that defines the name of the voice application.
location
Location of the Tcl script or VoiceXML document in URL format. Valid storage locations are TFTP, FTP, HTTP, and flash memory.
av-pair
Text string that defines attribute-value (AV) pairs specified by the Tcl script and understood by the RADIUS server. Multiple AV pairs can be enclosed in quotes; up to 512 entries are supported.
Command Default
None
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XH
This command was introduced.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T. The location argument was added.
12.1(3)T
The av-pair argument was added for AV pairs.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB
This command was modified to support VoiceXML applications and HTTP server locations on the Cisco AS5300, Cisco AS5350, and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T and implemented on the Cisco 1750.
12.2(4)XM
This command was implemented on the Cisco 1751. Support for other Cisco platforms is not included in this release.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice
Usage Guidelines
Release
Modification
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 2600 series, Cisco 3600 series, Cisco 3725, Cisco 3745, and Cisco 7200 series.
12.2(11)T
This command was implemented for VoiceXML applications. This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 in this release.
12.2(15)T
MCID AV-pairs were added for the av-pair argument; they are mcid-dtmf, mcid-release-timer, and mcid-retry-limit.
12.3(8)T
Support was added to allow up to 512 multiple AV pairs (enclosed in quotes) to be used in a single command.
12.3(14)T
This command was replaced. The call application voice command was replaced by the commands shown in Table 2.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
The call application voice command was replaced by the commands shown in Table 2. Table 2
call application voice Command Replacements
Command
Command Mode
Purpose
application
Global configuration Enters application configuration mode to configure voice applications and services.
service
Application configuration
Enters service configuration mode to configure a standalone application, such as a debit card script.
package
Application configuration
Use to load and configure a package. A package is a linkable set of C or Tcl functions that provide functionality invoked by applications or other packages.
param
Application parameter configuration
Use to configure parameters for services or packages.
Use this command when configuring interactive voice response (IVR) or one of the IVR-related features (such as Debit Card) to define the name of an application and to identify the location of the Tcl script or VoiceXML document associated with the application. A voice application must be configured by using this command before the application can be configured with the application command in a dial peer. Tcl scripts and VoiceXML documents can be stored in any of the following locations: on TFTP, FTP, or HTTP servers, in the flash memory of the gateway, or on the removable disks of the Cisco 3600 series. The audio files that they use can be stored in any of these locations, and on Real-Time Streaming Protocol (RTSP) servers. HTTP is the recommended protocol for loading applications and audio prompts because of its efficient design for loading information over the web. For example, it has methods for determining how long a file can be cached and whether a cached file is still valid. Include the file type extension in the filename (.vxml or .tcl) when specifying the document used by the application. Tcl files require the extension .tcl, and VoiceXML documents require .vxml.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice
Note
Examples
The no call application voice command causes all related call application commands—for instance, call application voice language and call application voice set-location—to be deleted. The no call application voice application-name command removes the entire application and all parameters, if configured.
Effective with Cisco IOS Release 12.3(14)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice Warning: This command has been deprecated. Please use the following: application service package param
The following example defines the fax-relay application and the TFTP server location of the associated Tcl script: call application voice fax-relay tftp://keyer/faxrelay.tcl
The following example defines the application “prepaid” and the TFTP server location of the associated Tcl script: call application voice prepaid tftp://keyer/debitcard.tcl
The following is an example of AV pair configuration: set avsend(h323-ivr-out,)) “payphone:true” set avsend(323-ivr-out,1) “creditTime:3400”
The AV pair (after the array is defined, as in the prior example) must be sent to the server using the authentication, authorization, and accounting (AAA) authenticate or AAA authorize verbs as follows: aaa authenticate $account $password $avsend
The script would use this AV pair whenever it is needed to convey information to the RADIUS server that cannot be represented by the standard vendor-specific attributes (VSAs). The following example shows how to define the VoiceXML application “vapptest1” and the flash memory location of the associated VoiceXML document “demo0.vxml”: call application voice vapptest1 flash:demo0.vxml
The following example specifies the MCID application name, the TFTP server location of the associated Tcl script, and the AV-pairs associated with the MCID application: call call call call
Cisco IOS Voice Commands: C call application voice
Command
Description
application (dial peer)
Defines the call application in the dial peer.
application (global configuation)
Enters application configuration mode to configure applications.
call application voice language
Defines the language of the audio file for the designated application and passes that information to the application.
call application voice load
Reloads the designated Tcl script or VoiceXML document.
call application voice pin-len
Defines the number of characters in the PIN for the application and passes that information to the application.
call application voice redirect-number
Defines the telephone number to which a call is redirected for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
call application voice security Sets the security level of a VoiceXML application to trusted so that trusted ANI is not blocked. call application voice set-location
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
call application voice uid-len
Defines the number of characters in the UID for the designated application and passes that information to the application.
call application voice warning-time
Defines, in seconds, how long in advance a user is warned before the allowed calling time expires for the designated application.
package
Enters application parameter configuration mode to load and configure a package.
param
Loads and configures parameters in a package or a service (application) on the gateway.
service
Loads and configures a specific, standalone application on a dial peer.
show call application voice
Displays information about voice applications.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice access-method
call application voice access-method Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice access-method command was replaced by the param access-method command in application parameter configuration mode. See the param access-method command for more information. To specify the access method for two-stage dialing for the designated application, use the call application voice access-method command in global configuration mode. To restore default values for this command, use the no form of this command. call application voice application-name access-method {prompt-user | redialer} no call application voice application-name access-method
Syntax Description
application-name
Name of the application.
prompt-user
Specifies that no DID is set in the incoming POTS dial peer and that a Tcl script in the incoming POTS dial peer is used for two-stage dialing.
redialer
Specifies that no DID is set in the incoming POTS dial peer and that the redialer device are used for two-stage dialing.
Command Default
Prompt-user (when DID is not set in the dial peer)
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300.
12.1(5)T
This command was integrated into the Cisco IOS Release 12.1(5)T.
12.2(4)T
This command was introduced on the Cisco 1750.
12.3(14)T
This command was replaced by the param access-method command in application parameter configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
Use the call application voice access-method command to specify the access method for two-stage dialing when DID is disabled in the POTS dial peer.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice access-method
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice access-method Warning: This command has been deprecated. Please use the following: param access-method
The following example specifies prompt-user as the access method for two-stage dialing for the app_libretto_onramp9 IVR application: call application voice app_libretto_onramp9 access-method prompt-user
Related Commands
Command
Description
call application voice
Loads a specified application onto the router from the TFTP server and gives it an application name by which it is known on the router.
call application voice language
Defines the language of the audio file for the designated application and passes that information to the application.
call application voice load
Reloads the designated Tcl script.
call application voice pin-len
Defines the number of characters in the PIN for the application and passes that information to the application.
call application voice redirect-number
Defines the telephone number to which a call is redirected—for example, the operator telephone number of the service provider—for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
call application voice set-location
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
call application voice uid-len
Defines the number of characters in the UID for the designated application and passes that information to the application.
call application voice warning-time
Defines, in seconds, how long in advance a user is warned before the allowed calling time expires for the designated application.
param access-method
Specifies the access method for two-stage dialing for the designated application.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice account-id-method
call application voice account-id-method Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice account-id-method command is replaced by the param account-id-method command in application parameter configuration mode. See the param account-id-method command for more information. To configure the fax detection IVR application to use a particular method to assign the account identifier, use the call application voice account-id-method command in global configuration mode. To remove configuration of this account identifier, use the no form of this command. call application voice application-name account-id-method {none | ani | dnis | gateway} no call application voice application-name account-id-method
Syntax Description
application-name
Name of the defined fax detection IVR application.
none
Account identifier is blank. This is the default.
ani
Account identifier is the calling party telephone number (automatic number identification, or ANI).
dnis
Account identifier is the dialed party telephone number (dialed number identification service, or DNIS).
gateway
Account identifier is a router-specific name derived from the hostname and domain name, displayed in the following format: router-name.domain-name.
Command Default
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.1(5)XM
This command was introduced for the Cisco AS5300.
12.2(2)XB
This command was implemented on the Cisco AS5400 and Cisco AS5350.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.2(11)T
This command was implemented on the Cisco AS5300, the Cisco AS5350, and Cisco AS5400.
12.3(14)T
This command was replaced by the param account-id-method command in application parameter configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice account-id-method
Usage Guidelines
When an on-ramp application converts a fax into an e-mail, the e-mail contains a field called x-account-id, which can be used for accounting or authentication. The x-account-id field can contain information supplied as a result of this command, such as the calling party’s telephone number (ani), the called party’s telephone number (dnis), or the name of the gateway (gateway). This command is not supported by Cisco IOS help; that is, if you type the call application voice fax_detect account-id-method command and a question mark (?), the Cisco IOS help does not supply a list of entries that are valid in place of the question mark.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application fax_detect account-id-method gateway Warning: This command has been deprecated. Please use the following: param account-id-method
The following example sets the fax detection IVR application account identifier to the router-specific name derived from the hostname and domain name: call application voice fax_detect account-id-method gateway
Related Commands
Command
Description
call application voice
Loads a specified IVR application onto the router from the TFTP server and gives it an application name by which it is known on the router.
call application voice fax-dtmf
Configures the fax detection IVR application to recognize a specified digit that indicates a fax call in default-voice and default-fax modes.
call application voice mode
Configures the fax detection IVR application to operate in one of its four modes.
call application voice prompt
Configures the fax detection IVR application to use the specified audio file as a user prompt in listen-first mode, default-voice mode, or default-fax mode.
call application voice voice-dtmf
Configures the fax detection IVR application to recognize a specified digit that indicate a voice call in default-voice and default-fax modes.
param account-id-method
Configures an application to use a particular method to assign the account identifier.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice authentication enable
call application voice authentication enable Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice authentication enable command is replaced by the param authentication enable command in application configuration mode. See the param authentication enable command for more information. To enable authentication, authorization, and accounting (AAA) services for a Tool Command Language (TCL) application, use the call application voice authentication enable command in global configuration mode. To disable authentication for a TCL application, use the no form of this command. call application voice application-name authentication enable no call application voice application-name authentication enable
Syntax Description
application-name
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
12.2(4)T
This command was implemented on the Cisco 1750.
12.2(8)T
This command was implemented on the Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.3(14)T
This command was replaced by the param authentication enable command in application configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
This command enables AAA authentication services for a TCL application if a AAA authentication method list has been defined using both the aaa authentication command and the call application voice authen-list command.
Cisco IOS Voice Command Reference
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Name of the application.
Cisco IOS Voice Commands: C call application voice authentication enable
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice authentication enable Warning: This command has been deprecated. Please use the following: param authentication enable
The following example enables a AAA authentication method list (called “sample”) to be used with outbound store-and-forward fax. call application voice app_eaample_onramp9 authen-list sample call application voice app_example_onramp9 authentication enable
Related Commands
Command
Description
aaa authentication
Enables AAA accounting of requested services when you use RADIUS or TACACS+.
call application voice authen-list
Specifies the name of an authentication method list for a TCL application.
call application voice authen-method
Specifies the authentication method for a TCL application.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice accounting-list
call application voice accounting-list Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice accounting-list command is replaced by the param accounting-list in application configuration mode. See the param accounting-list command for more information. To define the name of the accounting method list to be used for authentication, authorization, and accounting (AAA) with store-and-forward fax on a voice feature card (VFC), use the call application voice accounting-list command in global configuration mode. To undefine the accounting method list, use the no form of this command. call application voice application-name accounting-list method-list-name no call application voice application-name accounting-list method-list-name
Syntax Description
application-name
Name of the application.
method-list-name
Character string used to name a list of accounting methods to be used with store-and-forward fax.
Command Default
No AAA accounting method list is defined
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
12.2(4)T
This command was implemented on the Cisco 1750.
12.2(8)T
This command was implemented on the following platforms: Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.3(14)T
This command was replaced by the param accounting-list in application configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice accounting-list
Usage Guidelines
This command defines the name of the AAA accounting method list to be used with store-and-forward fax. The method list itself, which defines the type of accounting services provided for store-and-forward fax, is defined using the aaa accounting command. Unlike standard AAA (in which each defined method list can be applied to specific interfaces and lines), the AAA accounting method lists that are used in store-and-forward fax are applied globally. After the accounting method lists have been defined, they are enabled by using the mmoip aaa receive-accounting enable command. This command applies to both on-ramp and off-ramp store-and-forward fax functions on VFCs. The command is not used on modem cards.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice accounting-list Warning: This command has been deprecated. Please use the following: param accounting-list
The following example defines a AAA accounting method list “example” to be used with store-and-forward fax: aaa new-model call application voice app_libretto_onramp9 accounting-list example
Related Commands
Command
Description
aaa accounting
Enables AAA accounting of requested services when you use RADIUS or TACACS+.
call application voice accounting enable
Enables AAA accounting for a TCL application.
mmoip aaa receive-accounting enable
Enables on-ramp AAA accounting services.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice accounting-template
call application voice accounting-template Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice accounting-template command is obsolete. Use the call accounting-template command in application configuration mode to configure a voice accounting template. To configure T.37 fax accounting with VoIP authentication, authorization, and accounting (AAA) nonblocking Application Programming Interface (API), use the call application voice accounting-template command in global configuration mode. To remove the defined application and all configured parameters associated with it, use the no form of this command. call application voice application-name accounting-template template-name no call application voice application-name accounting-template template-name
Syntax Description
application-name
Defines the name of the T.37 voice application. •
template-name
Use the call application voice command to define the name of a voice application and specify the location of the Tool Command Language (Tcl) or VoiceXML document to load for this application.
Defines the name of the template. •
Use the call accounting-template voice command to define the template name.
Command Default
Disabled
Command Modes
Global configuration
Command History
Release
Modification
12.3(1)
This command was introduced.
12.3(14)T
This command is obsolete. Use the call accounting-template command in application configuration mode to configure a voice accounting template.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
This command enables T.37 fax to be consistent with VoIP AAA accounting services, which uses the Cisco IOS software nonblocking APIs. This command creates accounting templates for faxes by associating the template name with the T.37 onramp or offramp application. You can define an accounting template to specify information that is included in an accounting packet.
Note
This command applies only to T.37 fax.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice accounting-template
Use the show call active fax and the show call history fax commands to check the configuration.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice accounting-template Warning: This command has been deprecated. Please use the following: call accounting-template
The following is an example configuration using the T.37 accounting template: Router(config)# call application voice t37_onramp accounting-template sample-name Router(config)# call application voice t37_offramp accounting-template sample-name
Related Commands
Command
Description
application
Defines the call application in the dial peer.
call accounting-template
Selects an accounting template at a specific location.
call accounting-template voice
Selects an accounting template at a specific location.
call application voice
Defines the name of a voice application and specifies the location of the Tcl or VoiceXML document to load for this application.
call application voice language
Defines the language of the audio file for the designated application and passes that information to the application.
call application voice load
Reloads the designated Tcl script or VoiceXML document.
call application voice pin-len
Defines the number of characters in the PIN for the application and passes that information to the application.
call application voice redirect-number
Defines the telephone number to which a call is redirected for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
call application voice security trusted
Sets the security level of a VoiceXML application to trusted so that ANI is not blocked.
call application voice set-location
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
call application voice uid-len
Defines the number of characters in the UID for the designated application and passes that information to the application.
call application voice warning-time
Defines, in seconds, how long in advance a user is warned before the allowed calling time expires for the designated application.
show call active fax
Displays call information for fax transmissions in progress.
show call application voice Displays information about voice applications. show call history fax
Displays the call history table for fax transmissions.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice authen-list
call application voice authen-list Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice authen-list command was replaced by the param authen-list command in application configuration mode. See the param authen-list command for more information. To specify the name of an authentication method list for a Tool Command Language (Tcl) application, use the call application voice authen-list command in global configuration mode. To disable the authentication method list for a Tcl application, use the no form of this command. call application voice application-name authen-list method-list-name no call application voice application-name authen-list method-list-name
Syntax Description
application-name
Name of the application.
method-list-name
Character string used to name a list of authentication methods to be used with T.38 fax relay and T.37 store-and-forward fax.
Command Default
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
12.2(4)T
This command was implemented on the Cisco 1750.
12.2(8)T
This command was implemented on the Cisco 1751, Cisco 2600 series and Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.3(14)T
This command was replaced by the param authen-list command in application configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
This command defines the name of the authentication, authorization, and accounting (AAA) method list to be used with fax applications on voice feature cards. The method list itself, which defines the type of authentication services provided for store-and-forward fax, is defined using the aaa authentication command. Unlike standard AAA (in which each defined method list can be applied to specific interfaces and lines), AAA method lists that are used with fax applications are applied globally. After the authentication method lists have been defined, they are enabled by using the call application voice authentication enable command.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice authen-list
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice authen-list Warning: This command has been deprecated. Please use the following: param authen-list
The following example defines a AAA authentication method list (called “fax”) to be used with T.38 fax relay and T.37 store-and-forward fax: call application voice app_libretto_onramp9 authen-list fax
Related Commands
Command
Description
aaa authentication
Enable AAA accounting of requested services for billing or security purposes.
call application voice authen-method
Specifies the authentication method for a Tcl application.
call application voice authentication enable
Enables AAA authentication services for a Tcl application.
Cisco IOS Voice Command Reference
VR-193
Cisco IOS Voice Commands: C call application voice authen-method
call application voice authen-method Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice authen-method command is replaced by the param authen-method command in application configuration mode. See the param authen-method command for more information. To specify an authentication, authorization, and accounting (AAA) authentication method for a Tool Command Language (Tcl) application, use the call application voice authen-method command in global configuration mode. To disable the authentication method for a Tcl application, use the no form of this command. call application voice application-name authen-method {prompt-user | ani | dnis | gateway | redialer-id | redialer-dnis} no call application voice application-name authen-method {prompt-user | ani | dnis | gateway | redialer-id | redialer-dnis}
Syntax Description
application-name
Name of the application.
prompt-user
User is prompted for the Tcl application account identifier.
ani
Calling party telephone number (automatic number identification or ANI) is used as the Tcl application account identifier.
dnis
Called party telephone number (dialed number identification service or DNIS) is used as the Tcl application account identifier.
gateway
Router-specific name derived from the host name and domain name is used as the Tcl application account identifier, displayed in the following format: router-name.domain-name.
redialer-id
Account string returned by the external redialer device is used as the Tcl application account identifier. In this case, the redialer ID is either the redialer serial number or the redialer account number.
redialer-dnis
Called party telephone number (dialed number identification service or DNIS) is used as the Tcl application account identifier captured by the redialer if a redialer device is present.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
12.2(4)T
This command was implemented on the Cisco 1750.
Cisco IOS Voice Command Reference
VR-194
Cisco IOS Voice Commands: C call application voice authen-method
Release
Modification
12.2(8)T
This command was implemented on the Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.3(14)T
This command was replaced by the param authen-method command in application configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
Normally, when AAA is used for simple user authentication, AAA uses the username information defined in the user profile for authentication. With T.37 store-and-forward fax and T.38 real-time fax, you can specify that the ANI, DNIS, gateway ID, redialer ID, or redialer DNIS be used to identify the user for authentication or that the user be prompted for the Tcl application.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice authen-method Warning: This command has been deprecated. Please use the following: param authen-method
The following example configures the router-specific name derived from the hostname and domain name as he Tcl application account identifier for the app_sample_onramp9 Tcl application: call application voice app_sample_onramp9 authen-method gateway
Related Commands
Command
Description
call application voice authentication enable
Enables AAA authentication services for a Tcl application.
call application voice authen-list
Specifies the name of an authentication method list for a Tcl application.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice accounting enable
call application voice accounting enable Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice accounting enable command is replaced by the param accounting enable command in application configuration mode. See the param accounting enable command for more information. To enable authentication, authorization, and accounting (AAA) accounting for a Tool Command Language (Tcl) application, use the call application voice accounting enable command in global configuration mode. To disable accounting for a Tcl application, use the no form of this command. call application voice application-name accounting enable no call application voice application-name accounting enable
Syntax Description
application-name
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
12.2(4)T
This command was implemented on the Cisco 1750.
12.2(8)T
This command was implemented on the Cisco 1751, Cisco 2600 series and Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.3(14)T
This command was replaced by the param accounting enable command in application configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
Name of the application.
This command enables AAA accounting services if a AAA accounting method list has been defined using both the aaa accounting command and the mmoip aaa method fax accounting command. This command applies to off-ramp store-and-forward fax functions.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice accounting enable
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice accounting enable Warning: This command has been deprecated. Please use the following: param accounting enable
The following example enables AAA accounting to be used with outbound store-and-forward fax: call application voice app_libretto_onramp9 accounting enable
Related Commands
Command
Description
aaa accounting
Enables AAA accounting of requested services when you use RADIUS or TACACS+.
mmoip aaa method fax accounting
Defines the name of the method list to be used for AAA accounting with store-and-forward fax.
Cisco IOS Voice Command Reference
VR-197
Cisco IOS Voice Commands: C call application voice default disc-prog-ind-at-connect
Effective with Cisco IOS Release 12.3(14)T, the call application voice default disc-prog-ind-at-connect command is replaced. Use one of the following commands: •
To convert a DISCONNECT message with Progress Indicator set to PROG_INBAND (PI=8) to a regular DISCONNECT message when the call is in the active state, use the call application voice default disc-prog-ind-at-connect command in global configuration mode. To revert to a DISCONNECT message with Progress Indicator set to PROG_INBAND (PI=8) when the call is in the active state, use the no form of this command. call application voice default disc-prog-ind-at-connect [1 | 0] no call application voice default disc-prog-ind-at-connect [1 | 0]
Syntax Description
1
(Optional) Convert a DISCONNECT message with Progress Indicator set to PROG_INBAND (PI=8) to a regular DISCONNECT message when the call is in the active state.
0
(Optional) Revert to a DISCONNECT message with Progress Indicator set to PROG_INBAND (PI=8) when the call is in the active state.
Command Default
The DISCONNECT message has Progress Indicator set to PROG_INBAND (PI=8) when the call is in the active state.
Command Modes
Global configuration
Command History
Release
Modification
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
12.3(14)T
The call application voice default disc-prog-ind-at-connect command was replaced. Use one of the following commands:
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Cisco IOS Voice Commands: C call application voice default disc-prog-ind-at-connect
Usage Guidelines
This command has no effect if the call is not in the active state. This command is available for the default voice application. It may not be available when using some Tcl IVR applications. The Cisco IOS command-line interface command completion and help features do not work with this command.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice default disc-prog-ind-at-connect Warning: This command has been deprecated. Please use the following: param convert-discpi-after-connect paramspace session_xwork convert-discpi-after-connec
In the following example, a DISCONNECT message with Progress Indicator set to PROG_INBAND (PI=8) is converted to a regular DISCONNECT message when the call is in the active state: call application voice default disc-prog-ind-at-connect 1
Related Commands
Command
Description
param convert-discpi-after-connect
Enables or disables conversion of a DISCONNECT message with Progress Indicator set to PROG_INBAND (PI=8) to a regular DISCONNECT message when the call is in the active state.
Enables or disables conversion of a DISCONNECT message with Progress Indicator set to PROG_INBAND (PI=8) to a regular DISCONNECT message when the call is in the active state.
Cisco IOS Voice Command Reference
VR-199
Cisco IOS Voice Commands: C call application voice dsn-script
call application voice dsn-script Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice dsn-script command is replaced by the param dsn-script command in application parameter configuration mode. To specify the VoiceXML application to which the off-ramp mail application hands off calls for off-ramp delivery status notification (DSN) and message disposition notification (MDN) e-mail messages, use the call application voice dsn-script command in global configuration mode. To remove the application, use the no form of this command. call application voice mail-application-name dsn-script application-name no call application voice mail-application-name dsn-script application-name
Syntax Description
mail-application-name
Name of the off-ramp mail application that launches the app_voicemail_offramp.tcl script when the gateway receives an e-mail trigger.
application-name
Name of the VoiceXML application to which the off-ramp mail application hands off the call when the destination answers.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.2(11)T
This command was introduced on the Cisco 3640, Cisco 3660, Cisco AS5300, Cisco AS5350, and Cisco AS5400.
12.3(14)T
This command was replaced by the param dsn-script command in application parameter configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
When the off-ramp gateway receives a DSN or MDN e-mail message, it handles it in the same way as a voice e-mail trigger message. The dial peer is selected on the basis of dialed number identification service (DNIS), and the mail application hands off the call to the VoiceXML application that is configured with this command.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice dsn-script
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice dsn-script Warning: This command has been deprecated. Please use the following: param dsn-script
The following example shows how to define the DSN application and how to apply it to a dial peer: call application voice offramp-mapp tftp://sample/tftp-users/tcl/app_voicemail_offramp.tcl call application voice dsn-mapp-test tftp://sample/tftp-users/vxml/dsn-mapp-test.vxml call application voice offramp-mapp dsn-script dsn-mapp-test ! dial-peer voice 1000 mmoip application offramp-mapp incoming called-number 555.... information-type voice
Related Commands
Command
Description
application
Defines a specific voice application in the dial peer.
call application voice
Defines the name of a voice application and specifies the location of the document (Tcl or VoiceXML) to load for the application.
param dsn-script
Specifies the VoiceXML application to which the off-ramp mail application hands off calls for off-ramp DSN and MDN e-mail messages.
show call application voice
Displays information about the configured voice applications.
Cisco IOS Voice Command Reference
VR-201
Cisco IOS Voice Commands: C call application voice event-log
call application voice event-log Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice event-log is obsolete. To enable event logging for a specific voice application, use one of the following commands: •
paramspace appcommon event-log (service configuration mod
To enable event logging for a specific voice application, use the call application voice event-log command in global configuration mode. To reset to the default, use the no form of this command. call application voice application-name event-log [disable] no call application voice application-name event-log
Syntax Description
application-name
Name of the voice application.
disable
(Optional) Disables event logging for the named application.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.3(14)T
The call application voice event-log is obsolete. To enable event logging for a specific voice application, use one of the following commands:
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Cisco IOS Voice Commands: C call application voice event-log
Usage Guidelines
This command is application-specific; it takes precedence over the global configuration command, call application event-log, which enables event logging for all voice applications. Before you can use this command, you must configure the named application on the gateway by using the call application voice command.
Note
Examples
To prevent event logging from adversely impacting system resources for production traffic, the gateway uses a throttling mechanism. When free processor memory drops below 20 percent, the gateway automatically disables all event logging. It resumes event logging when free memory rises above 30 percent. While throttling is occurring, the gateway does not capture any new event logs even if event logging is enabled. You should monitor free memory and enable event logging only when necessary for isolating faults.
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call call application voice event-log Warning: This command has been deprecated. Please use the following: param event-log paramspace appcommon event-log
The following example enables event logging for all instances of the application named sample_app: call application voice sample_app event-log
The following example enables event logging for all applications except the application sample_app: call application event-log call application voice sample_app event-log disable
Related Commands
Command
Description
call application event-log
Enables event logging for voice application instances.
call application event-log max-buffer-size
Sets the maximum size of the event log buffer for each application instance.
call application voice
Defines the name of a voice application and specifies the location of the script to load for the application.
monitor call application event-log
Displays the event log for an active application instance in real-time.
param event-log
Enables or disables logging for linkable Tcl functions (packages).
paramspace appcommon event-log
Enable or disables logging for a service (application).
show call application session-level
Displays event logs and statistics for voice application instances.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice fax-dtmf
call application voice fax-dtmf Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice fax-dtmf command is replaced by the param fax-dtmf command in application parameter configuration mode. See the param fax-dtmf command for more information. To direct the fax detection interactive voice response (IVR) application to recognize a specified digit to indicate a fax call in default-voice and default-fax modes, use the call application voice fax-dtmf command in global configuration mode. To remove configuration of this digit, use the no form of this command. call application voice application-name fax-dtmf {0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 | * | #} no call application voice application-name fax-dtmf {0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 | * | #}
Syntax Description
application-name
The name of the fax detection IVR application that you defined when you loaded the application on the router.
0|1|2|3|4|5|6| 7|8|9|*|#
The telephone keypad digit processed by the calling party to indicate a fax call, in response to the audio prompt that plays during the default-voice or default-fax mode of the fax detection IVR application.
Command Default
2
Command Modes
Global configuration
Command History
Release
Modification
12.1(5)XM
This command was introduced for the Cisco AS5300.
12.2(2)T
This command was integrated into Cisco IOS Release 12.2(2)T.
12.2(2)XB
This command was implemented on the Cisco AS5400 and Cisco AS5350.
12.2(8)T
This command was implemented on the Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.2(11)T
This command was implemented on the Cisco AS5300, Cisco AS5350, and Cisco AS5400.
12.3(14)T
This command was replaced by the param fax-dtmf command in application parameter configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice fax-dtmf
Usage Guidelines
This command is useful only when the fax detection IVR application is being configured in default-voice mode or default-fax mode as defined by the call application voice mode command. Only one digit can be specified in this command, and that digit must be different from the digit specified in the call application voice voice-dtmf command. You are not notified immediately if you make the error of configuring them both to the same digit. To find this error, you must start the debugging with the debug voip ivr script command and then observe some failing calls. This command is not supported by Cisco IOS help; that is, if you type call application voice fax_detect fax-dtmf and a question mark (?), Cisco IOS help does not supply a list of entries that are valid in place of the question mark.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice fax-dtmf Warning: This command has been deprecated. Please use the following: param fax-dtmf
The following example selects DTMF digit 1 to indicate a fax call: call application voice fax_detect script_url call application voice fax_detect fax-dtmf 1 dial-peer voice 302 pots application fax_detect
Related Commands
Command
Description
call application voice
Loads an IVR application onto a router and gives it an application name.
call application voice account-id-method
Configures the fax detection IVR application to use a particular method to assign the account identifier.
call application voice mode
Configures the fax detection IVR application to operate in one of its four modes.
call application voice prompt
Configures the fax detection IVR application to use the specified audio file as a user prompt.
call application voice voice-dtmf
Configures the fax detection IVR application to recognize the specified digit to indicate a voice call.
debug voip ivr script
Displays debug information from the fax detection IVR script.
param fax-dtmf
Directs the fax detection IVR application to recognize a specified digit to indicate a fax call in default-voice and default-fax modes.
Cisco IOS Voice Command Reference
VR-205
Cisco IOS Voice Commands: C call application voice global-password
call application voice global-password Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice global-password command is replaced by the param global-password command in application parameter configuration mode. See the param global-password command for more information. To define a password to be used with CiscoSecure for Windows NT when using store-and-forward fax on a voice feature card, use the call application voice global-password command in global configuration mode. To restore the default value, use the no form of this command. call application voice application-name global-password password no call application voice application-name global-password password
Syntax Description
application-name
The name of the application.
password
Character string used to define the CiscoSecure for Windows NT password to be used with store-and-forward fax. The maximum length is 64 alphanumeric characters.
Command Default
No password is defined
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)XI
This command was introduced on the Cisco AS5300.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
12.3(14)T
This command is replaced by the param global-password command in application parameter configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
CiscoSecure for Windows NT might require a separate password to complete authentication, no matter what security protocol you use. This command defines the password to be used with CiscoSecure for Windows NT. All records on the Windows NT server use this defined password. This command applies to on-ramp store-and-forward fax functions on Cisco AS5300 universal access server voice feature cards. It is not used on modem cards.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice global-password
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice global-password Warning: This command has been deprecated. Please use the following: param global-password
The following example shows a password (abercrombie) being used by AAA for the app_sample_onramp9 Tcl application: call application voice app_sample_onramp9 global-password abercrombie
Related Commands
Command
Description
param global-password
Defines a password to be used with CiscoSecure for Windows NT when using store-and-forward fax on a voice feature card.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice language
call application voice language Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice language is replaced by the following commands: •
param language (application parameter configuration mode)
•
paramspace language (service configuration mode)
See these commands for more information.
To specify the language for dynamic prompts used by an interactive voice response (IVR) application (Tool Command Language (Tcl) or VoiceXML), use the call application voice language command in global configuration mode. To remove this language specification from the application, use the no form of this command. call application voice application-name language digit language no call application voice application-name language digit language
Syntax Description
application-name
Name of the application to which the language parameters are being passed.
digit
Number that identifies the language used by the audio files. Any number can represent any language. Enter 1 to indicate the primary language and 2 to indicate the secondary language. Range is from 0 to 9.
language
Two-character code that identifies the language of the associated audio files. Valid entries are as follows: •
en—English
•
sp—Spanish
•
ch—Mandarin
•
aa—all
Command Default
If this command is not configured, the default language is English.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB
This command was modified to support VoiceXML applications on the Cisco AS5300, Cisco AS5350, and Cisco AS5400.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice language
Release
Modification
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)T
This command was implemented on the Cisco 1750.
12.2(4)XM
This command was implemented on the Cisco 1751.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800 and Cisco AS5850 is not included in this release.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T for VoiceXML applications. This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco 5800, and Cisco AS5850 in this release.
12.3(14)T
The call application voice language was replaced by the following commands: param language (application parameter configuration mode) paramspace language (service configuration mode)
12.4(24)T
Usage Guidelines
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
This command identifies the number that users enter for a language; for example, “Enter 1 for English. Enter 2 for French.” This number is used only with the Tcl IVR Debit Card feature. Although it is not used by VoiceXML, you still must enter a number from 0 to 9. Instead of using this command, you can configure the language and location of the prerecorded audio files within a Tcl script or VoiceXML document. For more information, see the Tcl IVR API Version 2.0 Programmer’s Guide or Cisco VoiceXML Programmer’s Guide, respectively. To identify the location of the language audio files that are used for the dynamic prompts, use the call application voice set-location command. Tcl scripts and VoiceXML documents can be stored in any of the following locations: On the TFTP, FTP, or HTTP servers, in the flash memory of the gateway, or on the removable disks of the Cisco 3600 series. The audio files that they use can be stored in any of these locations, and on RTSP servers. With the Pre-Paid Debitcard Multi-Language feature, you can create Tcl scripts and a two-character code for any language. See the Cisco Pre-Paid Debitcard Multi-Language Programmer's Reference. With the multilanguage support for Cisco IOS IVR, you can create a Tcl language module for any language and any set of TTS notations for use with Tcl and VoiceXML applications. See the Enhanced Multi-Language Support for Cisco IOS Interactive Voice Response document.
Cisco IOS Voice Command Reference
VR-209
Cisco IOS Voice Commands: C call application voice language
Table 3 lists Tcl script names and the corresponding commands that are required for each Tcl script. Table 3
Tcl Scripts and Commands
Tcl Script Name
Description
Commands to Configure
app_libretto_onramp9.tcl
Authenticates the account and personal identification number (PIN) using the following: prompt-user, using automatic number identification (ANI), dialed number identification service (DNIS), gateway ID, redialer ID, and redialer DNIS.
None
app_libretto_offramp5.tcl
Authenticates the account and PIN using the following: envelope-from, envelope-to, gateway ID, and x-account ID.
None
clid_4digits_npw_3_cli.tcl
This script authenticates the account number and PIN, respectively, using ANI and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, the script allows the caller to retry. The retry number is also configured through the CLI.
call application voice uid-length Range is 1 to 20. The default is 10. call application voice pin-length Range is 0 to 10. The default is 4. call application voice retry-count Range is 1 to 5. The default is 3.
clid_authen_col_npw_cli.tcl
This script authenticates the account number and PIN, call application voice respectively, using ANI and NULL. If the authentication retry-count fails, it allows the caller to retry. The retry number is Range is 1 to 5. The default is 3. configured through CLI. The account number and PIN are collected separately.
clid_authen_collect_cli.tcl
This script authenticates the account number and PIN using ANI and DNIS. If the authentication fails, the script allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
clid_col_npw_3_cli.tcl
This script authenticates using ANI and NULL for call application voice account numbers and PINs, respectively. If the retry-count authentication fails, it allows the caller to retry. The retry Range is 1 to 5. The default is 3. number is configured through the CLI.
clid_col_npw_npw_cli.tcl
This script authenticates using ANI and NULL for call application voice account and PIN, respectively. If authentication fails, it retry-count allows the caller to retry. The retry number is configured Range is 1 to 5. The default is 3. through the CLI. The account number and PIN are collected together.
fax_rollover_on_busy.tcl
Used for on-ramp T.38 fax rollover to T.37 fax when the voice hunt user-busy destination fax line is busy.
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call application voice retry-count Range is 1 to 5. The default is 3.
Cisco IOS Voice Commands: C call application voice language
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice language Warning: This command has been deprecated. Please use the following: param language paramspace language
The following example shows how to define the application “prepaid” and then selects English and Spanish as the languages of the audio files that are associated with the application: call application voice prepaid tftp://keyer/debitcard.tcl call application voice prepaid language 1 en call application voice prepaid language 2 sp
Related Commands
Command
Description
call application voice
Specifies the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
call application voice load
Reloads the designated Tcl script.
call application voice pin-len Defines the number of characters in the PIN for the application and passes that information to the application. call application voice redirect-number
Specifies the telephone number to which a call is redirected for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
call application voice set-location
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
call application voice uid-len Defines the number of characters in the UID for the designated application and passes that information to the application. call application voice warning-time
Defines, in seconds, how long in advance a user is warned before the allowed calling time expires for the designated application.
param language
Configures the language parameter in a service or package on the gateway.
paramspace language
Defines the category and location of audio files that are used for dynamic prompts by an IVR application (Tcl or VoiceXML).
show call application voice
Displays information about voice applications.
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Cisco IOS Voice Commands: C call application voice load
call application voice load To reload the selected voice application script after it has been modified, use the call application voice load command in privileged EXEC mode. This command does not have a no form. call application voice load application-name
Syntax Description
application-name
Command Default
No default behavior or values
Command Modes
Privileged EXEC
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 series and Cisco 3600 series (except for the Cisco 3660), and on the Cisco AS5300.
12.1(3)T
Support for dynamic script loading of Media Gateway Control Protocol (MGCP) was added.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB
This command was modified to support VoiceXML applications.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1751.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T and implemented on the Cisco 1750.
12.2(8)T
This command and implemented on the Cisco 7200 series.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T for VoiceXML applications. This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and the Cisco AS5850 in this release.
Usage Guidelines
Name of the Tcl or VoiceXML application to reload.
Use this command to reload an application Tcl script or VoiceXML document onto the gateway after it has been modified. The location of the Tcl script or VoiceXML document for the specified application must have already been configured using the call application voice command. Do not include the file type extension in the filename (.vxml or .tcl) when specifying the document used by the application. Tcl scripts and VoiceXML documents can be stored in any of the following locations: on TFTP, FTP, or HTTP servers, in the flash memory of the gateway, or on the removable disks of the Cisco 3600 series. The audio files that they use can be stored on any of these locations, and on RTSP servers.
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Cisco IOS Voice Commands: C call application voice load
Before Cisco IOS Release 12.1(3)T, the software checked the signature in a Tcl script to ensure that it was supported by Cisco. A signature on Tcl scripts is no longer required. A signature has never been required for VoiceXML documents. A Tcl script or VoiceXML document cannot be reloaded if it has active calls. Use the show call application voice command to verify that no active calls are using this application.
Tip
If the call application voice load command fails to load the Tcl script or VoiceXML document that is associated with the application, enable the debug voip ivr command and retry. This debugging command can provide information on why loading fails.
Note
MGCP scripting is not supported on the Cisco 1750 router or on Cisco 7200 series routers.
Examples
The following example shows the loading of a Tcl script called “clid_4digits_npw_3.tcl”: call application voice load clid_4digits_npw_3.tcl
The following example shows how to reload the VoiceXML application called “vapptest”: call application voice load vapptest
Related Commands
Command
Description
call application cache reload time
Configures the interval for reloading MGCP scripts.
call application voice
Creates and calls the application that interacts with the IVR feature.
debug http client
Displays information about the load an application that was loaded with HTTP.
show call application voice Displays a list of the voice applications that are configured.
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Cisco IOS Voice Commands: C call application voice mail-script
call application voice mail-script Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice mail-script command is replaced by the param mail-script command in application parameter configuration mode. See the param mail-script command for more information. To specify the VoiceXML application to which the off-ramp mail application hands off a call when the destination telephone answers, use the call application voice mail-script command in global configuration mode. To remove the application, use the no form of this command. call application voice mail-application-name mail-script application-name no call application voice mail-application-name mail-script application-name
Syntax Description
mail-application-name
Name of the off-ramp mail application that launches the app_voicemail_offramp.tcl script when the gateway receives an e-mail trigger.
application-name
Name of the VoiceXML application to which the off-ramp mail application hands off the call when the destination answers.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.2(11)T
This command was introduced on the Cisco 3640, Cisco 3660, Cisco AS5300, Cisco AS5350, and Cisco AS5400.
12.3(14)T
This command was replaced by the param mail-script command in application parameter configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
To load the mail application onto the gateway, use the call application voice command. The off-ramp mail application must be configured in the Multimedia Mail over Internet Protocol (MMoIP) dial peer that matches the telephone number contained in the header of the incoming e-mail message. The off-ramp mail application must use the Tool Command Language (Tcl) script named “app_voicemail_offramp.tcl” that is provided by Cisco. This Tcl script can be downloaded from the Cisco website by following this path: Cisco > Technical Support Help - TAC > Select & Download Software > Software Center > Access Software > TclWare.
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Cisco IOS Voice Commands: C call application voice mail-script
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice mail-script Warning: This command has been deprecated. Please use the following: param mail-script
The following example shows that the off-ramp mail application named “offramp-mapp” hands calls to the application named “mapp-test” if the telephone number in the e-mail header is seven digits beginning with 555: call application voice offramp-mapp tftp://sample/tftp-users/tcl/app_voicemail_offramp.tcl call application voice mapp-test tftp://sample/tftp-users/vxml/user-test.vxml call application voice offramp-mapp mail-script mapp-test ! dial-peer voice 1001 mmoip application offramp-mapp incoming called-number 555.... information-type voice
Related Commands
Command
Description
application
Defines a specific voice application in the dial peer.
call application voice
Defines the name of a voice application and specifies the location of the document (Tcl or VoiceXML) to load for the application.
param mail-script
Specifies the VoiceXML application to which the off-ramp mail application hands off a call when the destination telephone answers.
show call application voice
Displays information about the configured voice applications.
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Cisco IOS Voice Commands: C call application voice mode
call application voice mode Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice mode command is replaced by the param mode command in application parameter configuration mode. See the param mode command for more information. To direct the fax detection interactive voice response (IVR) application to operate in one of its four connection modes, use the call application voice mode command in global configuration mode. To return to the default connection mode, use the no form of this command. call application voice application-name mode {connect-first | listen-first | default-voice | default-fax} no call application voice application-name mode {connect-first | listen-first | default-voice | default-fax}
Syntax Description
application-name
Fax detection IVR application that was defined when the application was loaded on the router.
connect-first
Incoming calls are connected to the Real-Time Streaming Protocol (RTSP) server. This is the default.
listen-first
The gateway listens to the call first and then connects to the RTSP server. Any Dual tone multifrequency (DTMF) tones take the call to the voice server, but subsequent DTMF is forwarded as configured.
default-voice
Incoming calls are connected as voice calls to the RTSP server.
default-fax
Incoming calls are connected to the fax relay or store-and-forward fax application that is configured on the gateway.
Command Default
connect-first
Command Modes
Global configuration
Command History
Release
Modification
12.1(5)XM
This command was introduced on the Cisco AS5300.
12.2(2)XB
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice mode
Usage Guidelines
Release
Modification
12.3(14)T
This command was replaced by the param mode command in application parameter configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
The call application voice mode commands control the way that the gateway handles fax detection IVR applications calls. When the connect-first keyword is selected and CNG (calling) tones from the originating fax machine are detected, the voice application is disconnected and the call is passed to the configured fax application. If the listen-first keyword is selected, the gateway listens for CNG and, if it is detected, passes the call to the fax relay or store-and-forward fax application, whichever is configured on the gateway. When the default-voice and default-fax keywords are selected, the gateway defaults to voice after listening for CNG or passes the call to the fax relay or store-and-forward fax application, whichever was configured on the gateway. If the gateway hears the Dual tone multifrequency (DTMF) tones that are specified in the call application voice voice-dtmf or call application voice fax-dtmf commands, the call is forwarded as appropriate. Note that in all four connection modes, the router continues to listen for CNG throughout the call, even if the call has been connected to the voice server; if CNG is detected, the call is connected to fax relay or store-and-forward fax, whichever has been configured. This command is not supported by Cisco IOS help. If you type the call application voice fax_detect mode command and a question mark (?), Cisco IOS help does not supply a list of valid entries in place of the question mark.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice mode Warning: This command has been deprecated. Please use the following: param mode
The following example shows a selection of default-voice mode for the fax detection application: call application voice fax_detect script_url call application voice fax_detect mode default-voice dial-peer voice 302 pots application fax_detect
Related Commands
Command
Description
call application voice
Loads a specified IVR application onto the router from the TFTP server and gives it an application name by which it is known on the router.
call application voice account-id-method
Configures the fax detection IVR application to use a particular method to assign the account identifier.
call application voice fax-dtmf
Configures the fax detection IVR application to recognize a specified digit to indicate a fax call.
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Cisco IOS Voice Commands: C call application voice mode
Command
Description
call application voice prompt
Configures the fax detection IVR application to use the specified audio file as a user prompt in listen-first mode, default-voice mode, or default-fax mode.
call application voice voice-dtmf
Configures the fax detection IVR application to recognize a specified digit to indicate a voice call.
param mode
Configures the call transfer mode for a package.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice pin-len
call application voice pin-len Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice pin-len command is replaced with the param pin-len command in application parameter configuration mode. See the param pin-len command for more information. To define the number of characters in the personal identification number (PIN) for the designated application, use the call application voice pin-len command in global configuration mode. To disable the PIN for the designated application, use the no form of this command. call application voice application-name pin-len number no call application voice application-name pin-len number
Syntax Description
application-name
Application name to which the PIN length parameter is being passed.
number
Number of allowable characters in PINs associated with the specified application. Range is from 0 to 10. The default is 4.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T and implemented on the Cisco 1750.
12.2(4)XM
This command was implemented on the Cisco 1751.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series.
12.2(11)T
This command is supported on the Cisco AS5350, Cisco AS5400 Cisco AS5800, and the Cisco AS5850 in this release.
12.3(14)T
The call application voice pin-len command was replaced with the param pin-len command in application parameter configuration mode.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice pin-len
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the Tool Command Language (Tcl) script being used—or one of the IVR-related features (such as Debit Card) to define the number of allowable characters in a PIN for the specified application and to pass that information to the specified application. Table 4 lists Tcl script names and the corresponding commands that are required for each Tcl script.
Table 4
Tcl Scripts and Commands
Tcl Script Name
Description
Commands to Configure
app_libretto_onramp9.tcl
Authenticates the account and personal identification number (PIN) using the following: prompt-user, using automatic number identification (ANI), dialed number identification service (DNIS), gateway ID, redialer ID, and redialer DNIS.
None
app_libretto_offramp5.tcl
Authenticates the account and PIN using the following: envelope-from, envelope-to, gateway ID, and x-account ID.
None
clid_4digits_npw_3_cli.tcl
This script authenticates the account number and PIN, respectively, using ANI and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, the script allows the caller to retry. The retry number is also configured through the CLI.
call application voice uid-length Range is 1 to 20. The default is 10. call application voice pin-length Range is 0 to 10. The default is 4 call application voice retry-count Range is 1 to 5. The default is 3.
clid_authen_col_npw_cli.tcl This script authenticates the account number and PIN, call application voice retry-count respectively, using ANI and NULL. If the authentication Range is 1 to 5. The default is 3. fails, it allows the caller to retry. The retry number is configured through CLI. The account number and PIN are collected separately. clid_authen_collect_cli.tcl
This script authenticates the account number and PIN call application voice retry-count using ANI and DNIS. If the authentication fails, the script Range is 1 to 5. The default is 3. allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
clid_col_npw_3_cli.tcl
This script authenticates using ANI and NULL for call application voice retry-count account numbers and PINs, respectively. If the Range is 1 to 5. The default is 3. authentication fails, it allows the caller to retry. The retry number is configured through the CLI.
clid_col_npw_npw_cli.tcl
This script authenticates using ANI and NULL for call application voice retry-count account and PIN, respectively. If authentication fails, it Range is 1 to 5. The default is 3. allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected together.
fax_rollover_on_busy.tcl
Used for on-ramp T.38 fax rollover to T.37 fax when the voice hunt user-busy destination fax line is busy.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice pin-len
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice pin-len Warning: This command has been deprecated. Please use the following: param pin-len
The following example shows how to define a PIN length of 4 characters for the application named “prepaid”: call application voice prepaid pin-len 4
Related Commands
Command
Description
call application voice
Specifies the name to be used for an application and indicates the location of the appropriate IVR script to be used with the application.
call application voice language
Specifies the language of the audio file for the designated application and passes that information to the application.
call application voice load
Reloads the designated Tcl script.
call application voice redirect-number
Specifies the telephone number to which a call is redirected for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
call application voice set-location
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
call application voice uid-len
Defines the number of characters in the UID for the designated application and passes that information to the application.
call application voice warning-time
Defines, in seconds, how long in advance a user is warned before the allowed calling time expires for the designated application.
param pin-len
Defines the number of characters in the PIN for an application.
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Cisco IOS Voice Commands: C call application voice prompt
call application voice prompt Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice prompt command is replaced by the param prompt command. See the param prompt command for more information. To direct the fax detection interactive voice response (IVR) application to use the specified audio file as a user prompt, use the call application voice prompt command in global configuration mode. To disable use of this audio file, use the no form of this command. call application voice application-name prompt prompt-url no call application voice application-name prompt prompt-url
Syntax Description
application-name
Name of the fax detection IVR application that you defined when you loaded the application on the router.
prompt-url
URL or Cisco IOS file system location on the TFTP server for the audio file containing the prompt for the application.
Command Default
The prompt space is empty and no prompt is played.
Command Modes
Global configuration
Command History
Release
Modification
12.1(5)XM
This command was introduced for the Cisco AS5300.
12.2(2)XB
This command was implemented on the Cisco AS5400 and Cisco AS5350.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.2(11)T
This command was implemented on the Cisco AS5300, Cisco AS5350, and Cisco AS5400.
12.3(14)T
This command was replaced by the param prompt command.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
This command is useful only in the listen-first, default-voice, and default-fax modes of the fax detection application. Audio files should be a minimum of 9 seconds long so that callers do not hear silence during the initial CNG detection period. Any .au file can be used; formats are described in the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2.
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Cisco IOS Voice Commands: C call application voice prompt
This command is not supported by Cisco IOS help. If you type the call application voice fax_detect prompt command with a question (?), the Cisco IOS help does not supply a list of entries that are valid in place of the question mark.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice prompt Warning: This command has been deprecated. Please use the following: param prompt
The following example associates the audio file” promptfile.au” with the application file “fax_detect”, and the application with the inbound POTS dial peer: call application voice fax_detect script_url call application voice fax_detect mode default-voice call application voice fax_detect prompt promptfile.au dial-peer voice 302 pots application fax_detect
Related Commands
Command
Description
call application voice
Loads a specified IVR application onto the router from the TFTP server and gives it an application name by which it is known on the router.
call application voice account-id-method
Configures the fax detection IVR application to use a particular method to assign the account identifier.
call application voice fax-dtmf Configures the fax detection IVR application to recognize a specified digit to indicate a fax call. call application voice mode
Configures the fax detection IVR application to operate in one of its four modes.
call application voice voice-dtmf
Configures the fax detection IVR application to recognize a specified digit to indicate a voice call.
param prompt
Directs the fax detection IVR application to use the specified audio file as a user prompt.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice redirect-number
call application voice redirect-number Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice redirect-number command is replaced with the param redirect-number command in application parameter configuration mode. See the the param redirect-number command for more information. To define the telephone number to which a call is redirected—for example, the operator telephone number of the service provider—for the designated application, use the call application voice redirect-number command in global configuration mode. To cancel the redirect telephone number, use the no form of this command. call application voice application-name redirect-number number no call application voice application-name redirect-number number
Syntax Description
application-name
Name of the application to which the redirect telephone number parameter is being passed.
number
Designated operator telephone number of the service provider (or any other number designated by the customer). This is the number where calls are terminated when, for example, allowed debit time has run out or the debit amount is exceeded.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 series, the Cisco 3600 series, and the Cisco AS5300.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1751.
12.2(4)T
This command was implemented on the Cisco 1750.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T. This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 in this release.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice redirect-number
Usage Guidelines
Release
Modification
12.3(14)T
This command was replaced by the param redirect-number.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Use this command when configuring interactive voice response (IVR)—depending on the Tool Command Language (Tcl) script being used—or one of the IVR-related features (such as Debit Card) to define the telephone number to which a call is redirected. Table 5 lists Tcl script names and the corresponding commands that are required for each Tcl script.
Table 5
Tcl Scripts and Commands
Tcl Script Name
Description
Commands to Configure
app_libretto_onramp9.tcl
Authenticates the account and personal identification number (PIN) using the following: prompt-user, using automatic number identification (ANI), dialed number identification service (DNIS), gateway ID, redialer ID, and redialer DNIS.
None
app_libretto_offramp5.tcl
Authenticates the account and PIN using the following: envelope-from, envelope-to, gateway ID, and x-account ID.
None
clid_4digits_npw_3_cli.tcl
This script authenticates the account number and PIN, respectively, using ANI and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, the script allows the caller to retry. The retry number is also configured through the CLI.
call application voice uid-length Range is 1 to 20. The default is 10. call application voice pin-length Range is 0 to 10. The default is 4. call application voice retry-count Range is 1 to 5. The default is 3.
clid_authen_col_npw_cli.tcl This script authenticates the account number and PIN, call application voice respectively, using ANI and NULL. If the authentication retry-count fails, it allows the caller to retry. The retry number is Range is 1 to 5. The default is 3. configured through the CLI. The account number and PIN are collected separately. clid_authen_collect_cli.tcl
This script authenticates the account number and PIN call application voice using ANI and DNIS. If the authentication fails, the script retry-count allows the caller to retry. The retry number is configured Range is 1 to 5. The default is 3. through the CLI. The account number and PIN are collected separately.
clid_col_npw_3_cli.tcl
This script authenticates using ANI and NULL for call application voice account numbers and PINs, respectively. If the retry-count authentication fails, it allows the caller to retry. The retry Range is 1 to 5. The default is 3. number is configured through the CLI.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice redirect-number
Table 5
Tcl Scripts and Commands (continued)
Tcl Script Name
Description
clid_col_npw_npw_cli.tcl
This script authenticates using ANI and NULL for call application voice account and PIN, respectively. If authentication fails, it retry-count allows the caller to retry. The retry number is configured Range is 1 to 5. The default is 3. through the CLI. The account number and PIN are collected together.
fax_rollover_on_busy.tcl
Used for on-ramp T.38 fax rollover to T.37 fax when the destination fax line is busy.
Examples
Commands to Configure
voice hunt user-busy
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice redirect-number Warning: This command has been deprecated. Please use the following: param redirect-number
The following example shows how to define a redirect number for the application named “prepaid”: call application voice prepaid redirect-number 5550111
Related Commands
Command
Description
call application voice
Specifies the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
call application voice language
Specifies the language of the audio file for the designated application and passes that information to the application.
call application voice load
Reloads the designated Tcl script.
call application voice pin-len
Defines the number of characters in the PIN for the application and passes that information to the application.
call application voice retry-count
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
call application voice set-location
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
call application voice uid-len
Defines the number of characters in the UID for the designated application and passes that information to the application.
call application voice warning-time
Defines, in seconds, how long in advance a user is warned before the allowed calling time expires for the designated application.
param redirect-number
Defines the telephone number to which a call is redirected—for example, the operator telephone number of the service provider—for an application.
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Cisco IOS Voice Commands: C call application voice retry-count
call application voice retry-count Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice retry-count command is replaced by the param retry-count command in application parameter configuration mode. See the the param retry-count command for more information. To define the number of times that a caller is permitted to reenter the personal identification number (PIN) for the designated application, use the call application voice retry-count command in global configuration mode. To cancel the retry count, use the no form of this command. call application voice application-name retry-count number no call application voice application-name retry-count number
Syntax Description
application-name
Name of the application to which the number of possible retries is being passed.
number
Number of times the caller is permitted to reenter PIN digits. Range is 1 to 5. The default is 3.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1751.
12.2(4)T
This command was introduced on the Cisco 1750.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T. This command is supported on the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 in this release.
12.3(14)T
This command was replaced by the param retry-count command.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
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Cisco IOS Voice Commands: C call application voice retry-count
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the Tool Command Language (Tcl) script being used—or one of the IVR-related features (such as Debit Card) to define how many times a user can reenter a PIN. Table 6 lists Tcl script names and the corresponding commands that are required for each Tcl script.
Table 6
Tcl Scripts and Commands
Tcl Script Name
Description
app_libretto_onramp9.tcl
Authenticates the account and personal identification None number (PIN) using the following: prompt-user, using automatic number identification (ANI), dialed number identification service (DNIS), gateway ID, redialer ID, and redialer DNIS.
app_libretto_offramp5.tcl
Authenticates the account and PIN using the following: None envelope-from, envelope-to, gateway ID, and x-account ID.
clid_4digits_npw_3_cli.tcl
This script authenticates the account number and PIN, respectively, using ANI and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, the script allows the caller to retry. The retry number is also configured through the CLI.
call application voice uid-length Range is 1 to 20. The default is 10.
clid_authen_col_npw_cli.tcl
This script authenticates the account number and PIN, respectively, using ANI and NULL. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
call application voice retry-count Range is 1 to 5. The default is 3.
clid_authen_collect_cli.tcl
This script authenticates the account number and PIN using ANI and DNIS. If the authentication fails, the script allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
call application voice retry-count Range is 1 to 5. The default is 3.
clid_col_npw_3_cli.tcl
This script authenticates using ANI and NULL for account numbers and PINs, respectively. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI.
call application voice retry-count Range is 1 to 5. The default is 3.
clid_col_npw_npw_cli.tcl
This script authenticates using ANI and NULL for call application voice retry-count account and PIN, respectively. If authentication fails, it Range is 1 to 5. The default is 3. allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected together.
fax_rollover_on_busy.tcl
Used for on-ramp T.38 fax rollover to T.37 fax when the voice hunt user-busy destination fax line is busy.
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Commands to Configure
call application voice pin-length Range is 0 to 10. The default is 4. call application voice retry-count Range is 1 to 5. The default is 3.
Cisco IOS Voice Commands: C call application voice retry-count
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application retry-count Warning: This command has been deprecated. Please use the following: param retry-count
The following example shows how to define that for the application named “prepaid” that a user can reenter a PIN three times before being disconnected: call application voice prepaid retry-count 3
Related Commands
Command
Description
call application voice
Specifies the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
call application voice language
Specifies the language of the audio file for the designated application and passes that information to the application.
call application voice load
Reloads the designated Tcl script.
call application voice pin-len Defines the number of characters in the PIN for the application and passes that information to the application. call application voice redirect-number
Specifies the telephone number to which a call is redirected for the designated application.
call application voice set-location
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
call application voice uid-len Defines the number of characters in the UID for the designated application and passes that information to the application. call application voice warning-time
Defines, in seconds, how long in advance a user is warned before the allowed calling time expires for the designated application.
param retry-count
Defines the number of times that a caller is permitted to reenter the PIN for a package.
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Cisco IOS Voice Commands: C call application voice security trusted
call application voice security trusted Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice security trusted command is replaced by the the following commands: •
To set the security level of a VoiceXML application to “trusted” so that automatic number identification (ANI) is not blocked, use the call application voice security trusted command in global configuration mode. To restore the default condition, use the no form of this command. call application voice application-name security trusted no call application voice application-name security trusted
Syntax Description
application-name
Command Default
The security level of the application is not set to trusted, and ANI is blocked.
Command Modes
Global configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced on the Cisco AS5300, Cisco AS5350, and Cisco AS5400.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 3640 and Cisco 3660.
12.3(14)T
The call application voice security trusted command was replaced by the following commands:
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
This command is applicable only for VoiceXML applications.
Tool Command Language (Tcl) applications provide the security parameter to the application but do not use it.
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Name of the application being configured as trusted.
Cisco IOS Voice Commands: C call application voice security trusted
If an application is configured as a trusted application, it is trusted not to provide the calling number to the destination party, so ANI is always provided if available. Normally, the voice gateway does not provide the calling number (ANI) to a VoiceXML application if the caller ID is blocked. Caller ID is blocked if a call that comes into the voice gateway has the presentation indication field set to “presentation restricted”. The session.telephone.ani variable is set to “blocked”. When the call application voice security trusted command is configured, the gateway does not block caller ID; it provides the calling number to the VoiceXML application. If the keyword of this command is set to anything other than trusted, the value is accepted and the application is treated as not trusted. For example, in the following configuration, the application “sample” is treated as not trusted, and caller ID is blocked: call application voice sample security not_trusted
To enable Generic Transparency Descriptor (GTD) parameters in call signaling messages to map to VoiceXML and Tcl session variables, configure the call application voice security trusted command. If this command is not configured, the VoiceXML variables that correspond to GTD parameters are marked as not available. For a detailed description of the VoiceXML and Tcl session variables, see the Cisco VoiceXML Programmer’s Guide and the Tcl IVR API Version 2.0 Programmer’s Guide, respectively.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice security trusted Warning: This command has been deprecated. Please use the following: param security trusted (application parameter configuration mode) paramspace appcommon security trusted
The following example configures the application “sample” as a trusted application. Caller ID is available to this VoiceXML application if it is supported by the service provider. call application voice sample flash:sample.vxml call application voice sample security trusted
The following example configures the application “example” as not trusted. Caller ID can be blocked. call application voice coldcall tftp://joeserver/sellcars.vxml no call application voice example security trusted
Related Commands
Command
Description
call application voice
Defines the name of a voice application and specifies the location of the document (Tcl or VoiceXML) to load for the application.
call application voice language
Defines the language of the audio files used for dynamic prompts by the designated application.
call application voice load
Reloads a Tcl or VoiceXML document.
call application voice pin-len
Defines the number of characters in the PIN for the Tcl application.
call application voice redirect-number
Defines the telephone number to which a call is redirected for the designated application.
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Cisco IOS Voice Commands: C call application voice security trusted
Command
Description
call application voice retry-count
Defines the number of times that a caller is permitted to reenter the PIN for a designated application.
call application voice uid-len
Defines the number of characters in the UID for the designated application.
call application voice warning-time
Defines the number of seconds for which a warning prompt is played before a user’s account time runs out.
param security
Configures security for linkable Tcl functions (packages).
paramspace appcommon security
Configures security for a service (application).
show call application voice
Displays the following information associated with a voice application: the audio files, the prompts, the caller interaction, and the abort key operation.
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Cisco IOS Voice Commands: C call application voice set-location
call application voice set-location Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice set-location command is replaced by the paramspace language command. See the paramspace language command for more information. To define the category and location of audio files that are used for dynamic prompts by the specified IVR application (Tcl or VoiceXML), use the call application voice set-location command in global configuration mode. To remove these definitions, use the no form of this command. call application voice application-name set-location language category location no call application voice application-name set-location language category location
Syntax Description
application-name
Name of the application to which the set-location parameters are being passed.
language
Two-character code that identifies the language associated with the audio files. Valid entries are as follows: •
en—English
•
sp—Spanish
•
ch—Mandarin
•
aa—All
This is the same language code that was entered when configuring the call application voice language command . category
Category group of the audio files (from 0 to 4). For example, audio files representing the days and months can be category 1, audio files representing units of currency can be category 2, and audio files representing units of time—seconds, minutes, and hours—can be category 3. Range is from 0 to 4; 0 means all categories.
location
URL of the audio files. Valid URLs refer to TFTP, FTP, HTTP, or RTSP servers, flash memory, or the removable disks on the Cisco 3600 series.
Command Default
No location or category is set.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and the Cisco AS5300.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
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Cisco IOS Voice Commands: C call application voice set-location
Usage Guidelines
Release
Modification
12.2(2)XB
This command was modified to support VoiceXML applications on the Cisco AS5300, Cisco AS5350, and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1751.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T and implemented on the Cisco 1750.
12.2(8)T
This command was implemented on the Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T for VoiceXML applications. This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 in this release.
12.3(14)T
This command was replaced by the paramspace language command.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Instead of using this command, you can configure the language and location of prerecorded audio files within a Tcl script or VoiceXML document. For more information, see the Tcl IVR API Version 2.0 Programmer’s Guide or Cisco VoiceXML Programmer’s Guide, respectively. To identify the language of the audio files, use the call application voice language command. Tcl scripts and VoiceXML documents can be stored in any of the following locations: On TFTP, FTP, or HTTP servers, in the flash memory on the gateway, or on the removable disks of the Cisco 3600 series. The audio files that they use can be stored in any of these locations, and on RTSP servers. You can configure multiple set-location lines for a single application. With the Pre-Paid Debitcard Multi-Language feature, you can create Tcl scripts and a two-character code for any language. See the Cisco Pre-Paid Debitcard Multi-Language Programmer's Reference. With the multilanguage support for Cisco IOS IVR, you can create a Tcl language module for any language and any set of Text-to-Speech (TTS) notations for use with Tcl and VoiceXML applications. See the Enhanced Multi-Language Support for Cisco IOS Interactive Voice Response document. Table 7 lists Tcl script names and the corresponding commands that are required for each Tcl script.
Table 7
Tcl Scripts and Commands
Tcl Script Name
Description
Commands to Configure
app_libretto_onramp9.tcl
Authenticates the account and personal identification number (PIN) using the following: prompt-user, using automatic number identification (ANI), dialed number identification service (DNIS), gateway ID, redialer ID, and redialer DNIS.
None
app_libretto_offramp5.tcl
Authenticates the account and PIN using the following: envelope-from, envelope-to, gateway ID, and x-account ID.
None
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Cisco IOS Voice Commands: C call application voice set-location
Table 7
Tcl Scripts and Commands (continued)
Tcl Script Name
Description
Commands to Configure
clid_4digits_npw_3_cli.tcl
Authenticates the account number and PIN, respectively, using ANI and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, the script allows the caller to retry. The retry number is also configured through the CLI.
call application voice uid-length Range is 1 to 20. The default is 10. call application voice pin-length Range is 0 to 10. The default is 4. call application voice retry-count Range is 1 to 5. The default is 3.
clid_authen_col_npw_cli.tcl Authenticates the account number and PIN, respectively, call application voice retry-count using ANI and NULL. If the authentication fails, it allows Range is 1 to 5. The default is 3. the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately. clid_authen_collect_cli.tcl
Authenticates the account number and PIN using ANI call application voice retry-count and DNIS. If the authentication fails, the script allows the Range is 1 to 5. The default is 3. caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
clid_col_npw_3_cli.tcl
Authenticates using ANI and NULL for account numbers call application voice retry-count and PINs, respectively. If the authentication fails, it Range is 1 to 5. The default is 3. allows the caller to retry. The retry number is configured through the CLI.
clid_col_npw_npw_cli.tcl
Authenticates using ANI and NULL for account and PIN, call application voice retry-count respectively. If authentication fails, it allows the caller to Range is 1 to 5. The default is 3. retry. The retry number is configured through the CLI. The account number and PIN are collected together.
fax_rollover_on_busy.tcl
Used for on-ramp T.38 fax rollover to T.37 fax when the voice hunt user-busy destination fax line is busy.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice set-location Warning: This command has been deprecated. Please use the following: paramspace language
The following example shows how to configure the call application voice set-location command for the application named “prepaid.” In this example, the language specified is English, the category into which the audio files are grouped is category 0 (meaning all), and the location is the keyer directory on the TFTP server. call application voice prepaid set-location en 0 tftp://keyer/
The following example shows how to configure the call application voice set-location command for a fictitious VoiceXML application named “sample.” In this example, as in the preceding example, the language defined is English, the category into which the audio files are grouped is category 0 (meaning “all”) and the location is the example directory on an HTTP server. call application voice sample set-location en 0 http://example/
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Cisco IOS Voice Commands: C call application voice set-location
The following example shows how to configure the call application voice set-location command for multiple set locations: call application voice sample set-location en 0 http://example/en_msg/ call application voice sample set-location sp 0 http://example/sp_msg/ call application voice sample set-location ch 0 http://example/ch_msg/
Related Commands
Command
Description
call application voice
Specifies the application name and indicates the location of the IVR script to be used with this application.
call application voice language
Specifies the audio file language for the designated application.
call application voice load
Reloads the designated Tcl script.
call application voice pin-len
Specifies the number of characters in the PIN.
call application voice redirect-number
Specifies the telephone number to which a call is redirected.
call application voice retry-count
Defines the number of times a caller is permitted to reenter the PIN.
call application voice uid-len
Defines the number of characters in the UID for the designated application.
call application voice warning-time
Defines, in seconds, how long in advance a user is warned before the allowed calling time expires for the designated application.
paramspace language
Defines the category and location of audio files that are used for dynamic prompts by an IVR application (Tcl or VoiceXML).
show call application voice
Displays information about voice applications.
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Cisco IOS Voice Commands: C call application voice transfer mode
call application voice transfer mode Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice transfer mode command is replaced by the following commands: •
To specify the call-transfer method for Tool Command Language (Tcl) or VoiceXML applications, use the call application voice transfer mode command in global configuration mode. To reset to the default, use the no form of this command. call application voice application-name transfer mode {redirect | redirect-at-alert | redirect-at-connect | redirect-rotary | rotary} no call application voice application-name transfer mode
Syntax Description
application-name
Name of the voice application for which the transfer method is set.
redirect
Gateway redirects the call leg to the redirected destination number.
redirect-at-alert
Gateway places a new call to the redirected destination number and initiates a call transfer when the outgoing call leg is in the alert state. If the call transfer is successful, the two call legs are disconnected on the gateway. If the transfer fails, the gateway bridges the two call legs. Provides support for Two B-Channel Transfer (TBCT).
redirect-at-connect
Gateway places a new call to the redirected destination number and initiates a call transfer when the outgoing call leg is in the connect state. If the call transfer is successful, the two call legs are disconnected on the gateway. If the transfer fails, the gateway bridges the two call legs. Provides support for TBCT.
redirect-rotary
Gateway redirects the call leg to the redirected destination number. If redirection fails, the gateway places a rotary call to the redirected destination number and hairpins the two call legs. For TBCT, this mode is the same as for the redirect-at-connect keyword.
rotary
Gateway places a rotary call for the outgoing call leg and hairpins the two call legs. Call redirection is not invoked. This is the default.
Command Default
Rotary method; call redirection is not invoked.
Command Modes
Global configuration
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Cisco IOS Voice Commands: C call application voice transfer mode
Command History
Release
Modification
12.3(1)
This command was introduced.
12.3(14)T
This command was replaced by the following commands:
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
This command determines whether a voice application can invoke TBCT or RTPvt. Before you can use this command, you must configure the named application on the gateway by using the call application voice command. Redirect-rotary is the preferred transfer method because it ensures that a call-redirect method is always selected if the call leg is capable of it. Tcl scripts can read the value of this command by using the info tag get cfg_avpair transfer-mode statement. For detailed information, see the Tcl IVR API Version 2.0 Programmer’s Guide. For VoiceXML applications, the value of this command becomes the default behavior if the com.cisco.transfer.mode property is not specified in the VoiceXML document. For detailed information, see the Cisco VoiceXML Programmer's Guide. The VoiceXML document property takes precedence over the gateway configuration.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice transfer mode Warning: This command has been deprecated. Please use the following: param mode paramspace callsetup mode
The following example sets the transfer method to redirect for the application callme: call application voice callme transfer mode redirect
Related Commands
Command
Description
application
Enables a voice application on a dial peer.
call application voice
Defines the name of a voice application and specifies the location of the Tcl or VoiceXML document to load for this application.
call application voice transfer reroute-mode
Specifies the call-forwarding behavior of a Tcl application.
debug voip ivr callsetup redirect
Displays debugging information about H.450 calls that are redirected during setup.
debug voip ivr redirect
Displays debugging information about redirected H.450 calls.
isdn supp-service tbct
Enables ISDN TBCT on PRI trunks.
param mode
Configures the call transfer mode for a package.
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Cisco IOS Voice Commands: C call application voice transfer mode
Command
Description
paramspace callsetup mode
Configures the call transfer mode for an application.
show call active voice redirect
Displays information about active calls that are being redirected using RTPvt or TBCT.
show call application voice
Displays information about voice applications.
show call history voice redirect Displays history information about calls that were redirected using RTPvt or TBCT.
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Cisco IOS Voice Commands: C call application voice transfer reroute-mode
call application voice transfer reroute-mode Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice transfer reroute-mode command is replaced by the following commands: •
To specify the call-forwarding behavior of a Tool Command Language (Tcl) application, use the call application voice transfer reroute-mode command in global configuration mode. To reset to the default, use the no form of this command. call application voice application-name transfer reroute-mode {none | redirect | redirect-rotary | rotary} no call application voice application-name transfer reroute-mode
Syntax Description
application-name
Name of the voice application for which the transfer reroute method is set.
none
Call forwarding is not performed by the voice application.
redirect
Two call legs are directly connected. Provides support for RTPvt.
redirect-rotary
Two call legs are directly connected (redirect). If that fails, the two call legs are hairpinned on the gateway (rotary).
rotary
Gateway places a rotary call for the outgoing call leg and hairpins the two calls together. RTPvt is not invoked. This is the default.
Command Default
Rotary method; RTPvt is not invoked.
Command Modes
Global configuration
Command History
Release
Modification
12.3(1)
This command was introduced.
12.3(14)T
This command was replaced by the following commands:
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Cisco IOS Voice Commands: C call application voice transfer reroute-mode
Usage Guidelines
Before you can use this command, you must configure the named application on the gateway by using the call application voice command. This command is not supported for VoiceXML applications or for TBCT. Redirect-rotary is the preferred transfer method because it ensures that a call-redirect method is always selected, provided that the call leg is capable of it. Tcl scripts can read the value of this command by using the info tag get cfg_avpair reroute-mode statement. For detailed information, see the Tcl IVR API Version 2.0 Programmer’s Guide.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice transfer reroute-mode Warning: This command has been deprecated. Please use the following: param reroutemode (application parameter configuration mode) paramspace callsetup reroutemode
The following example sets the call forwarding method to redirect for the application callme: call application voice callme transfer reroute-mode redirect
Related Commands
Command
Description
application
Enables a voice application on a dial peer.
call application voice
Defines the name of a voice application and specifies the location of the Tcl or VoiceXML document to load for this application.
call application voice transfer mode
Specifies the call-transfer behavior of a Tcl or VoiceXML application.
isdn supp-service tbct
Enables ISDN TBCT on PRI trunks.
param reroutemode
Configures the call transfer reroutemode (call forwarding) for a package.
paramspace callsetup reroutemode
Configures the call reroute mode (call forwarding) for an application.
show call active voice redirect
Displays information about active calls that are being redirected using RTPvt or TBCT.
show call application voice
Displays information about voice applications.
show call history voice redirect Displays history information about calls that were redirected using RTPvt or TBCT.
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Cisco IOS Voice Commands: C call application voice uid-length
call application voice uid-length Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice uid-length command is replaced by the param uid-len command. See the param uid-len command for more information. To define the number of characters in the user identification (UID) number for the designated application and to pass that information to the specified application, use the call application voice uid-length command in global configuration mode. To restore the default setting for this command, use the no form of this command. call application voice application-name uid-length number no call application voice application-name uid-length number
Syntax Description
application-name
Name of the application to which the UID length parameter is passed.
number
Number of allowable characters in UIDs that are associated with the specified application. Range is from 1 to 20. The default is 10.
Command Default
number
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and the Cisco AS5300.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1751. This release does not support any other Cisco platforms.
12.2(4)T
Support was added for the Cisco 1750.
12.2(8)T
This command was implemented on the Cisco 7200 series.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T. This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 in this release.
12.3(14)T
This command was replaced by the param uid-len command.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Cisco IOS Voice Command Reference
VR-242
Cisco IOS Voice Commands: C call application voice uid-length
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the Tool Command Language (Tcl) script being used—or one of the IVR-related features (such as Debit Card) to define the number of allowable characters in a UID for the specified application and to pass that information to the specified application. Table 8 lists Tcl script names and the corresponding commands that are required for each Tcl script.
Table 8
Tcl Scripts and Commands
Tcl Script Name
Description
Commands to Configure
app_libretto_onramp9.tcl
Authenticates the account and personal identification number (PIN) using the following: prompt-user, using automatic number identification (ANI), dialed number identification service (DNIS), gateway ID, redialer ID, and redialer DNIS.
None
app_libretto_offramp5.tcl
Authenticates the account and PIN using the following: envelope-from, envelope-to, gateway ID, and x-account ID.
None
clid_4digits_npw_3_cli.tcl
Authenticates the account number and PIN, respectively, using ANI and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, the script allows the caller to retry. The retry number is also configured through the CLI.
call application voice uid-length Range is 1 to 20. The default is 10. call application voice pin-length Range is 0 to 10. The default is 4. call application voice retry-count Range is 1 to 5. The default is 3.
clid_authen_col_npw_cli.tcl Authenticates the account number and PIN, respectively, call application voice using ANI and NULL. If the authentication fails, it allows retry-count the caller to retry. The retry number is configured through Range is 1 to 5. The default is 3. the CLI. The account number and PIN are collected separately. clid_authen_collect_cli.tcl
Authenticates the account number and PIN using ANI and call application voice DNIS. If the authentication fails, the script allows the retry-count caller to retry. The retry number is configured through the Range is 1 to 5. The default is 3. CLI. The account number and PIN are collected separately.
clid_col_npw_3_cli.tcl
Authenticates using ANI and NULL for account numbers call application voice and PINs, respectively. If the authentication fails, it retry-count allows the caller to retry. The retry number is configured Range is 1 to 5. The default is 3. through the CLI.
clid_col_npw_npw_cli.tcl
Authenticates using ANI and NULL for account and PIN, call application voice respectively. If authentication fails, it allows the caller to retry-count retry. The retry number is configured through the CLI. Range is 1 to 5. The default is 3. The account number and PIN are collected together.
fax_rollover_on_busy.tcl
Used for on-ramp T.38 fax rollover to T.37 fax when the destination fax line is busy.
voice hunt user-busy
Cisco IOS Voice Command Reference
VR-243
Cisco IOS Voice Commands: C call application voice uid-length
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice uid-length Warning: This command has been deprecated. Please use the following: param uid-len
The following example shows how to configure four allowable characters in the UID for the application named “sample”: call application voice sample uid-length 4
Related Commands
Command
Description
call application voice
Specifies the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
call application voice language
Specifies the language of the audio file for the designated application and passes that information to the application.
call application voice load
Reloads the designated Tcl script.
call application voice pin-len Defines the number of characters in the PIN for the application and passes that information to the application. call application voice redirect-number
Specifies the telephone number to which a call is redirected for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
call application voice set-location
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
call application voice warning-time
Defines, in seconds, how long in advance a user is warned before the allowed calling time expires for the designated application.
param uid-length
Defines the number of characters in the UID for a package.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice voice-dtmf
call application voice voice-dtmf Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice voice-dtmf command is replaced by the param voice-dtmf command. See the param voice-dtmf command for more information. To direct the fax detection interactive voice response (IVR) application to recognize a specified digit to indicate a voice call, use the call application voice voice-dtmf command in global configuration mode. To remove configuration of this digit, use the no form of this command. call application voice application-name voice-dtmf {keypad-character} no call application voice application-name voice-dtmf {keypad-character}
Syntax Description
application-name
The name of the fax detection application that you defined when you loaded the application on the router.
keypad-character
Single character that can be dialed on a telephone keypad pressed by the calling party to indicate a voice call, in response to the audio prompt configured in default-voice and default-fax mode of the fax detection IVR application. Default is 1.
Command Default
1
Command Modes
Global configuration
Command History
Release
Modification
12.1(5)XM
This command was introduced for the Cisco AS5300.
12.2(2)XB
This command was implemented on the Cisco AS5400 and Cisco AS5350.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(11)T
This command was supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
12.3(14)T
This command was replaced by the param voice-dtmf command.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Usage Guidelines
This command is useful only when the fax detection IVR application is being configured in default-voice mode or default-fax mode, as defined by the call application voice mode command. Only one digit can be specified in this command, and that digit must be different from the digit specified in the call
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice voice-dtmf
application voice fax-dtmf command. You are not notified immediately if you make the error of configuring them both to the same digit. To find this error, you must start debugging with the debug voip ivr script command and then observe some failing calls. This command is not supported by Cisco IOS help. If you type the call application voice fax_detect voice-dtmf command and a question mark (?), the Cisco IOS help does not supply a list of entries that are valid in place of the question mark.
Examples
Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice voice-dtmf Warning: This command has been deprecated. Please use the following: param voice-dtmf
The following example selects digit 2 dual tone multifrequency (DTMF) to indicate a voice call: call application voice fax_detect script_url call application voice fax_detect voice-dtmf 2 dial-peer voice 302 pots application fax_detect
Related Commands
Command
Description
call application voice
Loads a specified IVR application onto the router from the TFTP server and gives it an application name by which it is known on the router.
call application voice account-id-method
Configures the fax detection IVR application to use a particular method to assign the account identifier.
call application voice fax-dtmf
Configures the fax detection IVR application to recognize a specified digit to indicate a fax call.
call application voice mode
Configures the fax detection IVR application to operate in one of its four modes.
call application voice prompt
Configures the fax detection IVR application to use the specified audio file as a user prompt.
param voice-dtmf
Directs the fax detection IVR application to recognize a specified digit to indicate a voice call.
Cisco IOS Voice Command Reference
VR-246
Cisco IOS Voice Commands: C call application voice warning-time
call application voice warning-time Note
Effective with Cisco IOS Release 12.3(14)T, the call application voice warning-time command is replaced by the param warning-time command. See the param warning-time command for more information. To define the number of seconds of warning that a user receives before the allowed calling time expires use the call application voice warning-time command in global configuration mode. To remove the configured warning period, use the no form of this command. call application voice application-name warning-time seconds no call application voice application-name warning-time seconds
Syntax Description
application-name
Name of the application to which the warning time parameter is being passed.
seconds
Length of the warning period, in seconds, before the allowed calling time expires. Range is from 10 to 600. This argument has no default value.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1751. Support for other Cisco platforms is not included in this release.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T and implemented on the Cisco 1750.
12.2(8)T
This command was implemented on the Cisco 7200 series.
12.2(11)T
This command was implemented on the Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 in this release.
12.3(14)T
This command was replaced by the param warning-time command.
12.4(24)T
This command was modified. The automatic conversion to the new CLI is replaced with an explicit error message.
Cisco IOS Voice Command Reference
VR-247
Cisco IOS Voice Commands: C call application voice warning-time
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the Tool Command Language (Tcl) script being used—or one of the IVR-related features (such as Debit Card) to define the number of seconds in the warning period before the allowed calling time expires for the specified application and to pass that information to the specified application. Table 9 lists Tcl script names and the corresponding commands that are required for each Tcl script.
Table 9
Tcl Scripts and Commands
Tcl Script Name
Description
Commands to Configure
app_libretto_onramp9.tcl
Authenticates the account and personal identification number (PIN) using the following: prompt-user, using automatic number identification (ANI), dialed number identification service (DNIS), gateway ID, redialer ID, and redialer DNIS.
None
app_libretto_offramp5.tcl
Authenticates the account and PIN using the following: envelope-from, envelope-to, gateway ID, and x-account ID.
None
clid_4digits_npw_3_cli.tcl
This script authenticates the account number and PIN, respectively, using ANI and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, the script allows the caller to retry. The retry number is also configured through the CLI.
call application voice uid-length Range is 1 to 20. The default is 10. call application voice pin-length Range is 0 to 10. The default is 4. call application voice retry-count Range is 1 to 5. The default is 3.
clid_authen_col_npw_cli.tcl
This script authenticates the account number and PIN, call application voice respectively, using ANI and NULL. If the authentication retry-count fails, it allows the caller to retry. The retry number is Range is 1 to 5. The default is 3. configured through the CLI. The account number and PIN are collected separately.
clid_authen_collect_cli.tcl
This script authenticates the account number and PIN call application voice using ANI and DNIS. If the authentication fails, the script retry-count allows the caller to retry. The retry number is configured Range is 1 to 5. The default is 3. through the CLI. The account number and PIN are collected separately.
clid_col_npw_3_cli.tcl
This script authenticates using ANI and NULL for call application voice account numbers and PINs, respectively. If the retry-count authentication fails, it allows the caller to retry. The retry Range is 1 to 5. The default is 3. number is configured through the CLI.
clid_col_npw_npw_cli.tcl
This script authenticates using ANI and NULL for call application voice account and PIN, respectively. If authentication fails, it retry-count allows the caller to retry. The retry number is configured Range is 1 to 5. The default is 3. through the CLI. The account number and PIN are collected together.
fax_rollover_on_busy.tcl
Used for on-ramp T.38 fax rollover to T.37 fax when the voice hunt user-busy destination fax line is busy.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call application voice warning-time
Examples Effective with Cisco IOS Release 12.4(24)T, the following warning message is displayed to direct users to the replacement command options: Router(config)# call application voice param warning-time Warning: This command has been deprecated. Please use the following: param warning-time
The following example shows how to configure a 30-second warning time for the application named “sample”: call application voice sample warning-time 30
Related Commands
Command
Description
call application voice language Specifies the language of the audio file for the designated application and passes that information to the application. call application voice load
Reloads the designated Tcl script.
call application voice location
Specifies the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
call application voice pin-len
Defines the number of characters in the PIN for the application and passes that information to the application.
call application voice redirect-number
Specifies the telephone number to which a call is redirected for the designated application.
call application voice retry-count
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
call application voice set-location
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
call application voice uid-length
Defines the number of characters in the UID for the designated application and passes that information to the application.
param warning-time
Defines the number of seconds of warning that a user receives before the allowed calling time expires.
Cisco IOS Voice Command Reference
VR-249
Cisco IOS Voice Commands: C call-block (dial peer)
call-block (dial peer) To enable blocking of incoming calls, use the call-block command in dial peer configuration mode. To return to the default value, use the no form of this command. call-block {disconnect-cause incoming {call-reject | invalid-number | unassigned-number | user-busy} | translation-profile incoming name} no call-block {disconnect-cause incoming {call-reject | invalid-number | unassigned-number | user-busy} | translation-profile incoming name}
Syntax Description
disconnect-cause incoming
Associates a disconnect cause of incoming calls.
call-reject
Specifies call rejection as the cause for blocking a call during incoming call-number translation.
invalid-number
Specifies invalid number as the cause for blocking a call during incoming call-number translation.
unassigned-number
Specifies unassigned number as the cause for blocking a call during incoming call-number translation.
user-busy
Specifies busy as the cause for blocking a call during incoming call-number translation.
translation-profile incoming
Associates the translation profile for incoming calls.
name
Name of the translation profile.
Command Default
Disconnect cause: No Service (once the call-blocking translation profile is defined) Translation profile: No default behavior or values
Command Modes
Dial peer configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
Usage Guidelines
An incoming call can be blocked from the gateway if one of the call numbers (calling, called, or redirect) is matched with the reject translation rule of the incoming call-blocking translation profile. The cause value is returned to the source of the call when a call is blocked during the incoming call-number translation. This command is supported in POTS, VoIP, VoFR, and VoATM dial-peer configuration. For VoATM, only ATM Adaptation Layer 5 (AAL5) calls are supported.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: C call-block (dial peer)
The only option for call blocking is in the incoming direction. From the perspective of the voice gateway, the incoming direction can be either of the following: •
Incoming from a telephony device directly attached to a voice port on the gateway toward the gateway itself
•
Incoming by way of an inbound Voice over X (VoX) call from a peer gateway
To configure incoming call blocking, define a translation rule with a reject keyword. For example: voice translation-rule 1 rule 1 reject /408252*/
Apply the rule to a translation profile for called, calling, or redirect-called numbers, such as: voice translation profile call_block_profile translate calling 1
Include the translation profile within a dial peer definition. For example: dial-peer voice 111 pots call-block translation-profile incoming call_block_profile call-block disconnect-cause incoming invalid_number
In this example, the gateway blocks any incoming time-division multiplexing (TDM) call that successfully matches inbound dial-peer 111 and has a calling number that starts with 408252. The gateway also returns the disconnect cause “invalid number” to the source of the call. (Other disconnect causes can be assigned: unassigned-number, user-busy, or call-rejected.)
Examples
The following example assigns the translation profile “example” to be used for incoming calls and returns the message “invalid number” as a cause for blocked calls: Router(config)# dial-peer voice 5 pots Router(config-dial-peer)# call-block translation-profile incoming example Router(config-dial-peer)# call-block disconnect-cause incoming invalid-number
Following are two possible call-blocking scenarios: Scenario 1: Block Inbound Calls from the PSTN/PBX/CO
We place the rejection profile on a POTS dial peer that is associated with the voice port on which we expect the inbound call. When the inbound call attempt is made, we see in the CCAPI debugs that POTS dial-peer 9 is matched for the telephony call leg. The call-block rule is checked and we send back user-busy to the switch. voice translation-rule 1 rule 1 reject /9193927582/
The following example shows how to configure SIP digest credentials using the encrypted format: Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# credentials dhcp password 7 095FB01AA000401 realm example.com
The following example shows how to disable SIP digest credentials where the encryption type was specified: Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# no credentials dhcp password 7 095FB01AA000401 realm example.com
The following example shows how to configure SIP digest credentials for two different realms without specifying the encryption type: Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# credentials number 1111 username MyUser password MyPassword realm MyLocation1.example.com Router(config-sip-ua)# credentials number 1111 username MyUser password MyPassword realm MyLocation2.example.com
Related Commands
Command
Description
authentication (dial peer)
Enables SIP digest authentication on an individual dial peer.
authentication (SIP UA)
Enables SIP digest authentication.
localhost
Configures global settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages.
registrar
Enables Cisco IOS SIP TDM gateways to register E.164 numbers for FXS, EFXS, and SCCP phones on an external SIP proxy or SIP registrar.
voice-class sip localhost
Configures settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages on an individual dial peer, overriding the global setting.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: D This chapter contains commands to configure and maintain Cisco IOS voice applications. The commands are presented in alphabetical order beginning with the letter D. Some commands required for configuring voice may be found in other Cisco IOS command references. Use the master index of commands or search online to find these commands. For detailed information on how to configure these applications and features, refer to the Cisco IOS Voice Configuration Library.
Cisco IOS Voice Command Reference
VR-565
Cisco IOS Voice Commands: D default (auto-config application)
default (auto-config application) To configure an auto-config application configuration command to its default value, use the default command in auto-config application configuration mode. default command
Syntax Description
command
One of the auto-config application configuration commands. Valid choices are as follows: •
retries
•
server
•
shutdown
•
timeout
Command Default
No default behavior or values
Command Modes
Auto-config application configuration
Command History
Release
Modification
12.3(8)XY
This command was introduced on the Communication Media Module.
12.3(14)T
This command was integrated into Cisco IOS Release 12.3(14)T.
Examples
The following example shows the default command used to set the number of download retry attempts for an auto-configuration application to its default value. Router(auto-config-app)# default retries
Related Commands
Command
Description
auto-config
Enables auto-configuration or enters auto-config application configuration mode for the SCCP application.
show auto-config
Displays the current status of auto-config applications.
Cisco IOS Voice Command Reference
VR-566
Cisco IOS Voice Commands: D default (MGCP profile)
default (MGCP profile) To configure a Media Gateway Control Protocol (MGCP profile) command to its default value, use the default command in MGCP profile configuration mode. To disable the default command, use the no form of the command for that profile parameter. default command no default command
Syntax Description
command
One of the MGCP profile commands. Valid choices are as follows: •
call-agent
•
description (MGCP profile)
•
max1 lookup
•
max1 retries
•
max2 lookup
•
max2 retries
•
package persistent
•
timeout tcrit
•
timeout tdinit
•
timeout tdmax
•
timeout tdmin
•
timeout thist
•
timeout tone busy
•
timeout tone cot1
•
timeout tone cot2
•
timeout tone dial
•
timeout tone dial stutter
•
timeout tone mwi
•
timeout tone network congestion
•
timeout tone reorder
•
timeout tone ringback
•
timeout tone ringback connection
•
timeout tone ringing
•
timeout tone ringing distinctive
•
timeout tpar
•
timeout tsmax
•
voice-port (MGCP profile)
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: D default (MGCP profile)
Command Default
No default behaviors or values
Command Modes
MGCP profile configuration
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T.
12.2(11)T
This command implemented on the Cisco AS5300 and Cisco AS5850.
Usage Guidelines
This command is used when configuring values for an MGCP profile. The default (MGCP profile) command instructs the MGCP profile to use the default value of the specified command whenever the profile is called. This has the same effect as using the no form of the specified command, but the default command clearly specifies which commands are using their default values. To use the default values for more than one command, enter each command on a separate line.
Examples
The following example shows how to configure the default values for three MGCP profile commands: Router(config)# mgcp profile Router(config-mgcp-profile)# Router(config-mgcp-profile)# Router(config-mgcp-profile)#
Related Commands
Command
Description
mgcp
Starts and allocates resources for the MGCP daemon.
mgcp profile
Initiates MGCP profile mode to create and configure a named MGCP profile associated with one or more endpoints or to configure the default profile.
default (SIP) To reset a SIP command to its default value, use the default command in SIP configuration mode. default command
Syntax Description
command
One of the SIP configuration commands. Valid choices are: •
bind: Configures the source address of signaling and media packets to a specific interface’s IP address.
•
rel1xx: Enables all SIP provisional responses (other than 100 Trying) to be sent reliably to the remote SIP endpoint.
•
session-transport: Configures the underlying transport layer protocol for SIP messages to TCP or UDP.
•
url: Configures URLs to either the SIP or TEL format for your voip sip calls.
Defaults
The default is that binding is disabled (no bind).
Command Modes
SIP configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced on the Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco AS5300, Cisco AS5350, and Cisco AS5400 platforms.
12.2(2)XB2
This command was implemented on the Cisco AS5850 platform.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and support was added for the Cisco 3700 series. Cisco AS5300, Cisco AS5350, Cisco AS5850, and Cisco AS5400 platforms were not supported in this release.
12.2(11)T
Support was added for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms.
Cisco IOS XE Release 2.5
This command was integrated into Cisco IOS XE Release 2.5.
Examples
The following example shows how to reset the value of the SIP bind command: Router(config)# voice serv voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# default bind
Cisco IOS Voice Command Reference
VR-569
Cisco IOS Voice Commands: D default (SIP)
Related Commands
Command
Description
sip
Enter SIP configuration mode from voice-service VoIP configuration mode.
Cisco IOS Voice Command Reference
VR-570
Cisco IOS Voice Commands: D default-file vfc
default-file vfc To specify an additional (or different) file from the ones in the default file list and stored in voice feature card (VFC) Flash memory, use the default-file vfc command in global configuration mode. To delete the file from the default file list, use the no form of this command. default-file filename vfc slot no default-file filename vfc slot
Syntax Description
filename
Indicates the file to be retrieved from VFC Flash memory and used to boot up the system.
slot
Indicates the slot on the Cisco AS5300 in which the VFC is installed. Range is to 2. There is no default value.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
11.3(1)NA
This command was introduced on the Cisco AS5300.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
Usage Guidelines
When VCWare is unbundled, it automatically adds DSPWare to Flash memory, creates both the capability and default file lists, and populates these lists with the default files for that version of VCWare. The default file list includes the files that is used to boot up the system. Use the default-file vfc command to add a specified file to the default file list, replacing the existing default for that extension type.
Examples
The following example specifies that the bas-vfc-1.0.14.0.bin file, which is stored in VFC Flash memory, be added to the default file list: default-file bas-vfc-1.0.14.0.bin vfc 0
Related Commands
Command
Description
cap-list vfc
Adds a voice codec overlay file to the capability file list.
delete vfc
Deletes a file from VFC Flash memory.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: D define
define To define the transmit and receive bits for North American ear and mouth (E&M), E&M Mercury Exchange Limited Channel-Associated Signaling (MELCAS), and Land Mobile Radio (LMR) voice signaling, use the define command in voice-port configuration mode. To restore the default value, use the no form of this command. define {tx-bits | rx-bits} {seize | idle} {0000 | 0001 | 0010 | 0011 | 0100 | 0101 | 0110 | 0111 | 1000 | 1001 | 1010 | 1011 | 1100 | 1101 | 1110 | 1111} no define {tx-bits | rx-bits} {seize | idle} {0000 | 0001 | 0010 | 0011 | 0100 | 0101 | 0110 | 0111 | 1000 | 1001 | 1010 | 1011 | 1100 | 1101 | 1110 | 1111}
Syntax Description
Command Default
tx-bits
The bit pattern applies to the transmit signaling bits.
rx-bits
The bit pattern applies to the receive signaling bits.
seize
The bit pattern defines the seized state.
idle
The bit pattern defines the idle state.
0000 through 1111
Specifies the bit pattern.
The default is to use the preset signaling patterns as defined in American National Standards Institute (ANSI) and European Conference of Postal and Telecommunications Administrations (CEPT) standards, as follows: •
For North American E&M: – tx-bits idle 0000 (0001 if on E1 trunk) – tx-bits seize 1111 – rx-bits idle 0000 – rx-bits seize 1111
This command was implemented on the Cisco 2600 series and Cisco 3600 series.
12.1(2)T
The command was integrated into Cisco IOS Release 12.1(2)T.
12.3(4)XD
The LMR signaling type was added to the signaling types to which this command applies.
12.3(7)T
This command was integrated into Cisco IOS Release 12.3(7)T.
12.3(14)T
This command was implemented on the Cisco 2800 series and Cisco 3800 series.
12.4(2)T
This command was integrated into Cisco IOS Release 12.4(2)T.
The define command applies to E&M digital voice ports associated with T1/E1 controllers. Use the define command to match the E&M bit patterns with the attached telephony device. Be careful not to define invalid configurations, such as all 0000 on E1, or identical seized and idle states. Use this command with the ignore command. In LMR signaling, the define command is used to define polarity on E&M analog and digital voice ports.
Examples
To configure a voice port on a Cisco 2600 or Cisco 3600 series router that is sending traffic in North American E&M signaling format to convert the signaling to MELCAS format, enter the following commands: voice-port 1/0/0 define rx-bits idle 1101 define rx-bits seize 0101 define tx-bits idle 1101 define tx-bits seize 0101
In this example, reverse polarity is configured on a voice port on a Cisco 3700 series router that is sending traffic in LMR signaling format: voice-port 1/0/0 define rx-bits idle 1111 define rx-bits seize 0000 define tx-bits idle 1111 define tx-bits seize 0000
Related Commands
Command
Description
condition
Manipulates the signaling bit-pattern for all voice signaling types.
ignore
Configures a North American E&M or E&M MELCAS voice port to ignore specific receive bits.
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Cisco IOS Voice Commands: D delete vfc
delete vfc To delete a file from voice feature card (VFC) Flash memory, use the delete vfc command in privileged EXEC mode. delete filename vfc slot
Syntax Description
filename
Specifies the file in VFC Flash memory to be deleted.
slot
Specifies the slot on the Cisco AS5300 in which the specified VFC resides. Range is from 0 to 2.
Command Modes
Privileged EXEC
Command History
Release
Modification
11.3(1)NA
This command was introduced on the Cisco AS5300.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T.
Usage Guidelines
Note
Examples
Use the delete vfc command to delete a specific file from VFC Flash memory and to remove the file from the default list or capability list if the specified file is included in those lists.
Deleting a file from VFC Flash memory does not free the VFC Flash memory space that the file occupied. To free VFC Flash memory space, use the erase vfc command.
The following example deletes the bas-vfc-1.0.14.0.bin file, which is stored in VFC Flash memory of the VFC located in slot 0: Router# delete bas-vfc-1.0.14.0.bin vfc 0
Related Commands
Command
Description
default-file vfc
Specifies an additional (or different) file from the ones in the default file list and stored in VFC Flash memory.
erase vfc
Erases the Flash memory of a specified VFC.
show vfc directory
Displays the list of all files that reside on this VFC.
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Cisco IOS Voice Commands: D description
description To specify a description of the digital signal processor (DSP) interface, use the description command in voice-port or DSP farm interface configuration mode. To describe a MGCP profile that is being defined, use the description command in MGCP profile configuration mode. To specify the name or a brief description of a charging profile, use the description command in charging profile configuration mode. To delete a configured description, use the no form of the command in the appropriate configuration mode. description string no description
Syntax Description
string
Command Default
Enabled with a null string. The MGCP profile has no default description. Charging profiles have no default description.
This command was introduced on the Cisco 3600 series and Cisco 7200.
11.3(1)MA
This command in voice-port configuration mode was implemented on the Cisco MC3810.
12.0(5)XE
This command in DSP farm interface configuration mode was modified.
12.1(1)T
The DSP farm interface configuration mode modification was integrated into Cisco IOS Release 12.1(1)T.
12.2(2)XA
This command was implemented on the Cisco AS5300.
12.2(11)T
Support for the Cisco AS5300 and Cisco AS5850 was added.
12.3(8)XU
This command was introduced in charging profile configuration mode.
12.3(11)YJ
This command in charging profile configuration mode was integrated into Cisco IOS Release 12.3(11)YJ.
12.3(14)YQ
This command in charging profile configuration mode was integrated into Cisco IOS Release 12.3(14)YQ.
12.4(9)T
This command in charging profile configuration mode was integrated into Cisco IOS Release 12.4(9)T.
Character string from 1 to 80 characters for DSP interfaces and MGCP profiles, or from 1 to 99 characters for charging profiles.
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Cisco IOS Voice Commands: D description
Usage Guidelines
Use the description command to describe the DSP interface connection or a defined MGCP profile. The information is displayed when a show command is used, and it does not affect the operation of the interface in any way.
Examples
The following example identifies voice port 1/0/0 as being connected to the purchasing department: voice-port 1/0/0 description purchasing_dept
The following example identifies DSP farm interface 1/0 as being connected to the marketing department: dspint dspfarm 1/0 description marketing_dept
The following example shows a description for an MGCP profile: mgcp profile newyork description This is the head sales office in New York. dot ...(socket=0) S:. R:250 NAA09092 Message accepted for delivery S:QUIT R:221 [email protected] closing connection Freeing SMTP ctx at 0x6121D454 returned from work_routine, context freed
Related Commands
Command
Description
category
Identifies the subscriber category to which a charging profile applies.
cdr suppression
Specifies that CDRs be suppressed as a charging characteristic in a charging profile.
charging profile content dcca profile
Defines a DCCA client profile in a GGSN charging profile.
content postpaid time
Specifies, as a trigger condition for postpaid users in a charging profile, the time duration limit that when exceeded causes the GGSN to collect upstream and downstream traffic byte counts and close and update the G-CDR for a particular PDP context.
content postpaid validity
Specifies, as a trigger condition in a charging profile, that the amount of time quota granted to a postpaid user is valid.
content postpaid volume
Specifies, as a trigger condition for postpaid users in a charging profile, the maximum number of bytes that the GGSN maintains across all containers for a particular PDP context before closing and updating the G-CDR.
content rulebase
Associates a default rule-base ID with a charging profile.
gprs charging characteristics reject
Specifies that create PDP context requests for which no charging profile can be selected be rejected by the GGSN.
Creates a new charging profile (or modifies an existing one) and enters charging profile configuration mode.
Cisco IOS Voice Commands: D description
Command
Description
limit duration
Specifies, as a trigger condition in a charging profile, the time duration limit that when exceeded causes the GGSN to collect upstream and downstream traffic byte counts and close and update the G-CDR for a particular PDP context.
limit sgsn-change
Specifies, as a trigger condition in a charging profile, the maximum number of GGSN changes that can occur before closing and updating the G-CDR for a particular PDP context.
limit volume
Specifies, as a trigger condition in a charging profile, the maximum number of bytes that the GGSN maintains across all containers for a particular PDP context before closing and updating the G-CDR.
mgcp
Starts and allocates resources for the MGCP daemon.
mgcp profile
Initiates MGCP profile mode to create and configure a named MGCP profile associated with one or more endpoints or to configure the default profile.
tariff-time
Specifies that a charging profile use the tariff changes configured using the gprs charging tariff-time global configuration command.
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Cisco IOS Voice Commands: D description (dial peer)
description (dial peer) To add a description to a dial peer, use the description command in dial peer configuration mode. To remove the description, use the no form of this command. description description no description
Syntax Description
description
Command Default
Disabled
Command Modes
Dial peer configuration
Command History
Release
Modification
12.2(2)T
This command was introduced.
Text string up to 64 alphanumeric characters.
Usage Guidelines
Use this command to include descriptive text about the dial peer. The description displays in show command output and does not affect the operation of the dial peer.
Examples
The following example shows a description included in a dial peer: dial-peer voice 1 pots description inbound PSTN calls
Related Commands
Command
Description
dial-peer voice
Defines a dial peer.
show dial-peer voice
Displays configuration information for dial peers.
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Cisco IOS Voice Commands: D description (DSP Farm profile)
description (DSP Farm profile) To include a description about the digital signal processor (DSP) farm profile, use the description command in DSP farm profile configuration mode. To remove a description, use the no form of this command. description text no description
Syntax Description
text
Command Default
No default behavior or values
Command Modes
DSP farm profile configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
Character string from 1 to 80 characters.
Usage Guidelines
Use this command to include descriptive text about this DSP farm profile. This information displays in show commands and does not affect the operation of the interface.
Examples
The following example identifies the DSP farm profile as being designated to the art department: Router(config-dspfarm-profile)# description art_dept
Related Commands
Command
Description
codec (DSP Farm profile)
Specifies the codecs supported by a DSP farm profile.
dspfarm profile
Enters DSP farm profile configuration mode and defines a profile for DSP farm services.
maximum sessions (DSP Farm profile)
Specifies the maximum number of sessions that need to be supported by the profile.
shutdown (DSP Farm profile)
Allocates DSP farm resources and associates with the application.
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Cisco IOS Voice Commands: D description (dspfarm)
description (dspfarm) To include a specific description about the digital signal processor (DSP) interface, use the description command in DSPfarm interface configuration mode. To disable this feature, use the no form of this command. description string no description string
Syntax Description
string
Command Default
Enabled with a null string.
Command Modes
DSPfarm interface configuration
Command History
Release
Modification
11.3(1)T
This command was introduced for the Cisco 7200 series routers.
12.0(5)XE
This command was modified to reduce the maximum number of allowable characters in a text string from 255 to 80.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Character string from 1 to 80 characters.
Usage Guidelines
Use the description command to include descriptive text about this DSP interface connection. This information is displayed when you issue a show command and does not affect the operation of the interface in any way.
Examples
The following example identifies DSPfarm interface 1/0 on the Cisco 7200 series routers router as being connected to the marketing department: dspint dspfarm 1/0 description marketing_dept
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Cisco IOS Voice Commands: D description (SCCP Cisco CallManager)
description (SCCP Cisco CallManager) To include a description about the Cisco CallManager group, use the description command in SCCP Cisco CallManager configuration mode. To remove a description, use the no form of this command. description text no description
Syntax Description
text
Command Default
No default behavior or values
Command Modes
SCCP Cisco CallManager configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
Character string from 1 to 80 characters.
Usage Guidelines
Use this command to include descriptive text about a Cisco CallManager group. This information is displayed in show commands and does not affect the operation of the interface.
Examples
The following example identifies SCCP as being designated to the Boston office: Router(config-sccp-ccm)# description boston office
Related Commands
Command
Description
associate ccm
Associates a Cisco CallManager with a Cisco CallManager group and establishes its priority within the group.
connect retries
Specifies the number of times that a DSP farm attempts to connect to a Cisco CallManager when the current Cisco CallManager connections fails.
sccp ccm group
Creates a Cisco CallManager group and enters SCCP Cisco CallManager configuration mode.
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Cisco IOS Voice Commands: D description (trunk group)
description (trunk group) To add a description to a trunk group, use the description command in trunk group configuration mode. To delete the description, use the no form of this command. description text no description text
Syntax Description
text
Command Default
No default behavior or values
Command Modes
Trunk group configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
Examples
Trunk group description. Maximum length is 63 alphanumeric characters.
The following example shows a description for a trunk group: Router(config)# trunk group alpha1 Router(config-trunk-group)# description carrierAgroup1
Related Commands
Command
Description
trunk group
Initiates the definition of a trunk group.
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Cisco IOS Voice Commands: D description (voice source group)
description (voice source group) To add a description to a voice source group, use the description command in voice source-group configuration mode. To delete the description, use the no form of this command. description text no description text
Syntax Description
text
Command Default
No default behavior or values
Command Modes
Voice source-group configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
Examples
Describes a voice source group, Maximum length of the voice source group description is 63 alphanumeric characters.
The following example shows a description for a voice source group: Router(config)# voice source-group northern1 Router(cfg-source-grp)# description carrierBgroup3
Related Commands
Command
Description
voice source-group
Defines a source group for voice calls.
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Cisco IOS Voice Commands: D destination uri
destination uri To specify the voice class used to match a dial peer to the destination uniform resource identifier (URI) of an outgoing call, use the destination uri command in dial peer configuration mode. To remove the URI voice class, use the no form of this command. destination uri tag no destination uri
Syntax Description
tag
Command Default
No default behavior or values
Command Modes
Dial peer configuration
Command History
Release
Modification
12.3(4)T
This command was introduced.
Usage Guidelines
Alphanumeric label that uniquely identifies the voice class. This tag must be configured with the voice class uri command.
•
Before you use this command, configure the voice class by using the voice class uri command.
•
This command applies new rules for dial-peer matching. Table 16 shows the rules and the order in which they are applied when the destination uri command is used. The gateway compares the dial-peer command to the call parameter in its search to match an outbound call to a dial peer. All dial peers are searched based on the first match criteria. Only if no match is found does the gateway move on to the next criteria.
Table 16
Dial-Peer Matching Rules for Outbound URI
Match Order Cisco IOS Command 1
destination uri and carrier-id target
Application-provided URI and target carrier ID associated with the call
2
destination-pattern and carrier-id target
Called number and target carrier ID associated with the call
3
destination uri
Application-provided URI
4
destination-pattern
Called number
5
carrier-id target
Target carrier ID associated with the call
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Outgoing Call Parameter
Cisco IOS Voice Commands: D destination uri
Note
Examples
Calls whose destination is an E.164 number, rather than a URI, use the previously existing dial-peer matching rules. For information, see the Dial Peer Configuration on Voice Gateway Routers document, Cisco IOS Voice Library.
The following example matches the destination URI in the outgoing call by using voice class ab100: dial-peer voice 100 voip destination uri ab100
Related Commands
Command
Description
answer-address
Specifies calling number to match for a dial peer.
debug voice uri
Displays debugging messages related to URI voice classes.
destination-pattern
Specifies telephone number to match for a dial peer.
dial-peer voice
Enters dial peer configuration mode to create or modify a dial peer.
incoming uri
Specifies the voice class that a VoIP dial peer uses to match the URI of an incoming call.
pattern
Matches a call based on the entire SIP or TEL URI.
session protocol
Specifies a session protocol for calls between local and remote routers using the packet network.
show dialplan uri
Displays which outbound dial peer is matched for a specific destination URI.
voice class uri
Creates or modifies a voice class for matching dial peers to calls containing a SIP or TEL URI.
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Cisco IOS Voice Commands: D destination-pattern
destination-pattern To specify either the prefix or the full E.164 telephone number to be used for a dial peer, use the destination-pattern command in dial peer configuration mode. To disable the configured prefix or telephone number, use the no form of this command. destination-pattern [+]string[T] no destination-pattern [+]string[T]
Syntax Description
+
(Optional) Character that indicates an E.164 standard number.
string
Series of digits that specify a pattern for the E.164 or private dialing plan telephone number. Valid entries are the digits 0 through 9, the letters A through D, and the following special characters: •
The asterisk (*) and pound sign (#) that appear on standard touch-tone dial pads.
•
Comma (,), which inserts a pause between digits.
•
Period (.), which matches any entered digit (this character is used as a wildcard).
•
Percent sign (%), which indicates that the preceding digit occurred zero or more times; similar to the wildcard usage.
•
Plus sign (+), which indicates that the preceding digit occurred one or more times.
Note
T
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The plus sign used as part of a digit string is different from the plus sign that can be used preceding a digit string to indicate that the string is an E.164 standard number.
•
Circumflex (^), which indicates a match to the beginning of the string.
•
Dollar sign ($), which matches the null string at the end of the input string.
•
Backslash symbol (\), which is followed by a single character, and matches that character. Can be used with a single character with no other significance (matching that character).
•
Question mark (?), which indicates that the preceding digit occurred zero or one time.
•
Brackets ([ ]), which indicate a range. A range is a sequence of characters enclosed in the brackets; only numeric characters from 0 to 9 are allowed in the range.
•
Parentheses (( )), which indicate a pattern and are the same as the regular expression rule.
(Optional) Control character that indicates that the destination-pattern value is a variable-length dial string. Using this control character enables the router to wait until all digits are received before routing the call.
Cisco IOS Voice Commands: D destination-pattern
Command Default
The command is enabled with a null string.
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series.
11.3(1)MA
This command was implemented on the Cisco MC3810.
12.0(4)XJ
This command was modified for store-and-forward fax.
12.1(1)
The command was integrated into Cisco IOS Release 12.1(1).
12.0(7)XR
This command was implemented on the Cisco AS5300 and modified to support the plus sign, percent sign, question mark, brackets, and parentheses symbols in the dial string.
12.0(7)XK
This command was modified. Support for the plus sign, percent sign, question mark, brackets, and parentheses in the dial string was added to the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T and implemented on the Cisco 1750, Cisco 7200 series, and Cisco 7500 series. The modifications for the Cisco MC3810 in Cisco IOS Release12.0(7)XK are not supported in this release.
12.1(2)T
The modifications made in Cisco IOS Release 12.0(7)XK for the Cisco MC3810 were integrated into Cisco IOS Release 12.1(2)T.
12.2(8)T
This command was implemented on the Cisco 1751, Cisco 2600 series and Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.2(13)T
This command was integrated into Cisco IOS Release 12.2(13)T and implemented on the Cisco 2600XM, the Cisco ICS7750, and the Cisco VG200.
12.4(22)T
Support for IPv6 was added.
Usage Guidelines
Use the destination-pattern command to define the E.164 telephone number for a dial peer. The pattern you configure is used to match dialed digits to a dial peer. The dial peer is then used to complete the call. When a router receives voice data, it compares the called number (the full E.164 telephone number) in the packet header with the number configured as the destination pattern for the voice-telephony peer. The router then strips out the left-justified numbers that correspond to the destination pattern. If you have configured a prefix, the prefix is prepended to the remaining numbers, creating a dial string that the router then dials. If all numbers in the destination pattern are stripped out, the user receives a dial tone. There are areas in the world (for example, certain European countries) where valid telephone numbers can vary in length. Use the optional control character T to indicate that a particular destination-pattern value is a variable-length dial string. In this case, the system does not match the dialed numbers until the interdigit timeout value has expired.
Note
Cisco IOS software does not verify the validity of the E.164 telephone number; it accepts any series of digits as a valid number.
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Cisco IOS Voice Commands: D destination-pattern
Examples
The following example shows configuration of the E.164 telephone number 555-0179 for a dial peer: dial-peer voice 10 pots destination-pattern +5550179
The following example shows configuration of a destination pattern in which the pattern “43” is repeated multiple times preceding the digits “555”: dial-peer voice 1 voip destination-pattern 555(43)+
The following example shows configuration of a destination pattern in which the preceding digit pattern is repeated multiple times: dial-peer voice 2 voip destination-pattern 555%
The following example shows configuration of a destination pattern in which the possible numeric values are between 5550109 and 5550199: dial-peer voice 3 vofr destination-pattern 55501[0-9]9
The following example shows configuration of a destination pattern in which the possible numeric values are between 5550439, 5553439, 5555439, 5557439, and 5559439: dial-peer voice 4 voatm destination-pattern 555[03579]439
The following example shows configuration of a destination pattern in which the digit-by-digit matching is prevented and the entire string is received: dial-peer voice 2 voip destination-pattern 555T
Related Commands
Command
Description
answer-address
Specifies the full E.164 telephone number to be used to identify the dial peer of an incoming call.
dial-peer terminator
Designates a special character to be used as a terminator for variable-length dialed numbers.
incoming called-number (dial peer)
Specifies a digit string that can be matched by an incoming call to associate that call with a dial peer.
prefix
Specifies the prefix of the dialed digits for a dial peer.
timeouts interdigit
Configures the interdigit timeout value for a specified voice port.
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Cisco IOS Voice Commands: D destination-pattern (interface)
destination-pattern (interface) To specify the ISDN directory number for the telephone interface, use the destination-pattern command in interface configuration mode. To disable the specified ISDN directory number, use the no form of this command. destination-pattern isdn no destination-pattern
Syntax Description
isdn
Command Default
A default ISDN directory number is not defined for this interface.
Command Modes
Interface configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Usage Guidelines
Local ISDN directory number assigned by your telephone service provider.
This command is applicable to the Cisco 800 series routers. You must specify this command when creating a dial peer. This command does not work if it is not specified within the context of a dial peer. For information on creating a dial peer, refer to the Cisco 800 Series Routers Software Configuration Guide. Do not specify an area code with the local ISDN directory number.
Examples
The following example specifies 555-0101 as the local ISDN directory number: destination-pattern 5550101
Related Commands
Command
Description
dial-peer voice
Enters dial peer configuration mode, defines the type of dial peer, and defines the tag number associated with a dial peer.
no call-waiting
Disables call waiting.
port (dial peer)
Enables an interface on a PA-4R-DTR port adapter to operate as a concentrator port.
ring
Sets up a distinctive ring for telephones, fax machines, or modems connected to a Cisco 800 series router.
show dial-peer voice
Displays configuration information and call statistics for dial peers.
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Cisco IOS Voice Commands: D detect v54 channel-group
detect v54 channel-group To enable V.54 loopback detection for the command sent from the remote device, use the detect v54 channel-group command in controller configuration mode. To disable the V.54 loopback detection, use the no form of this command. detect v54 channel-group channel-number no detect v54 channel-group channel-number
Syntax Description
channel-number
Command Default
V.54 loopback detection is disabled.
Command Modes
Controller configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 series and Cisco 3600 series.
Channel number from 1 to 24 (T1) or from 1 to 31 (E1).
Usage Guidelines
Use the detect v54 channel-group controller configuration command to enable V.54 loopback detection. The remote device sends a loopup inband payload command sequence in fractional T1 (FT1).
Examples
The following example sets the loopback detection for channel-group 1; then the loopback detection is disabled for channel-group 1. detect v54 channel-group 1 no detect v54 channel-group 1
Related Commands
Command
Description
loopback remote v54 channel-group
Activates a remote V.54 loopback for the channel group on the far end.
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Cisco IOS Voice Commands: D device-id
device-id To identify a gateway associated with a settlement provider, use the device-id command in settlement configuration mode. To reset to the default value, use the no form of this command. device-id number no device-id number
Syntax Description
number
Command Default
The default device ID is 0
Command Modes
Settlement configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Device ID number as provided by the settlement server. Range is from 0 to 2147483647.
Usage Guidelines
It is optional to identify a gateway associated with a settlement provider.
Examples
The following example sets the device ID to 1000: settlement 0 device-id 1000
Related Commands
Command
Description
customer-id
Identifies a carrier or Internet service provider with the settlement provider.
settlement
Enters settlement configuration mode.
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Cisco IOS Voice Commands: D dhcp interface
dhcp interface To configure an interface type for Dynamic Host Configuration Protocol (DHCP) provisioning of Session Initiation Protocol (SIP) parameters, use the dhcp interface command in SIP user-agent configuration mode. dhcp interface type number
Syntax Description
type
Type of interface to be configured.
number
Port, connector, or interface card number. Note
The number format varies depending on the network module or line card type and the router’s chassis slot it is installed in. The numbers are assigned at the factory at the time of installation or when they are added to a system; they can be displayed with the show interfaces command.
Command Default
No interface type is configured for DHCP provisioning of SIP parameters.
Command Modes
SIP user-agent configuration (sip-ua)
Command History
Release
Modification
12.4(22)YB
This command was introduced.
15.0(1)M
This command was integrated in Cisco IOS Release 15.0(1)M.
Usage Guidelines
Multiple interfaces on the Cisco Unified Border Element can be configured with DHCP. The dhcp interface command specifies which one is the DHCP interface used with SIP. This command does not have a no form. Table 17 displays the keywords that represent the types of interfaces that can be configured with the dhcp interface command. Replace the type argument with the appropriate keyword from the table. Table 17
Interface Type Keywords
Keyword
Interface Type
ethernet
Ethernet IEEE 802.3 interface.
fastethernet
100-Mbps Ethernet interface. In RITE configuration mode, specifies the outgoing (monitored) interface for exported IP traffic.
gigabitethernet
1000-Mbps Ethernet interface.
tengigabitethernet
10-Gigabit Ethernet interface.
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Cisco IOS Voice Commands: D dhcp interface
Examples
The following example configures the Gigabit Ethernet interface of slot 0 port 0 as the DHCP interface for DHCP provisioning of SIP parameters: Router> enable Router# configure terminal Router(config)# interface gigabitethernet 0/0 Router(config-if)# ip address dhcp Router(config-if)# sip-ua Router(sip-ua)# dhcp interface gigabitethernet 0/0
Related Commands
Command
Description
show interfaces
Displays information about interfaces.
sip-ua
Enters SIP user-agent configuration mode.
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Cisco IOS Voice Commands: D dial-control-mib
dial-control-mib To specify attributes for the call history table, use the dial-control-mib command in global configuration mode. To restore the default maximum size or retention time of the call history table, use the no form of this command. dial-control-mib {max-size number | retain-timer number} no dial-control-mib {max-size number | retain-timer number}
Syntax Description
max-size number
Specifies the maximum size of the call history table. Range is from 0 to 1200 table entries. Note
retain-timer number
Specifying a value of 0 prevents any further entries from being added to the table. Any existing table entries will be preserved for the duration specified with the retain-timer keyword.
Specifies the duration, in minutes, for entries to remain in the call history table. Range is from 0 to 35791. Note
Specifying a value of 0 prevents any further table entries from being retained, but does not affect any timer currently in effect. Therefore, any existing table entries will remain for the duration previously specified with the retain-timer keyword.
Command Default
The default call history table length is 50 table entries. The default retain timer is 15 minutes.
Command Modes
Global configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
12.0(1)XA
This command was first applied to the CDR feature on the Cisco MC3810.
12.0(2)T
The command was integrated into Cisco IOS Release 12.0(2)T.
12.3T
The maximum value for the number argument following the max-size keyword was increased to 1200 entries.
12.3(8)T
The maximum value of the number argument following the retain-timer keyword was decreased to 35791 minutes.
Examples
The following example configures the call history table to hold 400 entries, with each entry remaining in the table for 10 minutes: dial-control-mib max-size 400 dial-control-mib retain-timer 10
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: D dial-peer cor custom
dial-peer cor custom To specify that named class of restrictions (COR) apply to dial peers, use the dial-peer cor custom command in global configuration mode. dial-peer cor custom
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or keywords.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced.
Usage Guidelines
You must use the dial-peer cor custom command and the name command to define the names of capabilities before you can specify COR rules and apply them to specific dial peers. Examples of possible names might include the following: call1900, call527, call9, and call911.
Note
Examples
You can define a maximum of 64 COR names.
The following example defines two COR names: dial-peer cor custom name 900blackhole name CatchAll
Related Commands
Command
Description
name (dial peer cor custom) Provides a name for a custom COR.
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Cisco IOS Voice Commands: D dial-peer cor list
dial-peer cor list To define a class of restrictions (COR) list name, use the dial-peer cor list command in global configuration mode. To remove a previously defined COR list name, use the no form of this command. dial-peer cor list list-name no dial-peer cor list list-name
Syntax Description
list-name
Command Default
No default behavior or keywords.
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced.
List name that is applied to incoming or outgoing calls to specific numbers or exchanges.
Usage Guidelines
A COR list defines a capability set that is used in the COR checking between incoming and outgoing dial peers.
Examples
The following example adds two members to the COR list named list1: dial-peer cor list list1 member 900block member 800_call
Related Commands
Command
Description
dial-peer cor custom
Specifies that named COR apply to dial peers.
member (dial peer cor list)
Adds a member to a dial peer COR list.
name (dial peer cor custom)
Provides a name for a custom COR.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: D dial-peer data
dial-peer data To create a data dial peer and to enter dial-peer configuration mode, use the dial-peer data command in global configuration mode. To remove a data dial peer, use the no form of this command. dial-peer data tag pots no dial-peer data tag
Syntax Description
tag
Specifies the dial-peer identifying number. Range is from 1 to 2147483647.
pots
Specifies an incoming POTS dial peer.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.2(13)T
This command was introduced.
12.4(4)XC
This command was implemented on the Cisco 2600XM series, Cisco 2800 series, Cisco 3700 series, and Cisco 3800 series.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T.
Usage Guidelines
Examples
A data dial peer should be defined only for incoming data calls. The incoming called-number and shutdown commands on the data dial peer are allowed. However, the following POTS dial-peer commands are disabled on a data dial peer: •
answer-address
•
carrier-id
•
destination-pattern
•
information-type
•
port
•
trunk-group-label
The following example is a data dial peer configuration: dial-peer data 100 pots incoming called-number 100
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Cisco IOS Voice Commands: D dial-peer data
The following example is a voice dial peer configuration: dial-peer voice 2001 pots destination-pattern 2001 no digit-strip port 3/1:1
Related Commands
Command
Description
dial-peer search
Optimizes voice or data dial-peer searches.
incoming called-number
Specifies an incoming called number of an MMoIP or POTS dial peer.
shutdown (dial peer)
Changes the administrative state of a selected dial peer from up to down.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: D dial-peer hunt
dial-peer hunt To specify a hunt selection order for dial peers, use the dial-peer hunt command in global configuration mode. To restore the default selection order, use the no form of this command. dial-peer hunt hunt-order-number no dial-peer hunt
Syntax Description
hunt-order-number
A number from 0 to 7 that selects a predefined hunting selection order: 0—Longest match in phone number, explicit preference, random selection. This is the default hunt order number. 1—Longest match in phone number, explicit preference, least recent use. 2—Explicit preference, longest match in phone number, random selection. 3—Explicit preference, longest match in phone number, least recent use. 4—Least recent use, longest match in phone number, explicit preference. 5—Least recent use, explicit preference, longest match in phone number. 6—Random selection. 7—Least recent use.
Command Default
The default is the longest match in the phone number, explicit preference, random selection (hunt order number 0).
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco MC3810, and Cisco AS5300.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
Use the dial-peer hunt dial peer configuration command if you have configured hunt groups. “Longest match in phone number” refers to the destination pattern that matches the greatest number of the dialed digits. “Explicit preference” refers to the preference setting in the dial peer configuration. “Least recent use” refers to the destination pattern that has waited the longest since being selected. “Random selection” weights all of the destination patterns equally in a random selection mode. This command applies to POTS, VoIP, Voice over Frame Relay (VoFR), Voice over ATM (VoATM), and Multimedia Mail over Internet Protocol (MMOIP) dial peers.
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Cisco IOS Voice Commands: D dial-peer hunt
Examples
The following example configures the dial peers to hunt in the following order: (1) longest match in phone number, (2) explicit preference, (3) random selection. dial-peer hunt 0
Related Commands
Command
Description
destination-pattern
Specifies the prefix or the complete telephone number for a dial peer.
preference
Specifies the preferred selection order of a dial peer within a hunt group.
show dial-peer voice
Displays configuration information for dial peers.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: D dial-peer inbound selection sip-trunk
dial-peer inbound selection sip-trunk To enable incoming SIP line-side calls to use the same dial-peer matching rules as SIP trunk-side calls, use the dial-peer inbound selection sip-trunk command in global configuration mode. To revert to the default behavior, use the no form of this command. dial-peer inbound selection sip-trunk no dial-peer inbound selection sip-trunk
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled (SIP line-side and SIP trunk-side calls use different dial-peer matching rules).
Command Modes
Global configuration (config)
Command History
Release
Modification
12.4(11)T2
This command was introduced.
Usage Guidelines
This command applies the same dial-peer matching rules used for calls from SIP trunks to incoming calls from SIP phones (line side). Table 18 shows the rules and the order in which they are applied by default to SIP line-side calls. Table 19 shows the rules and the order in which they are applied to SIP trunk-side calls and to SIP line-side calls when the dial-peer inbound selection sip-trunk command is used. The router compares the dial-peer configuration to the call parameter in its search to match an inbound call to a dial peer. All dial peers are searched based on the first match criteria. The router moves on to the next criteria only if no match is found. Table 18
Dial-Peer Matching Rules for Inbound Calls from SIP Phones (Line Side)
Match Order Cisco IOS Command
Incoming Call Parameter
1
destination-pattern
Calling number
2
answer-address
Calling number
3
incoming called-number
Called number
4
incoming uri request
Request-URI
5
incoming uri to
To URI
6
incoming uri from
From URI
7
carrier-id source
Carrier-is associated with the call
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Cisco IOS Voice Commands: D dial-peer inbound selection sip-trunk
Table 19
Examples
Dial-Peer Matching Rules for Inbound Calls from SIP Trunks
Match Order Cisco IOS Command
Incoming Call Parameter
1
incoming uri request
Request-URI
2
incoming uri to
To URI
3
incoming uri from
From URI
4
incoming called-number
Called number
5
answer-address
Calling number
6
destination-pattern
Calling number
7
carrier-id source
Carrier-is associated with the call
The following example shows SIP line-side calls use the same matching rules as trunk-side calls: dial-peer inbound selection sip-trunk
Related Commands
Command
Description
answer-address
Specifies calling number to match for a dial peer.
destination-pattern
Specifies telephone number to match for a dial peer.
dial-peer voice
Defines a specific dial peer.
incoming called-number
Incoming called number matched to a dial peer.
incoming uri
Specifies the voice class used to match a VoIP dial peer to the uniform resource identifier (URI) of an incoming call.
show dial-peer voice
Displays configuration information for voice dial peers.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: D dial-peer no-match disconnect-cause
dial-peer no-match disconnect-cause To disconnect the incoming ISDN or channel associated signaling (CAS) call when no inbound voice or modem dial peer is matched, use the dial-peer no-match disconnect-cause command in global configuration mode. To restore the default incoming call state (call is forwarded to the dialer), use the no form of this command. dial-peer no-match disconnect-cause cause-code-number no dial-peer no-match disconnect-cause cause-code-number
Syntax Description
cause-code-number
Command Default
The call is forwarded to the dialer to handle as a modem call.
Command Modes
Global configuration
Command History
Release
Modification
12.2(13)T
This command was introduced.
Usage Guidelines
An ISDN cause code number. Range is from 1 to 127.
By default, calls are forwarded to the dialer to handle as a modem call when no inbound dial peer is matched. The dial-peer no-match disconnect-cause command changes that behavior to disconnect the incoming ISDN or CAS calls when no inbound voice or modem dial peer is matched. Refer to the ISDN Cause Values table in the Cisco IOS Debug Command Reference, for a list of ISDN cause codes.
Examples
The following example shows that ISDN cause code 47 has been specified to match inbound voice or modem dial peers: dial-peer no-match disconnect-cause 47
Related Commands
Command
Description
show dial-peer voice
Displays configuration information for dial peers.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: D dial-peer outbound status-check pots
dial-peer outbound status-check pots To check the status of outbound POTS dial peers during call setup and to disallow, for that call, any whose status is down, use the dial-peer outbound status-check pots command in privileged EXEC mode. To disable status checking, use the no form of this command. dial-peer outbound status-check pots no dial-peer outbound status-check pots
Command Default
None
Command Modes
Privileged EXEC
Command History
Release
Modification
12.3
This command was introduced.
Usage Guidelines
Use this command to disallow, during call setup, outbound POTS dial peers (except those for e-phones) whose endpoints (voice ports or trunk groups) are down. When the dial-peer outbound status-check pots command is configured, if the voice-port configured under an outbound POTS dial-peer is down, that dial-peer is excluded while matching the corresponding destination-pattern. Therefore, if there are no other matching outbound POTS dial-peers for the specified destination-pattern, the gateway will disconnect the call with a cause code of 1 (Unallocated/unassigned number), which is mapped to the “404 Not Found” SIP response by default. When the no form of this command is configured, the outbound POTS dial-peer is matched even if the voice-port configured under is down and the gateway disconnects the call with a cause code of 34 (No circuit/channel available), which is mapped to the “503 Service Unavailable” SIP response by default.
Note
“503 Service Unavailable” was the default behavior before the dial-peer outbound status-check pots command was introduced. Users who need the original behavior should configure the no form of this command. Table 20 shows conditions under which an outbound POTS dial peer may be up or down. Table 20
Conditions Under Which an Outbound POTS Dial Peer Is Up or Down
If a Dial Peer’s,,,
And If...
Then the Dial Peer Is...
Operational state is up
Its voice port is up
Up
Its trunk groups and any associated trunks are up Operational state is down
—
Voice port is down Trunk groups are down
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All associated trunks are down
Down
Cisco IOS Voice Commands: D dial-peer outbound status-check pots
To show or verify the status (up or down) of all or selected dial peers, use the show dial-peer voice command.
Examples
The following examples of output for the related show dial-peer voice command show the status of all or selected dial peers. You can use the dial-peer outbound status-check pots command to disallow the outbound POTS dial peers that are down. The following example shows a short summary status for all dial peers. Outbound status is displayed in the OUT STAT field. POTS dial peers 31 and 42 are shown as down. Router# show dial-peer voice summary dial-peer hunt 0 AD TAG TYPE MIN 444 voip up 22 voip up 12 pots up 311 voip up 31 pots up 421 voip up 42 pots up
OPER PREFIX up up up up up up up
PRE PASS OUT FER THRU SESS-TARGET STAT PORT 0 0 syst up 4/0:15 0 syst 0 down 4/1:15 syst ipv4:1.8.56.2 0 down
DEST-PATTERN
5550123 0 5550111 5550199 0
The following example shows the status for dial peer 12. Outbound status is displayed in the Outbound state field. The dial peer is shown as up. Router# show dial-peer voice 12 VoiceEncapPeer12 peer type = voice, information type = voice, description = `', tag = 12, destination-pattern = `5550123', answer-address = `', preference=0, CLID Restriction = None CLID Network Number = `' CLID Second Number sent source carrier-id = `', target carrier-id = `', source trunk-group-label = `', target trunk-group-label = `', numbering Type = `unknown' group = 12, Admin state is up, Operation state is up, Outbound state is up, enable Router# configure terminal Router(config)# voice class resource-group 1 Router(config-class)# periodic-report interval 180
Related Commands
Command
Description
debug rai
Enables debugging for Resource Allocation Indication (RAI).
rai target
Configures the SIP RAI mechanism.
resource (voice)
Configures parameters for monitoring resources, use the resource command in voice-class configuration mode.
show voice class resource-group
Displays the resource group configuration information for a specific resource group or all resource groups.
voice class resource-group
Enters voice-class configuration mode and assigns an identification tag number for a resource group.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P permit hostname (SIP)
permit hostname (SIP) To store hostnames used during validatation of initial incoming INVITE messages, use the permit hostname command in SIP-ua configuration mode. To remove a stored hostname, use the no form of this command. permit hostname dns: domain name no permit hostname
Syntax Description
dns: domain name
Command Modes
SIP-ua configuration
Command History
Release
Modification
12.4(9)T
This command was introduced.
Usage Guidelines
Domain name in DNS format. Domain names can be up to 30 characters in length; domain names exceeding 30 characters will be truncated.
The permit hostname command allows you to specify hostnames in FQDN (fully qualified domain name) format used during validation of incoming initial INVITE messages. The length of the hostname can be up to 30 characters; hostnames exceeding 30 characters will be truncated. You can store up to 10 hostnames by repeating the permit hostname command. Once configured, initial INVITEs with a hostname in the requested Universal Resource Identifier (URI) are compared to the configured list of hostnames. If there is a match, the INVITE is processed; if there is a mismatch, a “400 Bad Request - Invalid Host” is sent, and the call is rejected.
Note
Examples
Before Software Release 12.4(9)T, hostnames in incoming INVITE-request messages were only validated when they were in IPv4 format; now you can specify hostnames in fully qualified domain name (FQDN) format.
The following example show you how to set the hostname to sip.example.com: Router(config)# sip-ua Router(conf-sip-ua)# permit hostname dns:sip.example.com
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Cisco IOS Voice Commands: P phone context
phone context To filter out uniform resource identifiers (URIs) that do not contain a phone-context field that matches the configured pattern, use the phone context command in voice URI class configuration mode. To remove the pattern, use the no form of this command. phone context phone-context-pattern no phone context
Syntax Description
phone-context-pattern
Command Default
No default behavior or values
Command Modes
Voice URI class configuration
Command History
Release
Modification
12.3(4)T
This command was introduced.
Usage Guidelines
Examples
Cisco IOS regular expression pattern to match against the phone context field in a SIP or TEL URI. Can be up to 32 characters.
•
Use this command with at least one other pattern-matching command, such as host, phone number, or user-id; using it alone does not result in any matches on the voice class.
•
You cannot use this command if you use the pattern command in the voice class. The pattern command matches on the entire URI, whereas this command matches only a specific field.
The following example sets a match on the phone context in the URI voice class: voice class uri 10 tel phone number ^408 phone context 555
Related Commands
Command
Description
destination uri
Specifies the voice class to use for matching the destination URI that is supplied by a voice application.
host
Matches a call based on the host field in a SIP URI.
incoming uri
Specifies the voice class used to match a VoIP dial peer to the URI of an incoming call.
pattern
Matches a call based on the entire SIP or TEL URI.
phone number
Matches a call based on the phone number field in a TEL URI.
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Cisco IOS Voice Commands: P phone context
Command
Description
show dialplan incall uri
Displays which dial peer is matched for a specific URI in an incoming voice call.
show dialplan uri
Displays which outbound dial peer is matched for a specific destination URI.
user-id
Matches a call based on the user-id field in the SIP URI.
voice class uri
Creates or modifies a voice class for matching dial peers to calls containing a SIP or TEL URI.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P phone number
phone number To match a call based on the phone-number field in a telephone (TEL) uniform resource identifier (URI), use the phone number command in voice URI class configuration mode. To remove the pattern, use the no form of this command. phone number phone-number-pattern no phone number
Syntax Description
phone-number-pattern
Command Default
No default behavior or values
Command Modes
Voice URI class configuration
Command History
Release
Modification
12.3(4)T
This command was introduced.
Usage Guidelines
Examples
Cisco IOS regular expression pattern to match against the phone-number field in a TEL URI. Can be up to 32 characters.
•
Use this command only in a voice class for TEL URIs.
•
You cannot use this command if you use the pattern command in the voice class. The pattern command matches on the entire URI, whereas this command matches only a specific field.
The following example defines a voice class that matches on the phone number field in a TEL URI: voice class uri r101 tel phone number ^408
Related Commands
Command
Description
debug voice uri
Displays debugging messages related to URI voice classes.
destination uri
Specifies the voice class to use for matching the destination URI that is supplied by a voice application.
incoming uri
Specifies the voice class used to match a VoIP dial peer to the URI of an incoming call.
pattern
Matches a call based on the entire SIP or TEL URI.
phone context
Filters out URIs that do not contain a phone-context field that matches the configured pattern.
voice class uri
Creates or modifies a voice class for matching dial peers to calls containing a SIP or TEL URI.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P pickup direct
pickup direct To define a feature code for a Feature Access Code (FAC) to access Pickup Direct on an analog phone, use the pickup direct command in STC application feature access-code configuration mode. To return the code to its default, use the no form of this command. pickup direct keypad-character no pickup direct
Syntax Description
keypad-character
Character string that can be dialed on a telephone keypad (0-9, *, #). Default: 6. Before Cisco IOS Release 12.4(20)YA, this is a single character. In Cisco IOS Release 12.4(20)YA and later releases, the string can be any of the following: •
A single character (0-9, *, #)
•
Two digits (00-99)
•
Two to four characters (0-9, *, #) and the leading or ending character must be an asterisk (*) or number sign (#)
In Cisco IOS Release 15.0(1)M and later releases, the string can also be any of the following: •
The length of the keypad-character argument was changed to 1 to 4 characters.
12.4(22)T
This command was integrated into Cisco IOS Release 12.4(22)T.
15.0(1)M
This command was modified.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P pickup direct
Usage Guidelines
This command changes the value of the feature code for Pickup Direct from the default (6) to the specified value. In Cisco IOS Release 12.4(20)YA and later releases, if the length of the keypad-character argument is at least two characters and the leading or ending character of the string is an asterisk (*) or a number sign (#), phone users are not required to dial a prefix to access this feature. Typically, phone users dial a feature access code (FAC) consisting of a prefix plus a feature code, for example **6. If the feature code is 78#, the phone user dials only 78#, without the FAC prefix, to access the corresponding feature. In Cisco IOS Release 15.0(1)M and later releases, if the length of the keypad-character argument is three or four digits, phone users are not required to dial a prefix or any special characters to access this feature. Typically, phone users dial a special feature access code (FAC) consisting of a prefix plus a feature code, for example **2. If the feature code is 788, the phone user dials only 788, without the FAC prefix, to access the corresponding feature. In Cisco IOS Release 12.4(20)YA and later releases, if you attempt to configure this command with a value that is already configured for another feature code, a speed-dial code, or the Redial FSD, you receive a message. If you configure a duplicate code, the system implements the first matching feature in the order of precedence shown in the output of the show stcapp feature codes command. In Cisco IOS Release 12.4(20)YA and later releases, if you attempt to configure this command with a value that precludes or is precluded by another FAC, a speed-dial code, or the Redial FSD, you receive a message. If you configure a feature code to a value that precludes or is precluded by another code, the system always executes the call feature with the shortest code and ignores the longer code. For example, #1 will always preclude #12 and #123. You must configure a new value for the precluded code in order to enable phone user access to that feature. To display a list of all FACs, use the show stcapp feature codes command.
Note
Examples
This FAC is not supported by Cisco Unified Communications Manager.
The following example shows how to change the value of the feature code for Pickup Direct from the default (6). This configuration also changes the value of the prefix for all FACs from the default (**) to ##. With this configuration, a phone user must press ##3 on the keypad and then the ringing extension number to pick up an incoming call. Router(config)# stcapp feature access-code Router(config-stcapp-fac)# prefix ## Router(config-stcapp-fac)# pickup direct 3 Router(config-stcapp-fac)# exit
Related Commands
Command
Description
pickup group
Defines a feature code for a feature access code (FAC) to Group Call Pickup from another group.
pickup local
Defines a feature code for a feature access code (FAC) to Group Call Pickup from the local group.
prefix (stcapp-fac)
Defines the prefix for feature access codes (FACs).
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P pickup direct
Command
Description
show stcapp feature codes
Displays all feature access codes (FACs).
stcapp feature access-code
Enables feature access codes (FACs) in STC application and enters STC application feature access-code configuration mode for changing values of the prefix and features codes from the default.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P pickup group
pickup group To define a feature code for a feature access code (FAC) to access Group Call Pickup on an analog phone, use the pickup group command in STC application feature access-code configuration mode. To return the code to its default, use the no form of this command. pickup group keypad-character no pickup group
Syntax Description
keypad-character
Character string that can be dialed on a telephone keypad (0-9, *, #). Default: 4. Before Cisco IOS Release 12.4(20)YA, this is a single character. In Cisco IOS Release 12.4(20)YA and later releases, the string can be any of the following: •
A single character (0-9, *, #)
•
Two digits (00-99)
•
Two to four characters (0-9, *, #) and the leading or ending character must be an asterisk (*) or number sign (#)
The length of the keypad-character argument was changed to 1 to 4 characters.
12.4(22)T
This command was integrated into Cisco IOS Release 12.4(22)T.
Usage Guidelines
This command changes the value of the feature code for Pickup Direct from the default (4) to the specified value. In Cisco IOS Release 12.4(20)YA and later releases, if the length of the keypad-character argument is at least two characters and the leading or ending character of the string is an asterisk (*) or a number sign (#), phone users are not required to dial a prefix to access this feature. Typically, phone users dial a special feature access code (FAC) consisting of a prefix plus a feature code, for example **4. If the feature code is 78#, the phone user dials only 78#, without the FAC prefix, to access the corresponding feature.
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Cisco IOS Voice Commands: P pickup group
In Cisco IOS Release 12.4(20)YA and later releases, if you attempt to configure this command with a value that is already configured for another feature code, a speed-dial code, or the Redial FSD, you receive a message. If you configure a duplicate code, the system implements the first matching feature in the order of precedence shown in the output of the show stcapp feature codes command. In Cisco IOS Release 12.4(20)YA and later releases, if you attempt to configure this command with a value that precludes or is precluded by another feature code, a speed-dial code, or the Redial FSD, you receive a message. If you configure a feature code to a value that precludes or is precluded by another code, the system always executes the call feature with the shortest code and ignores the longer code. For example, #1 will always preclude #12 and #123. You must configure a new value for the precluded code in order to enable phone user access to that feature. To display a list of all FACs, use the show stcapp feature codes command.
Examples
The following example shows how to change the value of the feature code for Pickup Direct from the default (4). This configuration also changes the value of the prefix for all FACs from the default (**) to ##. After these values are configured, a phone user must press ##3 on the keypad, then the pickup-group number for the ringing extension number to pick up the incoming call. Router(config)# stcapp feature access-code Router(config-stcapp-fac)# prefix ## Router(config-stcapp-fac)# pickup direct 3 Router(config-stcapp-fac)# exit
Related Commands
Command
Description
pickup direct
Defines a feature code for a feature access code (FAC) for Direct Call Pickup of a ringing extension number.
pickup local
Defines a feature code for a feature access code (FAC) for Group Call Pickup to pick up an incoming call from the local group.
prefix (stcapp-fac)
Defines the prefix for feature access codes (FACs).
show stcapp feature codes
Displays all feature access codes (FACs).
stcapp feature access-code
Enables feature access codes (FACs) and enters STC application feature access-code configuration mode for changing values of the prefix and features codes from the default.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P pickup local
pickup local To define a a feature code for a Feature Access Code (FAC) to access Group Call Pickup for a local group on an analog phone, use the pickup local command in STC application feature access-code configuration mode. To return the code to its default, use the no form of this command. pickup local keypad-character no pickup local
Syntax Description
keypad-character
Character string that can be dialed on a telephone keypad. Default: 3. Before Cisco IOS Release 12.4(20)YA, this is a single character. In Cisco IOS Release 12.5(20)YA and later releases, the string can be any o the following: •
A single character (0-9, *, #)
•
Two digits (00-99)
•
Two to four characters (0-9, *, #) and the leading or ending character must be an asterisk (*) or number sign (#)
In Cisco IOS Release 15.0(1)M and later releases, the string can also be any of the following: •
The length of the keypad-character argument was changed to 1 to 4 characters.
12.4(22)T
This command was integrated into Cisco IOS Release 12.4(22)T.
15.0(1)M
This command was modified.
Cisco IOS Voice Command Reference
VR-1465
Cisco IOS Voice Commands: P pickup local
Usage Guidelines
This command changes the value of the feature code for Local Group Pickup from the default (3) to the specified value. In Cisco IOS Release 12.4(20)YA and later releases, if the length of the keypad-character argument is at least two characters and the leading or ending character of the string is an asterisk (*) or a number sign (#), phone users are not required to dial a prefix to access this feature. Typically, phone users dial a special feature access code (FAC) consisting of a prefix plus a feature code, for example **3. If the feature code is 78#, the phone user dials only 78#, without the FAC prefix, to access the corresponding feature. In Cisco IOS Release 15.0(1)M and later releases, if the length of the keypad-character argument is three or four digits, phone users are not required to dial a prefix or any special characters to access this feature. Typically, phone users dial a special feature access code (FAC) consisting of a prefix plus a feature code, for example **2. If the feature code is 788, the phone user dials only 788, without the FAC prefix, to access the corresponding feature. In Cisco IOS Release 12.4(20)YA and later releases, if you attempt to configure this command with a value that is already configured for another feature code or speed-dial code, or for the Redial FSD, you receive a message. If you configure a duplicate code, the system implements the first matching feature in the order of precedence shown in the output of the show stcapp feature codes command. In Cisco IOS Release 12.4(20)YA and later releases, if you attempt to configure this command with a value that precludes or is precluded by another feature code or speed-dial code, or by the Redial FSD, you receive a message. If you configure a feature code to a value that precludes or is precluded by another code, the system always executes the call feature with the shortest code and ignores the longer code. For example, #1 will always preclude #12 and #123. You must configure a new value for the precluded code in order to enable phone user access to that feature. To display a list of all FACs, use the show stcapp feature codes command.
Examples
The following example shows how to change the value of the feature code for Pickup Direct from the default (3). This configuration also changes the value of the prefix for all FACs from the default (**) to ##. With this configuration, a phone user must press ##9 on the keypad to pick up an incoming call in the same group as this extension number. Router(config)# stcapp feature access-code Router(config-stcapp-fac)# prefix ## Router(config-stcapp-fac)# pickup local 9 Router(config-stcapp-fac)# exit
Related Commands
Command
Description
pickup direct
Defines a feature code for a feature access code (FAC) for Direct Call Pickup of a ringing extension number.
pickup group
Defines a feature code for a feature access code (FAC) for Group Call Pickup to pick up an incoming call from another group.
prefix (stcapp-fac)
Defines the prefix for feature access codes (FACs).
show stcapp feature codes
Displays all feature access codes (FACs).
stcapp feature access-code
Enables feature access codes (FACs) in STC application and enters STC application feature access-code configuration mode for changing values of the prefix and features codes from the default.
Cisco IOS Voice Command Reference
VR-1466
Cisco IOS Voice Commands: P playout-delay (dial peer)
playout-delay (dial peer) To tune the playout buffer on digital signal processors (DSPs) to accommodate packet jitter caused by switches in the WAN, use the playout-delay command in dial peer configuration mode. To reset the playout buffer to the default, use the no form of this command. playout-delay {fax milliseconds | maximum milliseconds | minimum {default | low | high} | nominal milliseconds} no playout-delay {fax | maximum | minimum | nominal}
Syntax Description
fax milliseconds
Amount of playout delay that the jitter buffer should apply to fax calls, in milliseconds. Range is from 0 to 700. Default is 300.
maximum milliseconds (Adaptive mode only) Upper limit of the jitter buffer, or the highest value to which the adaptive delay is set, in milliseconds. Range is from 40 to 1700, although this value depends on the type of DSP and how the voice card is configured for codec complexity. (See the codec complexity command.) Default is 200. If the voice card is configured for high codec complexity, the highest value that can be configured for maximum for compressed codecs is 250 ms. For medium-complexity codec configurations, the highest maximum value is 150 ms. Voice hardware that does not support the voice card complexity configuration (such as analog voice modules for the Cisco 3600 series router) has an upper limit of 200 ms. minimum
nominal milliseconds
(Adaptive mode only) Lower limit of the jitter buffer, or the lowest value to which the adaptive delay is set, in milliseconds. Values are as follows: •
default—40 ms. Use when there are normal jitter conditions in the network. This is the default.
•
low—10 ms. Use when there are low jitter conditions in the network.
•
high—40 ms. Use when there are high jitter conditions in the network.
Amount of playout delay applied at the beginning of a call by the jitter buffer in the gateway, in milliseconds. In fixed mode, this is also the maximum size of the jitter buffer throughout the call. Range is from 0 to 1500, although this value depends on the type of DSP and how the voice card is configured for codec complexity. Default is 60. For non-conference calls when you are using DSPware version 4.1.33 or a later version, the following values are allowed. •
If the voice card is configured for high codec complexity, the highest value that can be configured for the nominal keyword for compressed codecs is 200 ms.
•
For medium-complexity codec configurations, the highest nominal value is 150 ms.
Cisco IOS Voice Command Reference
VR-1467
Cisco IOS Voice Commands: P playout-delay (dial peer)
nominal milliseconds (continued)
For conference calls when you are using DSPware version 4.1.33 or a later version, the following values are allowed: •
The first decoder stream can be assigned a nominal value as high as 200 ms (high-complexity codec) or 150 ms (medium-complexity codec).
•
Subsequent decoder streams are limited to the highest nominal value of 150 ms (high-complexity) or 80 ms (medium-complexity).
When the playout-delay mode is configured for fixed operation and setting the expected jitter buffer size with the nominal value, the minimum effective value for the playout delay will depend on the codec in use and the configured minimum value. •
When the playout-delay minimum low is configured the minimum actual jitter buffer size will be 30ms even when setting the nominal to a value lower than 30msec.
•
When the playout-delay minimum default, the minimum jitter buffer size when running in fixed mode will be 60ms.
When fixed mode is configured, there is a 10msec added to the nominal value when setting the jitter buffer when configured for G.729 and a 5ms added using G.711 Voice hardware that does not support the voice-card complexity configuration (such as analog voice modules for the Cisco 3600 series router) has an upper limit of 200 ms for the first decoder stream and 150 ms for subsequent decoder streams. Note
With DSPware versions earlier than 4.1.33, the highest nominal value that can be configured is 150 ms for high-complexity codec configurations and analog modules. The highest nominal value for medium-complexity codec configurations is 80 ms.
This command was implemented on the Cisco 2600 series and Cisco 3600 series.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.1(3)XI
This command was implemented on the Cisco ICS7750.
Cisco IOS Voice Command Reference
VR-1468
Cisco IOS Voice Commands: P playout-delay (dial peer)
Usage Guidelines
Release
Modification
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T. Support for dial peer configuration mode was added on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco MC3810, Cisco AS5200, Cisco AS5300, Cisco AS5400, and Cisco AS5800. The minimum keyword was introduced.
12.2(13)T
The fax keyword was introduced.
12.2(13)T8
DSPware version 4.1.33 was implemented.
Before Cisco IOS Release 12.1(5)T, this command was used in voice-port configuration mode. For Cisco IOS Release 12.1(5)T and later releases, in most cases playout delay should be configured in dial-peer configuration mode on the Voice over IP (VoIP) dial peer that is on the receiving end of the voice traffic that is to be buffered. This dial peer senses network conditions and relays them to the DSPs, which adjust the jitter buffer as necessary. When multiple applications are configured on the gateway, playout delay should be configured in dial-peer configuration mode. When there are numerous dial peers to configure, it might be simpler to configure playout delay on a voice port. If conflicting playout-delay values have been configured on a voice port and on a dial peer, the dial-peer configuration takes precedence. Playout delay is the amount of time that elapses between the time at which a voice packet is received at the jitter buffer on the DSP and the time at which it is played out to the codec. In most networks with normal jitter conditions, the defaults are adequate and you will not need to configure this command. In situations in which you want to improve voice quality by reducing jitter or you want to reduce network delay, you can configure playout-delay parameters. The parameters are slightly different for each of the two playout-delay modes, adaptive and fixed (see the playout-delay mode command). In adaptive mode, the average delay for voice packets varies depending on the amount of interarrival variation that packets have as the call progresses. The jitter buffer grows and shrinks to compensate for jitter and to keep voice packets playing out smoothly, within the maximum and minimum limits that have been configured. The maximum limit establishes the highest value to which the adaptive delay is set. The minimum limit is the low-end threshold for the delay of incoming packets by the adaptive jitter buffer. Algorithms in the DSPs that control the growth and shrinkage of the jitter buffer are weighted toward the improvement of voice quality at the expense of network delay: jitter buffer size increases rapidly in response to spikes in network transmissions and decreases slowly in response to reduced congestion. In fixed mode, the nominal value is the amount of playout delay applied at the beginning of a call by the jitter buffer in the gateway and is also the maximum size of the jitter buffer throughout the call. As a general rule, if there is excessive breakup of voice due to jitter with the default playout-delay settings, increase playout delay times. If your network is small and jitter is minimal, decrease playout-delay times for a smaller overall delay. When there is bursty jitter in the network, voice quality can be degraded even though the jitter buffer is actually adjusting the playout delay correctly. The constant readjustment of playout delay to erratic network conditions causes voice quality problems that are usually alleviated by increasing the minimum playout delay-value in adaptive mode or by increasing the nominal delay for fixed mode. Use the show call active voice command to display the current delay, as well as high- and low-water marks for delay during a call. Other fields that can help determine the size of a jitter problem are ReceiveDelay, GapFillWith..., LostPackets, EarlyPackets, and LatePackets. The following is sample output from the show call active voice command:
Cisco IOS Voice Command Reference
VR-1469
Cisco IOS Voice Commands: P playout-delay (dial peer)
VOIP: ConnectionId[0xECDE2E7B 0xF46A003F 0x0 0x47070A4] IncomingConnectionId[0xECDE2E7B 0xF46A003F 0x0 0x47070A4] RemoteIPAddress=192.168.100.101 RemoteUDPPort=18834 RoundTripDelay=26 ms SelectedQoS=best-effort tx_DtmfRelay=inband-voice FastConnect=TRUE Separate H245 Connection=FALSE H245 Tunneling=FALSE SessionProtocol=cisco SessionTarget= OnTimeRvPlayout=417000 GapFillWithSilence=850 ms GapFillWithPrediction=2590 ms GapFillWithInterpolation=0 ms GapFillWithRedundancy=0 ms HiWaterPlayoutDelay=70 ms LoWaterPlayoutDelay=29 ms ReceiveDelay=39 ms LostPackets=0 EarlyPackets=0 LatePackets=86
Examples
The following example uses default adaptive mode with a minimum playout delay of 10 ms and a maximum playout delay of 60 ms on VoIP dial peer 80. The size of the jitter buffer is adjusted up and down on the basis of the amount of jitter that the DSP finds, but is never smaller than 10 ms and never larger than 60 ms. dial-peer 80 voip playout-delay minimum low playout-delay maximum 60
Related Commands
Command
Description
codec complexity
Specifies call density and codec complexity based on the codec standard you are using.
playout-delay (voice-port)
Tunes the playout buffer to accommodate packet jitter caused by switches in the WAN.
playout-delay mode
Selects fixed or adaptive mode for the jitter buffer on DSPs.
show call active voice
Displays active call information for voice calls.
Cisco IOS Voice Command Reference
VR-1470
Cisco IOS Voice Commands: P playout-delay (voice-port)
playout-delay (voice-port) To tune the playout buffer to accommodate packet jitter caused by switches in the WAN, use the playout-delay command in voice-port configuration mode. To reset the playout buffer to the default, use the no form of this command. playout-delay {fax | maximum | nominal} milliseconds no playout-delay {fax | maximum | nominal}
Syntax Description
fax milliseconds
Amount of playout delay that the jitter buffer should apply to fax calls, in milliseconds. Range is from 0 to 700. Default is 300.
maximum milliseconds Delay time that the digital signal processor (DSP) allows before starting to discard voice packets, in milliseconds. Range is from 40 to 320. Default is 160. nominal milliseconds
Initial (and minimum allowed) delay time that the DSP inserts before playing out voice packets, in milliseconds. Range is from 40 to 200. Default is 80.
This command was implemented on the Cisco 2600 series and Cisco 3600 series.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.2(13)T
The fax keyword was added.
Usage Guidelines
If there is excessive breakup of voice due to jitter with the default playout delay settings, increase the delay times. If your network is small and jitter is minimal, decrease the delay times to reduce delay. Before Cisco IOS Release 12.1(5)T, the playout-delay command was configured in voice-port configuration mode. For Cisco IOS Release 12.1(5)T and later releases, in most cases playout delay should be configured in dial-peer configuration mode on the Voice over IP (VoIP) dial peer that is on the receiving end of the voice traffic that is to be buffered. This dial peer senses network conditions and relays them to the DSPs, which adjust the jitter buffer as necessary. When multiple applications are configured on the gateway, playout delay should be configured in dial-peer configuration mode. When there are numerous dial peers to configure, it might be simpler to configure playout delay on a voice port. If conflicting playout-delay values have been configured on a voice port and on a dial peer, the dial-peer configuration takes precedence.
Cisco IOS Voice Command Reference
VR-1471
Cisco IOS Voice Commands: P playout-delay (voice-port)
Playout delay is the amount of time that elapses between the time at which a voice packet is received at the jitter buffer on the DSP and the time at which it is played out to the codec. In most networks with normal jitter conditions, the defaults are adequate and you will not need to configure the playout-delay command. In situations in which you want to improve voice quality by reducing jitter or you want to reduce network delay, you can configure playout-delay parameters. The parameters are slightly different for each of the two playout-delay modes, adaptive and fixed (see the playout-delay mode command). In adaptive mode, the average delay for voice packets varies depending on the amount of interarrival variation that packets have as the call progresses. The jitter buffer grows and shrinks to compensate for jitter and to keep voice packets playing out smoothly, within the maximum and minimum limits that have been configured. The maximum limit establishes the highest value to which the adaptive delay will be set. The minimum limit is the low-end threshold for incoming packet delay that is created by the adaptive jitter buffer. Algorithms in the DSPs that control the growth and shrinkage of the jitter buffer are weighted toward the improvement of voice quality at the expense of network delay: jitter buffer size increases rapidly in response to spikes in network transmissions and decreases slowly in response to reduced congestion. In fixed mode, the nominal value is the amount of playout delay applied at the beginning of a call by the jitter buffer in the gateway and is also the maximum size of the jitter buffer throughout the call. As a general rule, if there is excessive breakup of voice due to jitter with the default playout-delay settings, increase playout-delay times. If your network is small and jitter is minimal, decrease playout-delay times for a smaller overall delay. When there is bursty jitter in the network, voice quality can be degraded even though the jitter buffer is actually adjusting the playout delay correctly. The constant readjustment of playout delay to erratic network conditions causes voice quality problems that are usually alleviated by increasing the minimum playout-delay value in adaptive mode or by increasing the nominal delay for fixed mode.
Note
The minimum limit for playout delay is configured using the playout-delay (dial peer) command. Use the show call active voice command to display the current delay, as well as high- and low-water marks for delay during a call. Other fields that can help determine the size of a jitter problem are GapFillWith..., ReceiveDelay, LostPackets, EarlyPackets, and LatePackets. The following is sample output from the show call active voice command: VOIP: ConnectionId[0xECDE2E7B 0xF46A003F 0x0 0x47070A4] IncomingConnectionId[0xECDE2E7B 0xF46A003F 0x0 0x47070A4] RemoteIPAddress=192.168.100.101 RemoteUDPPort=18834 RoundTripDelay=26 ms SelectedQoS=best-effort tx_DtmfRelay=inband-voice FastConnect=TRUE Separate H245 Connection=FALSE H245 Tunneling=FALSE SessionProtocol=cisco SessionTarget= OnTimeRvPlayout=417000 GapFillWithSilence=850 ms GapFillWithPrediction=2590 ms GapFillWithInterpolation=0 ms GapFillWithRedundancy=0 ms HiWaterPlayoutDelay=70 ms LoWaterPlayoutDelay=29 ms ReceiveDelay=39 ms
Cisco IOS Voice Command Reference
VR-1472
Cisco IOS Voice Commands: P playout-delay (voice-port)
LostPackets=0 EarlyPackets=0 LatePackets=86
Examples
The following example sets nominal playout delay to 80 ms and maximum playout delay to 160 ms on voice port 1/0/0: voice-port 1/0/0 playout-delay nominal 80 playout-delay maximum 160
Related Commands
Command
Description
playout-delay (dial peer)
Tunes the playout buffer on DSPs to accommodate packet jitter caused by switches in the WAN.
playout-delay mode
Selects fixed or adaptive mode for playout delay from the jitter buffer on digital signal processors.
show call active
Shows active call information for voice calls or fax transmissions in progress.
vad
Enables voice activity detection.
Cisco IOS Voice Command Reference
VR-1473
Cisco IOS Voice Commands: P playout-delay mode (dial peer)
playout-delay mode (dial peer) To select fixed or adaptive mode for playout delay from the jitter buffer on digital signal processors (DSPs), use the playout-delay mode command in dial peer configuration mode. To reset to the default, use the no form of this command. playout-delay mode {adaptive | fixed} no playout-delay mode
Syntax Description
adaptive
Jitter buffer size and amount of playout delay are adjusted during a call, on the basis of current network conditions.
fixed
Jitter buffer size does not adjust during a call; a constant playout delay is added.
Command Default
Adaptive jitter buffer size
Command Modes
Dial peer configuration
Command History
Release
Modification
12.1(5)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco MC3810, and Cisco ICS 7750. The no-timestamps keyword was removed.
Usage Guidelines
Before Cisco IOS Release 12.1(5)T, this command was used only in voice-port configuration mode. For Cisco IOS Release 12.1(5)T and later releases, in most cases playout delay should be configured in dial peer configuration mode on the VoIP dial peer that is on the receiving end of the voice traffic that is to be buffered. This dial peer senses network conditions and relays them to the DSPs, which adjust the jitter buffer as necessary. When multiple applications are configured on the gateway, playout delay should be configured in dial peer configuration mode.
Tip
When there are numerous dial peers to configure, it might be simpler to configure playout delay on a voice port. If conflicting playout delay values have been configured on a voice port and on a dial peer, the dial peer configuration takes precedence. In most networks with normal jitter conditions, the default is adequate and you do not need to configure this command. The default is adaptive mode, in which the average delay for voice packets varies depending on the amount of interarrival variation that packets have as the call progresses. The jitter buffer grows and shrinks to compensate for jitter and to keep voice packets playing out smoothly, within the maximum and minimum limits that have been configured.
Cisco IOS Voice Command Reference
VR-1474
Cisco IOS Voice Commands: P playout-delay mode (dial peer)
Select fixed mode only when you understand your network conditions well, and when you have a network with very poor quality of service (QoS) or when you are interworking with a media server or similar transmission source that tends to create a lot of jitter at the transmission source. In most situations it is better to configure adaptive mode and let the DSP size the jitter buffer according to current conditions.
Examples
The following example sets adaptive playout-delay mode with a high (80 ms) minimum delay on a VoIP dial peer 80: dial-peer 80 voip playout-delay mode adaptive playout-delay minimum high
Related Commands
Command
Description
playout-delay
Tunes the jitter buffer on DSPs for playout delay of voice packets.
show call active voice
Displays active call information for voice calls.
Cisco IOS Voice Command Reference
VR-1475
Cisco IOS Voice Commands: P playout-delay mode (voice-port)
playout-delay mode (voice-port) To select fixed or adaptive mode for playout delay from the jitter buffer on digital signal processors (DSPs), use the playout-delay mode command in voice port configuration mode. To reset to the default, use the no form of this command. playout-delay mode {adaptive | fixed} no playout-delay mode
Syntax Description
adaptive
Jitter buffer size and amount of playout delay are adjusted during a call, on the basis of current network conditions.
fixed
Jitter buffer size does not adjust during a call; a constant playout delay is added.
Command Default
Adaptive jitter buffer size
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810.
12.0(7)XK
This command was implemented on the Cisco 2600 and Cisco 3600 series.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.1(3)XI
This command was implemented on the Cisco ICS 7750. The keyword mode was introduced.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T and the no-timestamps keyword was removed.
Usage Guidelines
Before Cisco IOS Release 12.1(5)T, this command was used only in voice-port configuration mode. For Cisco IOS Release 12.1(5)T and later releases, in most cases playout delay should be used in dial peer configuration mode on the VoIP dial peer that is on the receiving end of the voice traffic that is to be buffered. This dial peer senses network conditions and relays them to the DSPs, which adjust the jitter buffer as necessary. When multiple applications are configured on the gateway, playout delay should be configured in dial peer configuration mode.
Tip
When there are numerous dial peers to configure, it might be simpler to configure playout delay on a voice port. If conflicting playout delay values have been configured on a voice port and on a dial peer, the dial peer configuration takes precedence. In most networks with normal jitter conditions, the default is adequate and you do not need to configure the playout-delay mode command.
Cisco IOS Voice Command Reference
VR-1476
Cisco IOS Voice Commands: P playout-delay mode (voice-port)
The default is adaptive mode, in which the average delay for voice packets varies depending on the amount of interarrival variation that packets have as the call progresses. The jitter buffer grows and shrinks to compensate for jitter and to keep voice packets playing out smoothly, within the maximum and minimum limits that have been configured. Select fixed mode only when you understand your network conditions well, and when you have a network with very poor quality of service (QoS) or when you are interworking with a media server or similar transmission source that tends to create a lot of jitter at the transmission source. In most situations it is better to configure adaptive mode and let the DSP size the jitter buffer according to current conditions.
Examples
The following example sets fixed mode on a Cisco 3640 voice port with a nominal delay of 80 ms. voice-port 1/1/0 playout-delay mode fixed playout-delay nominal 80
Related Commands
Command
Description
playout-delay
Tunes the jitter buffer on DSPs for playout delay of voice packets.
show call active voice
Displays active call information for voice calls.
Cisco IOS Voice Command Reference
VR-1477
Cisco IOS Voice Commands: P port (Annex G neighbor BE)
port (Annex G neighbor BE) To configure the port number of the neighbor that is used for exchanging Annex G messages, use the port command in Annex G Neighbor BE configuration mode. To remove the port number, use the no form of this command. port neighbor-port no port
Syntax Description
neighbor-port
Defaults
2099
Command Modes
Annex G Neighbor BE configuration
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T. This command is supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
Port number of the neighbor. This number is used for exchanging Annex G messages. The default port number is 2099.
Usage Guidelines
When cofiguring the no port command the neighbor-port argument is not used.
Examples
The following example sets a neighbor BE to port number 2010. Router(config-annexg-neigh)# port 2010
Cisco IOS Voice Command Reference
VR-1478
Cisco IOS Voice Commands: P port (Annex G neighbor BE)
Related Commands
Command
Description
advertise (annex g)
Controls the types of descriptors that the BE advertises to its neighbors.
cache
Configures the local BE to cache the descriptors received from its neighbors.
id
Configures the local ID of the neighboring BE.
query-interval
Configures the interval at which the local BE will query the neighboring BE.
Cisco IOS Voice Command Reference
VR-1479
Cisco IOS Voice Commands: P port (dial-peer)
port (dial-peer) To associate a dial peer with a specific voice port, use the port command in dial peer configuration mode. To cancel this association, use the no form of this command. Cisco 1750 and Cisco 3700 Series
port slot-number/port no port slot-number/port Cisco 2600 Series, Cisco 3600 Series, and Cisco 7200 Series
port {slot-number/subunit-number/port | slot/port:ds0-group-number} no port {slot-number/subunit-number/port | slot/port:ds0-group-number} Cisco AS5300 and Cisco AS5800
port controller-number:D no port controller-number:D Cisco uBR92x Series
port slot/subunit/port no port slot/subunit/port
Syntax Description
Cisco 1750 and Cisco 3700 Series
slot-number
Number of the slot in the router in which the voice interface card (VIC) is installed. Valid entries are from 0 to 2, depending on the slot in which the VIC has been installed.
port
Voice port number. Valid entries are 0 and 1.
Cisco 2600 Series, Cisco 3600 Series, and Cisco 7200 Series
slot-number
Number of the slot in the router in which the VIC is installed. Valid entries are from 0 to 3, depending on the slot in which it has been installed.
subunit-number
Subunit on the VIC in which the voice port is located. Valid entries are 0 and 1.
port
Voice port number. Valid entries are 0 and 1.
slot
Router location in which the voice port adapter is installed. Valid entries are 0 and 3.
port
Voice interface card location. Valid entries are 0 and 3.
ds0-group-number
The DS0 group number. Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card.
Cisco IOS Voice Command Reference
VR-1480
Cisco IOS Voice Commands: P port (dial-peer)
Cisco AS5300
controller-number
The T1 or E1 controller.
:D
Indicates the D channel associated with the ISDN PRI.
Cisco uBR92x series
slot/subunit/port
The analog voice port. Valid entries for the slot/subunit/port are as follows: •
slot—A router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
•
subunit—A VIC in which the voice port is located. Valid entries are 0 and 1. (The VIC fits into the voice network module.)
•
port—An analog voice port number. Valid entries are 0 and 1.
Command Default
No port is configured.
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series.
11.3(3)T
This command was implemented on the Cisco 2600 series.
11.3(1)MA
This command was implemented on the Cisco MC3810.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T and implemented on the Cisco AS5300.
12.0(4)T
This command was implemented on the Cisco uBR924.
12.0(7)T
This command was implemented on the Cisco AS5800.
12.2(8)T
This command was implemented on the following platforms: Cisco 1751, Cisco 3725, and Cisco 3745.
12.2(13)T
This command was integrated into Cisco IOS Release 12.2(13)T. This command does not support the extended echo canceller (EC) feature on the Cisco AS5300 or the Cisco AS5800.
12.4(22)T
Support for IPv6 was added.
Usage Guidelines
This command enables calls that come from a telephony interface to select an incoming dial peer and for calls that come from the VoIP network to match a port with the selected outgoing dial peer. This command applies only to POTS peers.
Note
This command does not support the extended EC feature on the Cisco AS5300.
Cisco IOS Voice Command Reference
VR-1481
Cisco IOS Voice Commands: P port (dial-peer)
Examples
The following example associates POTS dial peer 10 with voice port 1, which is located on subunit 0 and accessed through port 0: dial-peer voice 10 pots port 1/0/0
The following example associates POTS dial peer 10 with voice port 0:D: dial-peer voice 10 pots port 0:D
The following example associates POTS dial peer 10 with voice port 1/0/0:D (T1 card): dial-peer voice 10 pots port 1/0/0:D
Related Commands
Command
Description
prefix
Specifies the prefix of the dialed digits for a dial peer.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P port (MGCP profile)
port (MGCP profile) To associate a voice port with the Media Gateway Control Protocol (MGCP) profile that is being configured, use the port command in MGCP profile configuration mode. To disassociate the voice port from the profile, use the no form of this command. port port-number no port port-number
Syntax Description
port-number
Command Default
No default behavior or values
Command Modes
MGCP profile configuration
Command History
Release
Modification
12.2(2)XA
This command was introduced as the voice-port (MGCP profile) command.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T.
12.2(8)T
This command was renamed the port (MGCP profile) command.
Usage Guidelines
Voice port or DS0-group number to be used as an MGCP endpoint associated with an MGCP profile.
This command is used when values for an MGCP profile are configured. This command associates a voice port with the MGCP profile that is being defined. To associate multiple voice ports with a profile, repeat this command with different voice port arguments. This command is not used when the default MGCP profile is configured because the values in the default profile configuration apply to all parameters that have not been otherwise configured for a user-defined MGCP profile.
Examples
The following example associates an analog voice port with an MGCP profile on a Cisco uBR925 platform: Router(config)# mgcp profile ny110ca Router(config-mgcp-profile)# port 0
Related Commands
Command
Description
mgcp
Starts and allocates resources for the MGCP daemon.
mgcp profile
Initiates MGCP profile mode to create and configure a named MGCP profile associated with one or more endpoints or to configure the default profile.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P port (supplementary-service)
port (supplementary-service) To enter the supplementary-service voice-port configuration mode for associating a voice port with STC application supplementary-service features, use the port command in supplementary-service configuration mode. To cancel the association, use the no form of this command. port port no port port
This command was integrated into Cisco IOS Release 12.4(22)T.
Location of port in Cisco ISR or Cisco VG224 Analog Phone Gateway. Syntax is platform-dependent; type ? to determine.
Usage Guidelines
This command associates an analog FXS port to STC application supplementary-service features being configured.
Examples
The following example shows how to enable Hold/Resume on analog endpoints connected to port 2/0 of a Cisco VG224. Router(config)# stcapp supplementary-services Router(config-stcapp-suppl-serv)# port 2/0 Router(config-stcapp-suppl-serv-port)# hold-resume Router(config-stcapp-suppl-serv-port)# end
Related Commands
Command
Description
hold-resume
Enables Hold/Resume in Feature mode on the port being configured.
Cisco IOS Voice Command Reference
VR-1484
Cisco IOS Voice Commands: P port media
port media To specify the serial interface to which the local video codec is connected for a local video dial peer, use the port media command in video dial peer configuration mode. To remove any configured locations from the dial peer, use the no form of this command. port media interface no port media
Syntax Description
interface
Command Default
No interface is specified
Command Modes
Video dial peer configuration
Command History
Release
Modification
12.0(5)XK
This command was introduced for ATM video dial peer configuration on the Cisco MC3810.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Examples
Serial interface to which the local codec is connected. Valid entries are 0 and 1.
The following example specifies serial interface 0 as the specified interface for the codec local video dial peer 10: dial-peer video 10 videocodec port media Serial0
Related Commands
Command
Description
port signal
Specifies the slot location of the VDM and the port location of the EIA/TIA-366 interface for signaling.
show dial-peer video
Displays dial peer configuration.
Cisco IOS Voice Command Reference
VR-1485
Cisco IOS Voice Commands: P port signal
port signal To specify the slot location of the video dialing module (VDM) and the port location of the EIA/TIA-366 interface for signaling for a local video dial peer, use the port signal command in video dial peer configuration mode. To remove any configured locations from the dial peer, use the no form of this command. port signal slot/port no port signal
Syntax Description
slot
Slot location of the VDM. Valid values are 1 and 2.
port
Port location of the EIA/TIA-366 interface.
Command Default
No locations are specified
Command Modes
Video dial peer configuration
Command History
Release
Modification
12.0(5)XK
This command was introduced for ATM video dial peer configuration on the Cisco MC3810.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Examples
The following example sets up the VDM and EIA/TIA-366 interface locations for the local video dial peer designated as 10: dial-peer video 10 videocodec port signal 1/0
Related Commands
Command
Description
port media
Specifies the serial interface to which the local video codec is connected.
show dial-peer video
Displays dial peer configuration.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P pots call-waiting
pots call-waiting To enable the local call-waiting feature, use the global configuration pots call-waiting command in global configuration mode. To disable the local call-waiting feature, use the no form of this command. pots call-waiting {local | remote} no pots call-waiting {local | remote}
Syntax Description
local
Enable call waiting on a local basis for the routers.
remote
Rely on the network provider service instead of the router to hold calls.
Command Default
Remote, in which case the call- holding pattern follows the settings of the service provider rather than those of the router.
Command Modes
Global configuration
Command History
Release
Modification
12.1.(2)XF
This command was introduced on the Cisco 800 series.
Usage Guidelines
To display the call-waiting setting, use the show running-config or show pots status command. The ISDN call waiting service is used if it is available on the ISDN line connected to the router even if local call waiting is configured on the router. That is, if the ISDN line supports call waiting, the local call waiting configuration on the router is ignored.
Examples
The following example enables local call waiting on a router: pots call-waiting local
Related Commands
Command
Description
call-waiting
Configures call waiting for a specific dial peer.
show pots status
Displays the settings of the physical characteristics and other information on the telephone interfaces of a Cisco 800 series router.
Cisco IOS Voice Command Reference
VR-1487
Cisco IOS Voice Commands: P pots country
pots country To configure your connected telephones, fax machines, or modems to use country-specific default settings for each physical characteristic, use the pots country command in global configuration mode. To disable the use of country-specific default settings, use the no form of this command. pots country country no pots country country
Syntax Description
country
Command Default
A default country is not defined.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Usage Guidelines
Country in which your router is located.
This command applies to the Cisco 800 series routers. If you need to change a country-specific default setting of a physical characteristic, you can use the associated command listed in the “Related Commands” section. Enter the pots country ? command to get a list of supported countries and the code you must enter to indicate a particular country.
Examples
The following example specifies that the devices connected to the telephone ports use default settings specific to Germany for the physical characteristics: pots country de
Related Commands
Command
Description
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed on your connected telephones, fax machines, or modems.
pots disconnect-supervision
Specifies how a Cisco 800 series router notifies the connected telephones, fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if telephones, fax machines, or modems connected to a Cisco 800 series router fail to detect that a calling party has disconnected.
pots distinctive-ring-guard-time
Specifies the delay in which a telephone port can be rung after a previous call is disconnected (Cisco 800 series routers).
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P pots country
Command
Description
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800 series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and other information on the telephone interfaces on a Cisco 800 series router.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P pots dialing-method
pots dialing-method To specify how the router collects and sends digits dialed on your connected telephones, fax machines, or modems, use the pots dialing-method command in global configuration mode. To disable the specified dialing method, use the no form of this command. pots dialing-method {overlap | enblock} no pots dialing-method {overlap | enblock}
Syntax Description
overlap
The router sends each digit dialed in a separate message.
enblock
The router collects all digits dialed and sends the digits in one message.
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Usage Guidelines
This command applies to Cisco 800 series routers. To interrupt the collection and transmission of dialed digits, enter a pound sign (#), or stop dialing digits until the interdigit timer runs out (10 seconds).
Examples
The following example specifies that the router uses the enblock dialing method: pots dialing-method enblock
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800 series router to use country-specific default settings for each physical characteristic.
pots disconnect-supervision
Specifies how a Cisco 800 series router notifies the connected telephones, fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if telephones, fax machines, or modems connected to a Cisco 800 series router fail to detect that a calling party has disconnected.
Cisco IOS Voice Command Reference
VR-1490
Cisco IOS Voice Commands: P pots dialing-method
Command
Description
pots distinctive-ring-guard-time
Specifies the delay in which a telephone port can be rung after a previous call is disconnected (Cisco 800 series routers).
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800 series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and other information on the telephone interfaces on a Cisco 800 series router.
Cisco IOS Voice Command Reference
VR-1491
Cisco IOS Voice Commands: P pots disconnect-supervision
pots disconnect-supervision To specify how a router notifies the connected telephones, fax machines, or modems when the calling party has disconnected, use the pots disconnect-supervision command in global configuration mode. To disable the specified disconnect method, use the no form of this command. pots disconnect-supervision {osi | reversal} no pots disconnect-supervision {osi | reversal}
Syntax Description
osi
Open switching interval (OSI) is the duration for which DC voltage applied between tip and ring conductors of a telephone port is removed.
reversal
Polarity reversal of tip and ring conductors of a telephone port.
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Usage Guidelines
This command applies to Cisco 800 series routers. Most countries except Japan typically use the osi option. Japan typically uses the reversal option.
Examples
The following example specifies that the router uses the OSI disconnect method: pots disconnect-supervision osi
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800 series router to use country-specific default settings for each physical characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed on your connected telephones, fax machines, or modems.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if telephones, fax machines, or modems connected to a Cisco 800 series router fail to detect that a calling party has disconnected.
Cisco IOS Voice Command Reference
VR-1492
Cisco IOS Voice Commands: P pots disconnect-supervision
Command
Description
pots distinctive-ring-guard-time
Specifies the delay in which a telephone port can be rung after a previous call is disconnected (Cisco 800 series routers).
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800 series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and other information on the telephone interfaces on a Cisco 800 series router.
Cisco IOS Voice Command Reference
VR-1493
Cisco IOS Voice Commands: P pots disconnect-time
pots disconnect-time To specify the interval in which the disconnect method is applied if your connected telephones, fax machines, or modems fail to detect that a calling party has disconnected, use the pots disconnect-time command in global configuration mode. To disable the specified disconnect interval, use the no form of this command. pots disconnect-time interval no pots disconnect-time interval
Syntax Description
interval
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Usage Guidelines
Interval, in milliseconds. Range is from 50 to 2000.
This command applies to Cisco 800 series routers. The pots disconnect-supervision command configures the disconnect method.
Examples
The following example specifies that the connected devices apply the configured disconnect method for 100 ms after a calling party disconnects: pots disconnect-time 100
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800 series router to use country-specific default settings for each physical characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed on your connected telephones, fax machines, or modems.
pots disconnect-supervision
Specifies how a Cisco 800 series router notifies the connected telephones, fax machines, or modems when the calling party has disconnected.
pots distinctive-ring-guard-time
Specifies the delay in which a telephone port can be rung after a previous call is disconnected (Cisco 800 series routers).
Cisco IOS Voice Command Reference
VR-1494
Cisco IOS Voice Commands: P pots disconnect-time
Command
Description
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800 series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and other information on the telephone interfaces on a Cisco 800 series router.
Cisco IOS Voice Command Reference
VR-1495
Cisco IOS Voice Commands: P pots distinctive-ring-guard-time
pots distinctive-ring-guard-time To specify the delay in which a telephone port can be rung after a previous call is disconnected, use the pots distinctive-ring-guard-time command in global configuration mode. To disable the specified delay, use the no form of this command. pots distinctive-ring-guard-time milliseconds no pots distinctive-ring-guard-time milliseconds
Syntax Description
milliseconds
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Delay, in milliseconds. Range is from 0 to 1000.
Usage Guidelines
This command applies to Cisco 800 series routers.
Examples
The following example specifies that a telephone port can be rung 100 ms after a previous call is disconnected: pots distinctive-ring-guard-time 100
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800 series router to use country-specific default settings for each physical characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed on your connected telephones, fax machines, or modems.
pots disconnect-supervision
Specifies how a Cisco 800 series router notifies the connected telephones, fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if telephones, fax machines, or modems connected to a Cisco 800 series router fail to detect that a calling party has disconnected.
Cisco IOS Voice Command Reference
VR-1496
Cisco IOS Voice Commands: P pots distinctive-ring-guard-time
Command
Description
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800 series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax machines, or modems connected to a Cisco 800 series router.
ring
Sets up a distinctive ring for telephones, fax machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and other information on the telephone interfaces on a Cisco 800 series router.
Cisco IOS Voice Command Reference
VR-1497
Cisco IOS Voice Commands: P pots encoding
pots encoding To specify the pulse code modulation (PCM) encoding scheme for your connected telephones, fax machines, or modems, use the pots encoding command in global configuration mode. To disable the specified scheme, use the no form of this command. pots encoding {alaw | ulaw} no pots encoding {alaw | ulaw}
Syntax Description
alaw
A-law. International Telecommunication Union Telecommunication Standardization Section (ITU-T) PCM encoding scheme used to represent analog voice samples as digital values.
ulaw
Mu-law. North American PCM encoding scheme used to represent analog voice samples as digital values.
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Usage Guidelines
This command applies to Cisco 800 series routers. Europe typically uses a-law. North America typically uses u-law.
Examples
The following example specifies a-law as the PCM encoding scheme: pots encoding alaw
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800 series router to use country-specific default settings for each physical characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed on your connected telephones, fax machines, or modems.
pots disconnect-supervision
Specifies how a Cisco 800 series router notifies the connected telephones, fax machines, or modems when the calling party has disconnected.
Cisco IOS Voice Command Reference
VR-1498
Cisco IOS Voice Commands: P pots encoding
Command
Description
pots disconnect-time
Specifies the interval in which the disconnect method is applied if telephones, fax machines, or modems connected to a Cisco 800 series router fail to detect that a calling party has disconnected.
pots distinctive-ring-guard-time
Specifies the delay in which a telephone port can be rung after a previous call is disconnected (Cisco 800 series routers).
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800 series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and other information on the telephone interfaces on a Cisco 800 series router.
Cisco IOS Voice Command Reference
VR-1499
Cisco IOS Voice Commands: P pots forwarding-method
pots forwarding-method To configure the type of call-forwarding method to be used for Euro-ISDN (formerly NET3) switches, use the pots forwarding-method command in global configuration mode. To turn forwarding off, use the no form of this command. pots forwarding-method {keypad | functional} no pots forwarding-method {keypad | functional}
Syntax Description
keypad
Gives forwarding control to the Euro-ISDN switch.
functional
Gives forwarding control to the router. If you select this method, use the dual-tone multifrequency (DTMF) keypad commands listed in Table 34 to configure call-forwarding service.
Command Default
Forwarding is off
Command Modes
Global configuration
Command History
Release
Modification
12.2(2)T
This command was introduced.
Usage Guidelines
Use this command to select the type of forwarding method to be used for Euro-ISDN switches. This command does not affect any other switch types. You can select one or more call-forwarding services at a time, but keep the following Euro-ISDN switch characteristics in mind: •
Call forward unconditional (CFU) redirects a call without restriction and takes precedence over other call-forwarding service types.
•
Call forward busy (CFB) redirects a call to another number if the dialed number is busy.
•
Call forward no reply (CFNR) forwards a call to another number if the dialed number does not answer within a specified period of time.
If all three call-forwarding services are enabled, CFU overrides CFB and CFNR. The default is that no call-forwarding service is selected. If you select the functional forwarding method, use the DTMF keypad commands in Table 34 to configure the call-forwarding service. Table 34
DTMF Keypad Commands for Call-Forwarding Service
Task
DTMF Keypad Command1
Activate CFU
**21*number#
Deactivate CFU
#21#
Cisco IOS Voice Command Reference
VR-1500
Cisco IOS Voice Commands: P pots forwarding-method
Table 34
DTMF Keypad Commands for Call-Forwarding Service (continued)
Task
DTMF Keypad Command1
Activate CFNR
**61*number#
Deactivate CFNR
#61#
Activate CFB
**67*number#
Deactivate CFB
#67#
1. Where number is the telephone number to which your calls are forwarded.
When you enable or disable the call-forwarding service, it is enabled or disabled for four basic services: speech, audio at 3.1 kilohertz (kHz), telephony at 3.1 kHz, and telephony at 7 kHz. You should hear a dial tone after you enter the DTMF keypad command when the call-forwarding service is successfully enabled for at least one of the four basic services. If you hear a busy tone, the command is invalid or the switch does not support that service.
Examples
The following example gives forwarding control to the router: pots forwarding-method functional
Related Commands
Command
Description
pots prefix filter
Sets a filter that prevents a dial prefix from being added to a dialed number when the digits in the dialed number match the filter.
pots prefix number
Sets a prefix to be added to a called telephone number for analog or modem calls.
Cisco IOS Voice Command Reference
VR-1501
Cisco IOS Voice Commands: P pots line-type
pots line-type To specify the impedance of your connected telephones, fax machines, or modems, use the pots line-type command in global configuration mode. To disable the specified line type, use the no form of this command. pots line-type {type1 | type2 | type3} no pots line-type {type1 | type2 | type3}
Syntax Description
type1
Runs at 600 ohms.
type2
Runs at 900 ohms.
type3
Runs at 300 or 400 ohms.
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Usage Guidelines
This command applies to Cisco 800 series routers.
Examples
The following example sets the line type to type1: pots line-type type1
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800 series router to use country-specific default settings for each physical characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed on your connected telephones, fax machines, or modems.
pots disconnect-supervision
Specifies how a Cisco 800 series router notifies the connected telephones, fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if telephones, fax machines, or modems connected to a Cisco 800 series router fail to detect that a calling party has disconnected.
Cisco IOS Voice Command Reference
VR-1502
Cisco IOS Voice Commands: P pots line-type
Command
Description
pots distinctive-ring-guard-time
Specifies the delay in which a telephone port can be rung after a previous call is disconnected (Cisco 800 series routers).
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or modems connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800 series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and other information on the telephone interfaces on a Cisco 800 series router.
Cisco IOS Voice Command Reference
VR-1503
Cisco IOS Voice Commands: P pots prefix filter
pots prefix filter To set a filter that prevents a dial prefix from being added to a dialed number when the digits in the dialed number match the filter, use the pots prefix filter command in global configuration mode. To remove the filter, use the no form of this command. pots prefix filter number no pots prefix filter number
Syntax Description
number
Command Default
No default filter is set.
Command Modes
Global configuration
Command History
Release
Modification
12.2(2)T
This command was introduced on the Cisco 803 and Cisco 804.
Usage Guidelines
Prefix filter numbers, up to a maximum of eight characters.
The pots prefix filter command is used to set a filter for prefix dialing. A maximum of ten filters can be set. Once the maximum number of filters have been configured, an additional filter is not accepted nor does it overwrite any of the existing filters. To configure a new filter, remove at least one filter using the no pots prefix filter command. You can set matching criteria for the filter using the * wildcard character. For example, if you configure the filter 1* and a dialed number starts with 1, the called number is not prefixed. Prefix filters can be of variable length. All configured prefix filters are compared to the number dialed, up to the length of the prefix filter. If there is a match, no prefix is added to the dialed number.
Cisco IOS Voice Command Reference
VR-1504
Cisco IOS Voice Commands: P pots prefix filter
Examples
The following example configures five filters that prevent dial prefixes from being added to dialed numbers: pots pots pots pots pots
prefix prefix prefix prefix prefix
filter filter filter filter filter
192 1 9 0800 08456
With these filters configured, a prefix is not added to the following dialed numbers: 192 Directory calls 100 Operator services 999 Emergency services 0800... Toll-free calls 08456...Calls on an Energis network information controller
Related Commands
Command
Description
pots forwarding-method
Configures the type of forwarding method to be used for Euro-ISDN (formerly NET3) switches.
pots prefix number
Sets a prefix to be added to a called telephone number for analog or modem calls.
Cisco IOS Voice Command Reference
VR-1505
Cisco IOS Voice Commands: P pots prefix number
pots prefix number To set a prefix to be added to a called telephone number for analog or modem calls, use the pots prefix number command in global configuration mode. To remove the prefix, use the no form of this command. pots prefix number number no pots prefix number number
Syntax Description
number
Command Default
No prefix is associated with the called number for analog or modem calls
Command Modes
Global configuration
Command History
Release
Modification
12.2(2)T
This command was introduced on the Cisco 803 and Cisco 804.
Usage Guidelines
Prefix, up to a maximum of five digits.
Only one prefix can be configured using this command. If a prefix already exists, the next prefix configured with this command overwrites the old prefix. Prefixes can be of variable length, up to five digits. The no pots prefix number command removes the prefix. As numbers are dialed on the keypad, a comparison is made to the configured prefix filter. When a match is determined, the number is dialed without adding the prefix. In the unlikely event that the prefix filter has more digits than the dialed number, and the dialed number matches the first digits of the prefix filter, the prefix is not added to the dialed number. For example, if the prefix filter is 5554000 and you dial 555 and stop, the router considers the called number to be 555 and does not add a prefix to the number. This event is unlikely to occur because the number of digits in dialed numbers is typically greater than the number of digits in prefix filters.
Examples
The following example sets the prefix to 12345: pots prefix number 12345
This prefix is added to any number dialed for analog or modem calls that do not match the prefix filter.
Related Commands
Command
Description
pots prefix filter
Sets a filter that prevents a dial prefix from being added to a dialed number when the digits in the dialed number match the filter.
Cisco IOS Voice Command Reference
VR-1506
Cisco IOS Voice Commands: P pots ringing-freq
pots ringing-freq To specify the frequency on the Cisco 800 series router at which connected telephones, fax machines, or modems ring, use the pots ringing-freq command in global configuration mode. To disable the specified frequency, use the no form of this command. pots ringing-freq {20Hz | 25Hz | 50Hz} no pots ringing-freq {20Hz | 25Hz | 50Hz}
Syntax Description
20Hz
Connected devices ring at 20 Hz.
25Hz
Connected devices ring at 25 Hz.
50Hz
Connected devices ring at 50 Hz.
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Usage Guidelines
This command applies to Cisco 800 series routers.
Examples
The following example sets the ringing frequency to 50 Hz: pots ringing-freq 50Hz
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800 series router to use country-specific default settings for each physical characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed on your connected telephones, fax machines, or modems.
pots disconnect-supervision
Specifies how a Cisco 800 series router notifies the connected telephones, fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if telephones, fax machines, or modems connected to a Cisco 800 series router fail to detect that a calling party has disconnected.
Cisco IOS Voice Command Reference
VR-1507
Cisco IOS Voice Commands: P pots ringing-freq
Command
Description
pots distinctive-ring-guard-time
Specifies the delay in which a telephone port can be rung after a previous call is disconnected (Cisco 800 series routers).
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected to a Cisco 800 series router.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800 series router).
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and other information on the telephone interfaces on a Cisco 800 series router.
Cisco IOS Voice Command Reference
VR-1508
Cisco IOS Voice Commands: P pots silence-time
pots silence-time To specify the interval of silence after a calling party disconnects, use the pots silence-time command in global configuration mode. To disable the specified silence time, use the no form of this command. pots silence-time interval no pots silence-time interval
Syntax Description
interval
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Number from 0 to 10 (seconds).
Usage Guidelines
This command applies to Cisco 800 series routers.
Examples
The following example sets the interval of silence to 10 seconds: pots silence-time 10
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800 series router to use country-specific default settings for each physical characteristic.
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed on your connected telephones, fax machines, or modems.
pots disconnect-supervision
Specifies how a Cisco 800 series router notifies the connected telephones, fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if telephones, fax machines, or modems connected to a Cisco 800 series router fail to detect that a calling party has disconnected.
pots distinctive-ring-guard-time
Specifies the delay in which a telephone port can be rung after a previous call is disconnected (Cisco 800 series routers).
Cisco IOS Voice Command Reference
VR-1509
Cisco IOS Voice Commands: P pots silence-time
Command
Description
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems connected to a Cisco 800 series router ring.
pots tone-source
Specifies the source of dial, ringback, and busy tones for telephones, fax machines, or modems connected to a Cisco 800 series router.
show pots status
Displays the settings of the telephone port physical characteristics and other information on the telephone interfaces on a Cisco 800 series router.
Cisco IOS Voice Command Reference
VR-1510
Cisco IOS Voice Commands: P pots tone-source
pots tone-source To specify the source of dial, ringback, and busy tones for your connected telephones, fax machines, or modems, use the pots tone-source command in global configuration mode. To disable the specified source, use the no form of this command. pots tone-source {local | remote} no pots tone-source {local | remote}
Syntax Description
local
Router supplies the tones.
remote
Telephone switch supplies the tones.
Command Default
Local (router supplies the tones)
Command Modes
Global configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Usage Guidelines
This command applies to Cisco 800 series routers. This command applies only to ISDN lines connected to a EURO-ISDN (NET3) switch.
Examples
The following example sets the tone source to remote: pots tone-source remote
Related Commands
Command
Description
pots country
Configures telephones, fax machines, or modems connected to a Cisco 800 series router to use country-specific default settings for each physical characteristic
pots dialing-method
Specifies how the Cisco 800 series router collects and sends digits dialed on your connected telephones, fax machines, or modems.
pots disconnect-supervision
Specifies how a Cisco 800 series router notifies the connected telephones, fax machines, or modems when the calling party has disconnected.
pots disconnect-time
Specifies the interval in which the disconnect method is applied if telephones, fax machines, or modems connected to a Cisco 800 series router fail to detect that a calling party has disconnected.
Cisco IOS Voice Command Reference
VR-1511
Cisco IOS Voice Commands: P pots tone-source
Command
Description
pots distinctive-ring-guard-time
Specifies the delay in which a telephone port can be rung after a previous call is disconnected (Cisco 800 series routers).
pots encoding
Specifies the PCM encoding scheme for telephones, fax machines, or modems connected to a Cisco 800 series router.
pots line-type
Specifies the impedance of telephones, fax machines, or modems connected to a Cisco 800 series router.
pots ringing-freq
Specifies the frequency at which telephones, fax machines, or modems connected to a Cisco 800 series router ring.
pots silence-time
Specifies the interval of silence after a calling party disconnects (Cisco 800 series router).
show pots status
Displays the settings of the telephone port physical characteristics and other information on the telephone interfaces on a Cisco 800 series router.
Cisco IOS Voice Command Reference
VR-1512
Cisco IOS Voice Commands: P pre-dial delay
pre-dial delay To configure a delay on an Foreign Exchange Office (FXO) interface between the beginning of the off-hook state and the initiation of dual-tone multifrequency (DTMF) signaling, use the pre-dial delay command in voice-port configuration mode. To reset to the default, use the no form of the command. pre-dial delay seconds no pre-dial delay
Syntax Description
seconds
Command Default
1 second
Command Modes
Voice-port configuration
Command History
Release
Modification
11.(7)T
This command was introduced on the Cisco 3600 series.
12.0(2)T
This command was integrated into Cisco IOS Release 12.0(2)T.
Delay, in seconds, before signaling begins. Range is from 0 to 10. Default is 1.
Usage Guidelines
To disable the command, set the delay to 0. When an FXO interface begins to draw loop current (off-hook state), a delay is required between the initial flow of loop current and the beginning of signaling. Some devices initiate signaling too quickly, resulting in redial attempts. This command allows a signaling delay.
Examples
The following example sets a predial delay value of 3 seconds on the FXO port: voice-port 1/0/0 pre-dial delay 3
Related Commands
Command
Description
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timing delay-duration
Configures delay dial signal duration for a specified voice port.
Cisco IOS Voice Command Reference
VR-1513
Cisco IOS Voice Commands: P preference (dial peer)
preference (dial peer) To indicate the preferred order of a dial peer within a hunt group, use the preference command in dial peer configuration mode. To remove the preference, use the no form of this command. preference value no preference
Syntax Description
value
Command Default
0 (highest preference)
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810.
12.0(3)T
This command was integrated into Cisco IOS Release 12.0(3)T and implemented on the Cisco 2600 series and Cisco 3600 series.
12.0(4)T
This command was modified to support VoFR dial peers on the Cisco 2600 series and Cisco 3600 series.
Usage Guidelines
Integer from 0 to 10, where the lower the number, the higher the preference. Default is 0 (highest preference).
This command applies to POTS, VoIP, VoFR, and VoATM dial peers. Use this command to indicate the preference order for matching dial peers in a rotary group. Setting the preference enables the desired dial peer to be selected when multiple dial peers within a hunt group are matched for a dial string.
Note
If POTS and voice-network peers are mixed in the same hunt group, the POTS dial peers must have priority over the voice-network dial peers. Use this command with the Rotary Calling Pattern feature described in the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2 chapter “Configuring H.323 Gateways.” The hunting algorithm precedence is configurable. For example, if you wish a call processing sequence to go to destination A first, to destination B second, and to destination C third, you would assign preference (0 being the highest priority) to the destinations in the following order: •
Preference 0 to A
•
Preference 1 to B
•
Preference 2 to C
Cisco IOS Voice Command Reference
VR-1514
Cisco IOS Voice Commands: P preference (dial peer)
Examples
The following example sets POTS dial peer 10 to a preference of 1, POTS dial peer 20 to a preference of 2, and VoFR dial peer 30 to a preference of 3: dial-peer voice 10 pots destination pattern 5550150 preference 1 exit dial-peer voice 20 pots destination pattern 5550150 preference 2 exit dial-peer voice 30 vofr destination pattern 5550150 preference 3 exit
The following examples show different dial peer configurations: Dialpeer 1 2 3 4 5
If the destination number is 4085550148, the order of attempts is 1, 2, 3, 5, 4: Dialpeer destpat preference 1 408555 0 2 4085550148 1 3 4085550 0 4 ..............4085550.........0
If the number dialed is 4085550148, the order is 2, 3, 4, 1.
Note
Related Commands
The default behavior is that the longest matching dial peer supersedes the preference value.
Command
Description
called-number (dial peer)
Enables an incoming VoFR call leg to get bridged to the correct POTS call leg when using a static FRF.11 trunk connection.
codec (dial peer)
Specifies the voice coder rate of speech for a Voice over Frame Relay dial peer.
cptone
Specifies a regional analog voice interface-related tone, ring, and cadence setting.
destination-pattern
Specifies the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.
dtmf-relay (Voice over Frame Relay)
Enables the generation of FRF.11 Annex A frames for a dial peer.
session protocol
Establishes a session protocol for calls between the local and remote routers via the packet network.
Cisco IOS Voice Command Reference
VR-1515
Cisco IOS Voice Commands: P preference (dial peer)
Command
Description
session target
Specifies a network-specific address for a specified dial peer or destination gatekeeper.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
Cisco IOS Voice Command Reference
VR-1516
Cisco IOS Voice Commands: P preemption enable
preemption enable To enable preemption capability on a trunk group, use the preemption enable command in trunk group configuration mode. To disable preemption capabilities, use the no form of this command. preemption enable no preemption enable
Syntax Description
This command has no arguments or keywords.
Command Default
Preemption is disabled on the trunk group.
Command Modes
Trunk group configuration
Command History
Release
Modification
12.4(4)XC
This command was introduced.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T.
Examples
The following command example enables preemption capabilities on trunk group test: Router(config)# trunk group test Router(config-trunk-group)# preemption enable
Related Commands
Command
Description
isdn integrate all
Enables integrated mode on an ISDN PRI interface.
max-calls
Sets the maximum number of calls that a trunk group can handle.
preemption guard timer
Defines time for a DDR call and allows time to clear the last call from the channel.
preemption level
Sets the preemption level of the selected outbound dial peer. Voice calls can be preempted by a DDR call with higher preemption level.
preemption tone timer Defines the expiry time for the preemption tone for the outgoing call being preempted by a DDR backup call.
Cisco IOS Voice Command Reference
VR-1517
Cisco IOS Voice Commands: P preemption guard timer
preemption guard timer To define the time for a DDR call and to allow time to clear the last call from the channel, use the preemption guard timer command in trunk group configuration mode. To disable the preemption guard time, use the no form of this command. preemption guard timer value no preemption guard timer
Syntax Description
value
Command Default
No preemption guard timer is configured.
Command Modes
Trunk group configuration
Command History
Release
Modification
12.4(4)XC
This command was introduced.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T.
Examples
Number, in milliseconds for the preemption guard timer. The range is 60 to 500. The default is 60.
The following set of commands configures a 60-millisecond preemption guard timer on the trunk group dial2. Router(config)# trunk group dial2 Router(config-trunk-group)# preemption enable Router(config-trunk-group)# preemption guard timer 60
Related Commands
Command
Description
isdn integrate all
Enables integrated mode on an ISDN PRI interface.
max-calls
Sets the maximum number of calls that a trunk group can handle.
preemption enable
Enables preemption capabilities on a trunk group.
preemption level
Sets the preemption level of the selected outbound dial-peer. Voice calls can be preempted by a DDR call with higher preemption level.
preemption tone timer Sets the expiry time for the preemption tone for the outgoing call being preempted by a DDR backup call.
Cisco IOS Voice Command Reference
VR-1518
Cisco IOS Voice Commands: P preemption level
preemption level To set the precedence for voice calls to be preempted by a dial-on demand routing (DDR) call for the trunk group, use the preemption level command in dial peer configuration mode. To restore the default preemption level setting, use the no form of this command preemption level {flash-override | flash | immediate | priority | routine} no preemption level
Syntax Description
flash-override
Sets the precedence for voice calls to preemption level 0 (highest).
flash
Sets the precedence for voice calls to preemption level 1.
immediate
Sets the precedence for voice calls to preemption level 2.
priority
Sets the precedence for voice calls to preemption level 3.
routine
Sets the precedence for voice calls to preemption level 4 (lowest). This is the default.
Command Default
The preemption level default is routine (lowest).
Command Modes
Dial peer configuration
Command History
Release
Modification
12.4(4)XC
This command was introduced.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T.
Examples
The following command example sets a preemption level of flash (level 1) on POTS dial-peer 20: Router(config)# dial-peer voice 20 pots Router(config-dial-peer)# preemption level flash
Related Commands
Command
Description
dialer preemption level
Sets the precedence for voice calls to be preempted by a DDR call for the dialer map.
isdn integrate all
Enables integrated mode on an ISDN PRI interface.
max-calls
Sets the maximum number of calls that a trunk group can handle.
preemption enable
Enables preemption capabilities on a trunk group.
preemption guard timer
Defines time for a DDR call and allows time to clear the last call from the channel.
preemption tone timer Defines the expiry time for the preemption tone for the outgoing call being preempted by a DDR backup call.
Cisco IOS Voice Command Reference
VR-1519
Cisco IOS Voice Commands: P preemption tone timer
preemption tone timer To set the expiry time for the preemption tone for the outgoing call being preempted by a DDR backup call, use the preemption tone timer command in trunk group configuration mode. To clear the expiry time, use the no form of this command. preemption tone timer seconds no preemption tone timer
Syntax Description
seconds
Command Default
No preemption tone timer is configured.
Command Modes
Trunk group configuration
Command History
Release
Modification
12.4(4)XC
This command was introduced.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T.
Examples
Length of preemption tone, in seconds. Range: 4 to 30. Default: 10.
The following set of commands configures a 20-second preemption tone timer on trunk group dial2. Router(config)# trunk group dial2 Router(config-trunk-group)# preemption enable Router(config-trunk-group)# preemption tone timer 20
Related Commands
Command
Description
isdn integrate all
Enables integrated mode on an ISDN PRI interface.
max-calls
Sets the maximum number of calls that a trunk group can handle.
preemption enable
Enables preemption capabilities on a trunk group.
preemption level
Sets the preemption level of the selected outbound dial peer. Voice calls can be preempted by a DDR call with higher preemption level.
Cisco IOS Voice Command Reference
VR-1520
Cisco IOS Voice Commands: P prefix
prefix To specify the prefix of the dialed digits for a dial peer, use the prefix command in dial peer configuration mode. To disable this feature, use the no form of this command. prefix string no prefix
Syntax Description
string
Command Default
Null string
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series.
12.0(4)XJ
This command was implemented on the Cisco AS5300. It and modified for store-and-forward fax.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
12.2(4)T
This command was implemented on the Cisco 1750.
12.2(8)T
This command was implemented on the following platforms: Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.2(13)T
This command was supported in Cisco IOS Release 12.2(13)T and implemented on the Cisco 2600XM, Cisco ICS7750, and Cisco VG200.
Usage Guidelines
Integers that represent the prefix of the telephone number associated with the specified dial peer. Valid values are 0 through 9 and a comma (,). Use a comma to include a pause in the prefix.
Use this command to specify a prefix for a specific dial peer. When an outgoing call is initiated to this dial peer, the prefix string value is sent to the telephony interface first, before the telephone number associated with the dial peer. If you want to configure different prefixes for dialed numbers on the same interface, you need to configure different dial peers. This command is applicable only to plain old telephone service (POTS) dial peers. This command applies to off-ramp store-and-forward fax functions.
Cisco IOS Voice Command Reference
VR-1521
Cisco IOS Voice Commands: P prefix
Examples
The following example specifies a prefix of 9 and then a pause: dial-peer voice 10 pots prefix 9,
The following example specifies a prefix of 5120002: Router(config-dial-peer)# prefix 5120002
Related Commands
Command
Description
answer-address
Specifies the full E.164 telephone number to be used to identify the dial peer of an incoming call.
destination-pattern
Specifies either the prefix or the full E.164 telephone number to be used for a dial peer.
Cisco IOS Voice Command Reference
VR-1522
Cisco IOS Voice Commands: P prefix (Annex G)
prefix (Annex G) To restrict the prefixes for which the gatekeeper should query the Annex G border element (BE), use the prefix command in gatekeeper border element configuration mode. prefix prefix* [seq | blast]
Syntax Description
prefix*
Prefix for which BEs should be queried.
seq
(Optional) Queries are sent out to the neighboring BEs sequentially.
blast
(Optional) Queries are sent out to the neighboring BEs simultaneously.
Command Default
Any time a remote zone query occurs, the BE is also queried.
Command Modes
Gatekeeper border element configuration
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
Usage Guidelines
By default, the gatekeeper sends all remote zone requests to the BE. Use this command only if you want to restrict the queries to the BE to a specific prefix or set of prefixes.
Examples
The following example directs the gatekeeper to query the BE using a prefix of 408. Router(config-gk-annexg)# prefix 408* seq
Related Commands
Command
Description
h323-annexg
Enables the BE on the gatekeeper and enters border element configuration mode.
Cisco IOS Voice Command Reference
VR-1523
Cisco IOS Voice Commands: P prefix (stcapp-fac)
prefix (stcapp-fac) To designate a prefix string to precede the dialing of SCCP telephony control (STC) feature access codes, use the prefix command in STC application feature access-code configuration mode. To return the prefix to its default, use the no form of this command. prefix prefix-string no prefix
Syntax Description
prefix-string
Command Default
The default prefix is ** (two asterisks).
Command Modes
STC application feature access-code configuration
Command History
Release
Modification
12.4(2)T
This command was introduced.
Usage Guidelines
String of one to ten characters that can be dialed on a telephone keypad. String must start with * (asterisk) or # (pound sign). Default is **.
This command is used with the STC application, which enables certain features on analog FXS endpoints that use Skinny Client Control Protocol (SCCP) for call control. Phone users dial the feature access code (FAC) prefix string before dialing a FAC that activates a feature. For example, to set call forwarding for all calls using the default prefix and FAC, a phone user dials **1. Use this command only if you want to change the prefix from its default (**). The show running-config command displays nondefault FACs and prefixes only. The show stcapp feature codes command displays all FACs and prefixes.
Examples
The following example sets a FAC prefix of two pound signs (##). After this value is configured, a phone user dials ##2 on the keypad to forward all calls for that extension. Router(config)# stcapp feature access-code Router(stcapp-fac)# prefix ## Router(stcapp-fac)# call forward all 2 Router(stcapp-fac)# call forward cancel 3 Router(stcapp-fac)# pickup local 6 Router(stcapp-fac)# pickup group 5 Router(stcapp-fac)# pickup direct 4 Router(stcapp-fac)# exit
Cisco IOS Voice Command Reference
VR-1524
Cisco IOS Voice Commands: P prefix (stcapp-fac)
Related Commands
Command
Description
call forward all
Designates an STC application feature access code to activate the forwarding of all calls.
call forward cancel
Designates an STC application feature access code to cancel the forwarding of all calls.
pickup direct
Designates an STC application feature access code for directed call pickup.
pickup group
Designates an STC application feature access code for group call pickup from another group.
pickup local
Designates an STC application feature access code for group call pickup from the local group.
show running-config
Displays current nondefault configuration settings.
show stcapp feature codes
Displays configured and default STC application feature access codes.
stcapp feature access-code
Enters STC application feature access-code configuration mode to set feature access codes.
Cisco IOS Voice Command Reference
VR-1525
Cisco IOS Voice Commands: P prefix (stcapp-fsd)
prefix (stcapp-fsd) To designate a prefix string to precede the dialing of SCCP telephony control (STC) application feature speed-dial codes, use the prefix command in STC application feature speed-dial configuration mode. To return the prefix to its default, use the no form of this command. prefix prefix-string no prefix
Syntax Description
prefix-string
Command Default
The default prefix is * (one asterisk).
Command Modes
STC application feature speed-dial configuration
Command History
Release
Modification
12.4(2)T
This command was introduced.
Usage Guidelines
String of one to ten characters that can be dialed on a telephone keypad. String must start with * (asterisk) or # (pound sign). Default is *.
This command is used with the STC application, which enables certain features on analog FXS endpoints that use Skinny Client Control Protocol (SCCP) for call control. Phone users dial the feature speed-dial (FSD) prefix string before dialing an FSD code that dials a telephone number. For example, to dial the telephone number that is stored in speed-dial position 2, a phone user dials *2. Use this command only if you want to change the prefix from its default (*). The show running-config command displays nondefault FSDs and prefixes only. The show stcapp feature codes command displays all feature speed-dial FSDs and prefixes.
Examples
The following example sets an FSD prefix of three asterisks (***). After this value is configured, a phone user presses ***2 on the keypad to dial speed-dial number 2. Router(config)# stcapp feature speed-dial Router(stcapp-fsd)# prefix *** Router(stcapp-fsd)# speed dial from 2 to 7 Router(stcapp-fsd)# redial 9 Router(stcapp-fsd)# voicemail 8 Router(stcapp-fsd)# exit
Cisco IOS Voice Command Reference
VR-1526
Cisco IOS Voice Commands: P prefix (stcapp-fsd)
Related Commands
Command
Description
redial
Designates an STC application feature speed-dial code to dial again the last number that was dialed.
show stcapp feature codes
Displays configured and default STC application feature access codes.
speed dial
Designates a range of STC application feature speed-dial codes.
stcapp feature speed-dial
Enters STC application feature speed-dial configuration mode to set feature speed-dial codes.
voicemail (stcapp-fsd)
Designates an STC application feature speed-dial code to dial the voice-mail number.
Cisco IOS Voice Command Reference
VR-1527
Cisco IOS Voice Commands: P preloaded-route
preloaded-route To enable preloaded route support for VoIP Session Initiation Protocol (SIP) calls, use the preloaded-route command in SIP configuration mode. To reset to the default, use the no form of this command. preloaded-route [sip-server] service-route no preloaded-route
Syntax Description
sip-server
(Optional) Adds SIP server information to the Route header.
service-route
Adds the Service-Route information to the Route header.
Command Default
Route support is not enabled.
Command Modes
SIP configuration (conf-serv-sip)
Command History
Release
Modification
12.4(22)YB
This command was introduced.
15.0(1)M
This command was integrated into Cisco IOS Release 15.0(1)M.
Usage Guidelines
The voice-class preloaded-route command, in dial-peer configuration mode, takes precedence over the preloaded-route command in SIP configuration mode. However, if the voice-class preloaded-route command is configured with the system keyword, the gateway uses the global settings configured by the preloaded-route command. Enter SIP configuration mode after entering voice-service VoIP configuration mode, as shown in the “Examples” section.
Examples
The following example shows how to configure the system to include SIP server and Service-Route information in the Route header: voice service voip sip preloaded-route sip-server service-route
The following example shows how to configure the system to include only Service-Route information in the Route header: voice service voip sip preloaded-route service-route
Cisco IOS Voice Command Reference
VR-1528
Cisco IOS Voice Commands: P preloaded-route
Related Commands
Command
Description
sip
Enters SIP configuration mode from voice-service VoIP configuration mode.
voice-class preloaded-route
Enables preloaded route support for dial-peer SIP calls.
Cisco IOS Voice Command Reference
VR-1529
Cisco IOS Voice Commands: P presence
presence To enable presence service and enter presence configuration mode, use the presence command in global configuration mode. To disable presence service, use the no form of this command. presence no presence
Syntax Description
This command has no arguments or keywords.
Command Default
Presence service is disabled.
Command Modes
Global configuration (config)
Command History
Release
Cisco Product
Modification
12.4(11)XJ
Cisco Unified CME 4.1
This command was introduced.
12.4(15)T
Cisco Unified CME 4.1
This command was integrated into Cisco IOS Release 12.4(15)T.
Usage Guidelines
Examples
This command enables the router to perform the following presence functions: •
Process presence requests from internal lines to internal lines. Notify internal subscribers of any status change.
•
Process incoming presence requests from a SIP trunk for internal lines. Notify external subscribers of any status change.
•
Send presence requests to external presentities on behalf of internal lines. Relay status responses to internal lines.
The following example shows how to enable presence and enter presence configuration mode to set the maximum subscriptions to 150: Router(config)# presence Router(config-presence)# max-subscription 150
Related Commands
Command
Description
allow watch
Allows a directory number on a phone registered to Cisco Unified CME to be watched in a presence service.
debug presence
Displays debugging information about the presence service.
max-subscription
Sets the maximum number of concurrent watch sessions that are allowed.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P presence
Command
Description
presence enable
Allows the router to accept incoming presence requests.
server
Specifies the IP address of a presence server for sending presence requests from internal watchers to external presence entities.
show presence global
Displays configuration information about the presence service.
show presence subscription
Displays information about active presence subscriptions.
Cisco IOS Voice Command Reference
VR-1531
Cisco IOS Voice Commands: P presence call-list
presence call-list To enable Busy Lamp Field (BLF) monitoring for call lists and directories on phones registered to the Cisco Unified CME router, use the presence call-list command in ephone, presence, or voice register pool configuration mode. To disable BLF indicators for call lists, use the no form of this command. presence call-list no presence call-list
This command was integrated into Cisco IOS Release 12.4(15)T.
Usage Guidelines
This command enables a phone to monitor the line status of directory numbers listed in a directory or call list, such as a missed calls, placed calls, or received calls list. Using this command in presence mode enables the BLF call-list feature for all phones. To enable the feature for an individual SCCP phone, use this command in ephone configuration mode. To enable the feature for an individual SIP phone, use this command in voice register pool configuration mode. If this command is disabled globally and enabled in voice register pool or ephone configuration mode, the feature is enabled for that voice register pool or ephone. If this command is enabled globally, the feature is enabled for all voice register pools and ephones regardless of whether it is enabled or disabled on a specific voice register pool or ephone. To display a BLF status indicator, the directory number associated with a telephone number or extension must have presence enabled with the allow watch command. For information on the BLF status indicators that display on specific types of phones, see the Cisco Unified IP Phone documentation for your phone model.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P presence call-list
Examples
The following example shows the BLF call-list feature enabled for ephone 1. The line status of a directory number that appears in a call list or directory is displayed on phone 1 if the directory number has presence enabled. Router(config)# ephone 1 Router(config-ephone)# presence call-list
Related Commands
Command
Description
allow watch
Allows a directory number on a phone registered to Cisco Unified CME to be watched in a presence service.
blf-speed-dial
Enables BLF monitoring for a speed-dial number on a phone registered to Cisco Unified CME.
presence
Enables presence service and enters presence configuration mode.
show presence global
Displays configuration information about the presence service.
Cisco IOS Voice Command Reference
VR-1533
Cisco IOS Voice Commands: P presence enable
presence enable To allow incoming presence requests, use the presence enable command in SIP user-agent configuration mode. To block incoming requests, use the no form of this command. presence enable no presence enable
Syntax Description
This command has no arguments or keywords.
Command Default
Incoming presence requests are blocked.
Command Modes
SIP UA configuration (config-sip-ua)
Command History
Release
Modification
12.4(11)XJ
This command was introduced.
12.4(15)T
This command was integrated into Cisco IOS Release 12.4(15)T.
Usage Guidelines
This command allows the router to accept incoming presence requests (SUBSCRIBE messages) from internal watchers and SIP trunks. It does not impact outgoing presence requests.
Examples
The following example shows how to allow incoming presence requests: Router(config)# sip-ua Router(config-sip-ua)# presence enable
Related Commands
Command
Description
allow subscribe
Allows internal watchers to monitor external presence entities (directory numbers).
allow watch
Allows a directory number on a phone registered to Cisco Unified CME to be watched in a presence service.
max-subscription
Sets the maximum number of concurrent watch sessions that are allowed.
show presence global
Displays configuration information about the presence service.
show presence subscription
Displays information about active presence subscriptions.
watcher all
Allows external watchers to monitor internal presence entities (directory numbers).
Cisco IOS Voice Command Reference
VR-1534
Cisco IOS Voice Commands: P pri-group (pri-slt)
pri-group (pri-slt) To specify an ISDN PRI on a channelized T1 or E1 controller, use the pri-group (pri-slt) command in controller configuration mode. To remove the ISDN PRI configuration, use the no form of this command. pri-group [timeslots timeslot-range [nfas_d [backup | none | primary [nfas_int number]] [nfas-group number [iua as-name]]] no pri-group
Syntax Description
timeslots timeslot-range
Specifies a single range of timeslot values in the PRI goup. For T1, the allowable range is from 1 to 23. For E1, the allowable range is from 1 to 31.
nfas_d
Specifies the operation of the D channel timeslot.
backup
(Optional) Specifes that the operation of the D channel timeslot on this controller is the NFAS D backup.
none
(Optional) Specifes that the D channel timeslot is used as an additional B channel.
primary
Specifies that the D channel timeslot on this controller in NFAS D.
nfas_int range
Specifies the provisioned NFAS interface value. Valid values range from 0 to 32.
nfas-group number
Specifies the NFAS group and the NFAS group number. Valid values range from 0 to 31.
iua as-name
Binds the Non-Facility Associated Signaling (NFAS) group to the IDSN User Adaptation Layer (IUA) application server (AS).
Command Default
No ISDN-PRI group is configured.
Command Modes
Controller configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
12.2(15)T
This command was integrated on the Cisco 2420, Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series; and Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 network access server (NAS) platforms.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P pri-group (pri-slt)
Usage Guidelines
The pri-group (pri-slt) command provides another way to bind a D channel to a specific IUA AS. This option allows the RLM group to be configured at the pri-group level instead of in the D channel configuration. For example, a typical configuration would look like the following: controller t1 1/0/0 pri-group timeslots 1-24 nfas_d pri nfas_int 0 nfas_group 1 iua asname
Before you enter the pri-group command, you must specify an ISDN-PRI switch type and an E1 or T1 controller. When configuring NFAS, you use an extended version of the pri-group command to specify the following values for the associated channelized T1 controllers configured for ISDN: •
The range of PRI timeslots to be under the control of the D channel (timeslot 24).
•
The function to be performed by timeslot 24 (primary D channel, backup, or none); the latter specifies its use as a B channel.
•
The group identifier number for the interface under the control of a particular D channel.
The iua keyword is used to bind an NFAS group to the IUA AS. When binding the D channel to an IUA AS, the as-name must match the name of an AS set up during IUA configuration. Before you can modify a PRI group on a Media Gateway Controller (MGC), you must first shut down the D channel. The following shows how to shut down the D channel: Router# configure terminal Enter configuration commands, one per line.
The following example configures the NFAS primary D channel on one channelized T1 controller, and binds the D channel to an IUA AS. This example uses the Cisco AS5400 and applies to T1, which has 24 timeslots and is used mainly in North America and Japan: Router(config-controller)# pri-group timeslots 1-23 nfas-d primary nfas-int 0 nfas-group 1 iua as5400-4-1
The following example applies to E1, which has 32 timeslots and is used by the rest of the world: Router(config-controller)# pri-group timeslots 1-31 nfas-d primary nfas-int 0 nfas-group 1 iua as5400-4-1
The following example configures ISDN-PRI on all time slots of controller E1: Router(config)# controller E1 4/1 Router(config-controller)# pri-group timeslots 1-7,16
In the following example, the rlm-timeslot keyword automatically creates interface serial 4/7:11 (4/7:0:11 if you are using the CT3 card) for the D channel object on a Cisco AS5350. You can choose any timeslot other than 24 to be the virtual container for the D channel parameters for ISDN. Router(config-controller)# pri-group timeslots 1-23 nfas-d primary nfas-int 0 nfas-group 0 rlm-timeslot 3
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: P pri-group (pri-slt)
Related Commands
Command
Description
isdn switch-type
Configures the Cisco 2600 series router PRI interface to support QSIG signaling.
Cisco IOS Voice Command Reference
VR-1537
Cisco IOS Voice Commands: P pri-group nec-fusion
pri-group nec-fusion To configure your NEC PBX to support Fusion Call Control Signaling (FCCS), use the pri-group nec-fusion command in controller configuration mode. To disable FCCS, use the no form of this command. pri-group nec-fusion {pbx-ip-address | pbx-ip-host-name} pbx-port number no pri-group nec-fusion {pbx-ip-address | pbx-ip-host-name} pbx-port number
Syntax Description
pbx-ip-address
IP address of the NEC PBX.
pbx-ip-host-name
Host name of the NEC PBX.
pbx-port number
Port number for the PBX. Range is from 49152 to 65535. Default is 55000. If this value is already in use, the next greater value is used.
Command Default
PBX port number: 55000
Command Modes
Controller configuration
Command History
Release
Modification
12.0(7)T
This command was introduced on the Cisco AS5300.
12.2(1)
This command was modified to add support for setup messages from a POTS dial peer.
Usage Guidelines
This command is used only if the PBX in your configuration is an NEC PBX, and if you are configuring it to run FCCS and not QSIG signaling.
Examples
The following example directs this NEC PBX to use FCCS: pri-group nec-fusion 172.31.255.255 pbx-port 60000
Related Commands
Command
Description
isdn protocol-emulate
Configures the Layer 2 and Layer 3 port protocol of a BRI voice port or a PRI interface to emulate NT (network) or TE (user) functionality.
isdn switch type
Configures the Cisco AS5300 universal access server PRI interface to support QSIG signaling.
show cdapi
Displays the CDAPI.
show rawmsg
Displays the raw messages owned by the required component.
Cisco IOS Voice Command Reference
VR-1538
Cisco IOS Voice Commands: P pri-group timeslots
pri-group timeslots To specify an ISDN PRI group on a channelized T1 or E1 controller, and to release the ISDN PRI signaling time slot, use the pri-group timeslots command in controller configuration mode. To remove or change the ISDN PRI configuration, use the no form of this command. pri-group timeslots timeslot-range [nfas_d {backup nfas_int number nfas_group number [service mgcp] | none nfas_int number nfas_group number [service mgcp] | primary nfas_int number nfas_group number [iua as-name | rlm-group number | service mgcp]} | service mgcp] no pri-group timeslots timeslot-range [nfas_d {backup nfas_int number nfas_group number [service mgcp] | none nfas_int number nfas_group number [service mgcp] | primary nfas_int number nfas_group number [iua as-name | rlm-group number | service mgcp]} | service mgcp]
Syntax Description
timeslot-range
A value or range of values for time slots on a T1 or E1 controller that consists of an ISDN PRI group. Use a hyphen to indicate a range. Note
Groups of time slot ranges separated by commas (1-4,8-23 for example) are also accepted.
nfas_d
(Optional) Configures the operation of the ISDN PRI D channel.
backup
The D-channel time slot is used as the Non-Facility Associated Signaling (NFAS) D backup.
none
The D-channel time slot is used as an additional B channel.
primary
The D-channel time slot is used as the NFAS D primary.
nfas_int number
Specifies the provisioned NFAS interface as a value. Valid values for the NFAS interface range from 0 to 44.
nfas_group number
Specifies the NFAS group. Valid values for the NFAS group number range from 0 to 31.
iua as-name
(Optional) Configures the ISDN User Adaptation Layer (IUA) application server (AS) name.
rlm-group number
(Optional) Specifies the Redundant Link Manager (RLM) group and releases the ISDN PRI signaling channel. Valid values for the RLM group number range from 0 to 255.
service mgcp
(Optional) Configures the service type as Media Gateway Control Protocol (MGCP) service.
Defaults
No ISDN PRI group is configured. The switch type is automatically set to the National ISDN switch type (primary-ni keyword) when the pri-group timeslots command is configured with the rlm-group subkeyword.
Command Modes
Controller configuration
Cisco IOS Voice Command Reference
VR-1539
Cisco IOS Voice Commands: P pri-group timeslots
Command History
Usage Guidelines
Release
Modification
11.0
This command was introduced.
11.3
This command was enhanced to support NFAS.
12.0(2)T
This command was implemented on the Cisco MC3810 multiservice concentrator.
12.0(7)XK
This command was implemented on the Cisco 2600 and Cisco 3600 series routers.
12.1(2)T
The modifications in Cisco IOS Release 12.0(7)XK were integrated into Cisco IOS Release 12.1(2)T.
12.2(8)B
This command was modified with the rlm-group subkeyword to support release of the ISDN PRI signaling channels.
12.2(15)T
The modifications in Cisco IOS Release 12.2(8)B were integrated into Cisco IOS Release 12.2(15)T.
12.4(16)b
This command was modified to ensure that the NFAS primary interface is configured before the NFAS backup or NFAS none interfaces are configured.
12.4(24)T
Support was extended to provide backup functionality for the NFAS interface in MGCP backhaul mode. With this support, if the primary fails, backup can become active and calls can be maintained.
The pri-group command supports the use of DS0 time slots for Signaling System 7 (SS7) links, and therefore the coexistence of SS7 links and PRI voice and data bearer channels on the same T1 or E1 span. In these configurations, the command applies to voice applications. In SS7-enabled Voice over IP (VoIP) configurations when an RLM group is configured, High-Level Data Link Control (HDLC) resources allocated for ISDN signaling on a digital subscriber line (DSL) interface are released and the signaling slot is converted to a bearer channel (B24). The D channel will be running on IP. The chosen D-channel time slot can still be used as a B channel by using the isdn rlm-group interface configuration command to configure the NFAS groups. NFAS allows a single D channel to control multiple PRI interfaces. Use of a single D channel to control multiple PRI interfaces frees one B channel on each interface to carry other traffic. A backup D channel can also be configured for use when the primary NFAS D channel fails. When a backup D channel is configured, any hard system failure causes a switchover to the backup D channel and currently connected calls remain connected. NFAS is supported only with a channelized T1 controller and, as a result, must be ISDN PRI capable. When the channelized T1 controllers are configured for ISDN PRI, only the NFAS primary D channel must be configured; its configuration is distributed to all members of the associated NFAS group. Any configuration changes made to the primary D channel will be propagated to all NFAS group members. The primary D channel interface is the only interface shown after the configuration is written to memory. The channelized T1 controllers on the router must also be configured for ISDN. The router must connect to either an AT&T 4ESS, Northern Telecom DMS-100 or DMS-250, or National ISDN switch type. The ISDN switch must be provisioned for NFAS. The primary and backup D channels should be configured on separate T1 controllers. The primary, backup, and B-channel members on the respective controllers should be the same configuration as that configured on the router and ISDN switch. The interface ID assigned to the controllers must match that of the ISDN switch.
Cisco IOS Voice Command Reference
VR-1540
Cisco IOS Voice Commands: P pri-group timeslots
You can disable a specified channel or an entire PRI interface, thereby taking it out of service or placing it into one of the other states that is passed in to the switch using the isdn service interface configuration command. In the event that a controller belonging to an NFAS group is shut down, all active calls on the controller that is shut down will be cleared (regardless of whether the controller is set to primary, backup, or none), and one of the following events will occur: •
If the controller that is shut down is configured as the primary and no backup is configured, all active calls on the group are cleared.
•
If the controller that is shut down is configured as the primary, and the active (In service) D channel is the primary and a backup is configured, then the active D channel changes to the backup controller.
•
If the controller that is shut down is configured as the primary, and the active D channel is the backup, then the active D channel remains as backup controller.
•
If the controller that is shut down is configured as the backup, and the active D channel is the backup, then the active D channel changes to the primary controller.
The expected behavior in NFAS when an ISDN D channel (serial interface) is shut down is that ISDN Layer 2 should go down but keep ISDN Layer 1 up, and that the entire interface will go down after the amount of seconds specified for timer T309.
Note
The active D channel changeover between primary and backup controllers happens only when one of the link fails and not when the link comes up. The T309 timer is triggered when the changeover takes place.
Note
You must first configure the NFAS primary D channel before configuring the NFAS backup or NFAS none interfaces. If this order is not followed, this message is displayed: "NFAS backup and none interfaces are not allowed to be configured without primary. First configure primary D channel." To remove the NFAS primary D channel after the NFAS backup or NFAS none interfaces are configured, you must remove the NFAS backup or NFAS none interfaces first, and then remove the NFAS primary D channel.
Examples
The following example configures T1 controller 1/0 for PRI and for the NFAS primary D channel. This primary D channel controls all the B channels in NFAS group 1. controller t1 1/0 framing esf linecode b8zs pri-group timeslots 1-24 nfas_d primary nfas_int 0 nfas_group 1
The following example specifies ISDN PRI on T1 slot 1, port 0, and configures voice and data bearer capability on time slots 2 through 6: isdn switch-type primary-4ess controller t1 1/0 framing esf linecode b8zs pri-group timeslots 2-6
Cisco IOS Voice Command Reference
VR-1541
Cisco IOS Voice Commands: P pri-group timeslots
The following example configures a standard ISDN PRI interface: ! Standard PRI configuration: controller t1 1 pri-group timeslots 1-23 nfas_d primary nfas_int 0 nfas_group 0 exit ! Standard ISDN serial configuration: interface serial1:23 ! Set ISDN parameters: isdn T309 4000 exit
The following example configures a dedicated T1 link for SS7-enabled VoIP: controller T1 1 pri-group timeslots 1-23 nfas_d primary nfas_int 0 nfas_group 0 exit ! In a dedicated configuration, we assume the 24th timeslot will be used by ISDN. ! Serial interface 0:23 is created for configuring ISDN parameters. interface Serial:24 ! The D channel is on the RLM. isdn rlm 0 isdn T309 4000 exit
The following example configures a shared T1 link for SS7-enabled VoIP. The rlm-group 0 portion of the pri-group timeslots command releases the ISDN PRI signaling channel. controller T1 1 pri-group timeslots 1-3 nfas_d primary nfas_int 0 nfas_group 0 rlm-group 0 channel group 23 timeslot 24 end ! D-channel interface is created for configuration of ISDN parameters: interface Dchannel1 isdn T309 4000 end
Related Commands
Command
Description
controller
Configures a T1 or E1 controller and enters controller configuration mode.
interface Dchannel Specifies an ISDN D-channel interface for VoIP applications that require release of the ISDN PRI signaling time slot for RLM configurations. interface serial
Specifies a serial interface created on a channelized E1 or channelized T1 controller for ISDN PRI signaling.
isdn rlm-group
Specifies the RLM group number that ISDN will start using.
isdn switch-type
Specifies the central office switch type on the ISDN PRI interface.
isdn timer t309
Changes the value of the T309 timer to clear network connections and release the B channels when there is no signaling channel active, that is, when the D channel has failed and cannot recover by switching to an alternate D channel. Calls remain active and able to transfer data when the D channel fails until the T309 timer expires. The T309 timer is canceled when D-channel failover succeeds.
show isdn nfas group
Displays all the members of a specified NFAS group or all NFAS groups.
Cisco IOS Voice Command Reference
VR-1542
Cisco IOS Voice Commands: P primary (gateway accounting file)
primary (gateway accounting file) To set the primary location for storing the call detail records (CDRs) generated for file accounting, use the primary command in gateway accounting file configuration mode. To reset to the default, use the no form of this command. primary {ftp path/filename username username password password | ifs device:filename} no primary {ftp | ifs}
Syntax Description
ftp path/filename
Name and location of the file on an external FTP server. Filename is limited to 25 characters.
ifs device:filename
Name and location of the file in flash memory or other internal file system on this router. Values depend on storage devices available on the router, for example flash or slot0. Filename is limited to 25 characters.
This command was integrated into Cisco IOS Release 12.4(20)T.
Usage Guidelines
This command specifies the name and location of the primary file where CDRs are stored during the file accounting process. The filename you assign is appended with the gateway hostname and time stamp at the time the file is created to make the filename unique. For example, if you specify the filename cdrtest1 on a router with the hostname cme-2821, a file is created with the name cdrtest1.cme-2821.2007_10_28T22_21_41.000, where 2007_10_28T22_21_41.000 is the time that the file was created. Limit the filename you assign with this command to 25 characters, otherwise it could be truncated when the accounting file is created because the full filename, including the appended hostname and timestamp, is limited to 63 characters. If the file transfer to this primary device fails, the file accounting process retries the primary device up to the number of times defined by the maximum retry-count command and then switches over to the secondary device defined with the secondary command. To manually switch back to the primary device when it becomes available, use the file-acct reset command. The system does not automatically switch back to the primary device. A syslog warning message is generated when flash becomes full.
Cisco IOS Voice Command Reference
VR-1543
Cisco IOS Voice Commands: P primary (gateway accounting file)
Examples
The following example shows the primary location of the accounting file is set to an external FTP server and the filename is cdrtest1: gw-accounting file primary ftp server1/cdrtest1 username bob password temp secondary flash ifs:cdrtest2 maximum buffer-size 25 maximum retry-count 3 maximum fileclose-timer 720 cdr-format compact
The following examples show how the accounting file is named when it is created. The router hostname and time stamp are appended to the filename that you assign with this command: cme-2821(config)# primary ftp server1/cdrtest1 username bob password temp
The name of the accounting file that is created has the following format: cdrtest1.cme-2821.06_04_2007_18_44_51.785
Related Commands
Command
Description
file-acct flush
Manually flushes the CDRs from the buffer to the accounting file.
file-acct reset
Manually switches back to the primary device for file accounting.
maximum retry-count Sets the maximum number of times the router attempts to connect to the primary file device before switching to the secondary device. secondary
Cisco IOS Voice Command Reference
VR-1544
Sets the backup location for storing CDRs if the primary location becomes unavailable.
Cisco IOS Voice Commands: P privacy
privacy To set privacy support at the global level as defined in RFC 3323, use the privacy command in voice service voip sip configuration mode. To remove privacy support as defined in RFC 3323, use the no form of this command. privacy {pstn | privacy-option [critical]} no privacy
Syntax Description
pstn
Requests that the privacy service implements a privacy header using the default Public Switched Telephone Network (PSTN) rules for privacy (based on information in Octet 3a). When selected, this becomes the only valid option.
privacy-option
The privacy support options to be set at the global level. The following keywords can be specified for the privacy-option argument: •
header — Requests that privacy be enforced for all headers in the Session Initiation Protocol (SIP) message that might identify information about the subscriber.
•
history — Requests that the information held in the history-info header is hidden outside the trust domain.
•
id — Requests that the Network Asserted Identity that authenticated the user be kept private with respect to SIP entities outside the trusted domain.
•
session — Requests that the information held in the session description is hidden outside the trust domain.
•
user — Requests that privacy services provide a user-level privacy function.
Note
critical
The keywords can be used alone, altogether, or in any combination with each other, but each keyword can be used only once.
(Optional) Requests that the privacy service performs the specified service or fail the request. Note
This optional keyword is only available after at least one of the privacy-option keywords (header, history, id, session, or user) has been specified and can be used only once per command.
Command Default
Privacy support is disabled.
Command Modes
Voice service voip sip configuration (conf-serv-sip)
Cisco IOS Voice Command Reference
VR-1545
Cisco IOS Voice Commands: P privacy
Command History
Usage Guidelines
Release
Modification
12.4(15)T
This command was introduced.
12.4(22)T
The history keyword was added to provide support for the history-info header information.
Use the privacy command to instruct the gateway to add a Proxy-Require header set to a value supported by RFC 3323 in outgoing SIP request messages. Use the privacy critical command to instruct the gateway to add a Proxy-Require header with the value set to critical. If a user agent sends a request to an intermediary that does not support privacy extensions, the request fails.
Examples
The following example shows how to set the privacy to PSTN: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# privacy pstn
Related Commands
Command
Description
asserted-id
Sets the privacy level and enables either PAI or PPI privacy headers in outgoing SIP requests or response messages.
calling-info pstn-to-sip Specifies calling information treatment for PSTN-to-SIP calls. clid (voice-service-voip)
Passes the network-provided ISDN numbers in an ISDN calling party information element screening indicator field, removes the calling party name and number from the calling-line identifier in voice service voip configuration mode, or allows a presentation of the calling number by substituting for the missing Display Name field in the Remote-Party-ID and From headers.
voice-class sip privacy Sets privacy support at the dial-peer configuration level as defined in RFC 3323.
Cisco IOS Voice Command Reference
VR-1546
Cisco IOS Voice Commands: P privacy-policy
privacy-policy To configure the privacy header policy options at the global level, use the privacy-policy command in voice service VoIP SIP configuration mode. To disable privacy header policy options, use the no form of this command. privacy-policy {passthru | send-always | strip {diversion | history-info}} no privacy-policy {passthru | send-always | strip { diversion | history-info}}
Syntax Description
passthru
Passes the privacy values from the received message to the next call leg.
send-always
Passes a privacy header with a value of None to the next call leg, if the received message does not contain privacy values but a privacy header is required.
strip
Strips the diversion or history-info headers received from the next call leg.
diversion
Strips the diversion headers received from the next call leg.
history-info
Strips the history-info headers received from the next call leg.
Command Default
No privacy-policy settings are configured.
Command Modes
Voice service VoIP SIP configuration (conf-serv-sip)
Command History
Release
Modification
12.4(22)YB
This command was introduced.
15.0(1)M
This command was integrated into Cisco IOS Release 15.0(1)M.
15.1(2)T
This command was modified. The strip, diversion, and history-info keywords were added.
Usage Guidelines
If a received message contains privacy values, use the privacy-policy passthru command to ensure that the privacy values are passed from one call leg to the next. If the received message does not contain privacy values but the privacy header is required, use the privacy-policy send-always command to set the privacy header to None and forward the message to the next call leg. If you want to strip the diversion and history-info from the headers received from the next call leg, use the privacy-policy strip command. You can configure the system to support all the options at the same time.
Examples
The following example shows how to enable the pass-through privacy policy: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# privacy-policy passthru
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Cisco IOS Voice Commands: P privacy-policy
The following example shows how to enable the send-always privacy policy: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# privacy-policy send-always
The following example shows how to enable the strip privacy policy: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# privacy-policy strip diversion Router(conf-serv-sip)# privacy-policy strip history-info
The following example shows how to enable the pass-through, send-always privacy, and strip policies: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# privacy-policy Router(conf-serv-sip)# privacy-policy Router(conf-serv-sip)# privacy-policy Router(conf-serv-sip)# privacy-policy
Related Commands
Command
Description
asserted-id
Sets the privacy level and enables either PAID or PPID privacy headers in outgoing SIP requests or response messages.
voice-class sip privacy-policy
Configures the privacy header policy options at the dial-peer configuration level.
progress_ind To configure an outbound dial peer on a Cisco IOS voice gateway or Cisco Unified Border Element (Cisco UBE) to override and remove or replace the default progress indicator (PI) in specified call messages, use the progress_ind command in dial peer voice configuration mode. To disable removal or replacement of the default PI in specific call messages, use the no form of this command. progress_ind {{alert | callproc} {enable pi-number | disable | strip [strip-pi-number]} | {connect | disconnect | progress | setup} {enable pi-number | disable}} no progress_ind {alert | callproc | connect | disconnect | progress | setup}
Syntax Description
alert
Specifies that the configuration applies to call Alert messages.
callproc
Specifies that the configuration applies to Session Initiation Protocol (SIP) 183 Session In Progress (Call_Proceeding) messages.
connect
Specifies that the configuration applies to call Connect messages.
disconnect
Specifies that the configuration applies to call Disconnect messages.
progress
Specifies that the configuration applies to call Progress messages.
setup
Specifies that the configuration applies to call Setup messages.
enable
Enables user-specified configuration of the progress indicator on the specified call message type.
pi-number
Specifies the PI to be used in place of the default PI. The following are acceptable PI values according to the call message type: •
Alert, Connect, Progress, and SIP 183 Session In Progress messages: 1, 2, or 8.
•
Disconnect messages: 8.
•
Setup messages: 0, 1, or 3.
disable
Disables user-specified configuration of the progress indicator on the specified call message type.
strip
Configures the dial peer to remove all or specific progress indicators in the specified call message type. Note
strip-pi-number
This option applies only to call Alert message on POTS dial peers or to call Proceeding messages on VoIP dial peers.
(optional) Specifies that only a specific PI is to be removed from the specified call message. The value can be 1, 2, or 8.
Command Default
This command is disabled on the outbound dial peer and the default progress indicator received in the incoming call message is passed intact (it is not intercepted, modified, or removed).
Command Modes
Dial peer voice configuration (conf-dial-peer)
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Cisco IOS Voice Commands: P progress_ind
Command History
Usage Guidelines
Release
Modification
12.1(3)XI
This command was introduced on the Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco 7500 series, Cisco MC3810, Cisco AS5300, and Cisco AS5800.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
12.2(1)
This command was modified. Support was added for setup messages from a POTS dial peer.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
15.0(1)XA
This command was modified. Support was added for stripping of PIs in call Alert and SIP 183 Session In Progress (Call_Proceeding) messages.
15.1(1)T
This command was integrated into Cisco IOS Release 5.1(1)T.
Before configuring the progress_ind command on an outbound dial peer, you must configure a destination pattern on the dial peer. To configure a destination pattern for an outbound dial peer, use the destination-pattern command in dial peer voice configuration mode. Once you have set a destination pattern on the dial peer, you can then use the progress_ind command, also in dial peer voice configuration mode, to override and replace or remove the default PI in specific call message types. You can use the progress_ind command to configure replacement behavior on outbound dial peers on a Cisco IOS voice gateway or Cisco UBE to ensure proper end-to-end signaling of VoIP calls. You can also use this command to configure removal (stripping) of PIs on outbound dial peers on Cisco IOS voice gateways or Cisco UBEs, such as when configuring a Cisco IOS SIP gateway (or SIP-SIP Cisco UBE) to not generate additional SIP 183 Session In Progress messages. For messages that contain multiple PIs, behavior configured using the progress_ind command will override only the first PI in the message. Additionally, configuring a replacement PI will not result in an override of the default PI in call Progress messages if the Progress message is sent after a backward cut-through event, such as when an Alert message with a PI of 8 was sent before the Progress message. Use the no progress_ind command in dial peer voice configuration mode to disable PI override configurations on a dial peer on a Cisco IOS voice gateway or Cisco UBE.
Examples
The following example shows how to configure POTS dial peer 3 to override default PIs in call Progress and Connect messages and replace them with a PI of 1: Router(config)# dial-peer Router(config-dial-peer)# Router(config-dial-peer)# Router(config-dial-peer)#
The following example configures outbound VoIP dial peer 1 to override SIP 183 Session In Progress messages and to strip out any PIs with a value of 8: Router(config)# dial-peer voice 1 voip Router(config-dial-peer)# destination-pattern 777 Router(config-dial-peer)# progress_ind callproc strip 8
Related Commands
Command
Description
destination-pattern
Specifies the destination pattern (prefix or full E.164 telephone number) to be used on an outbound dial peer.
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Cisco IOS Voice Commands: P protocol mode
protocol mode To configure the Cisco IOS Session Initiation Protocol (SIP) stack, use the protocol mode command in SIP user-agent configuration mode. To disable the configuration, use the no form of this command. protocol mode {ipv4 | ipv6 | dual-stack [preference {ipv4 | ipv6}]} no protocol mode
Syntax Description
ipv4
Specifies the IPv4-only mode.
ipv6
Specifies the IPv6-only mode.
dual-stack
Specifies the dual-stack (that is, IPv4 and IPv6) mode.
preference {ipv4 | ipv6}
(Optional) Specifies the preferred dual-stack mode, which can be either IPv4 (the default preferred dual-stack mode) or IPv6.
Command Default
No protocol mode is configured. The Cisco IOS SIP stack operates in IPv4 mode when the no protocol mode or protocol mode ipv4 command is configured.
Command Modes
SIP user-agent configuration (config-sip-ua)
Command History
Release
Modification
12.4(22)T
This command was introduced.
Usage Guidelines
The protocol mode command is used to configure the Cisco IOS SIP stack in IPv4-only, IPv6-only, or dual-stack mode. For dual-stack mode, the user can (optionally) configure the preferred family, IPv4 or IPv6. For a particular mode (for example, IPv6-only), the user can configure any address (for example, both IPv4 and IPv6 addresses) and the system will not hide or restrict any commands on the router. SIP chooses the right address for communication based on the configured mode on a per-call basis. For example, if the domain name system (DNS) reply has both IPv4 and IPv6 addresses and the configured mode is IPv6-only (or IPv4-only), the system discards all IPv4 (or IPv6) addresses and tries the IPv6 (or IPv4) addresses in the order they were received in the DNS reply. If the configured mode is dual-stack, the system first tries the addresses of the preferred family in the order they were received in the DNS reply. If all of the addresses fail, the system tries addresses of the other family.
Examples
The following example configures dual-stack as the protocol mode: Router(config-sip-ua)# protocol mode dual-stack
The following example configures IPv6 only as the protocol mode: Router(config-sip-ua)# protocol mode ipv6
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Cisco IOS Voice Commands: P protocol mode
The following example configures IPv4 only as the protocol mode: Router(config-sip-ua)# protocol mode ipv4
The following example configures no protocol mode: Router(config-sip-ua)# no protocol mode
Related Commands
Command
Description
sip ua
Enters SIP user-agent configuration mode.
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Cisco IOS Voice Commands: P protocol rlm port
protocol rlm port To configure the RLM port number, use the protocol rlm port RLM configuration command. To disable this function, use the no form of this command. protocol rlm port port-number no protocol rlm port port-number
Syntax Description
port-number
Command Default
3000
Command Modes
RLM configuration
Command History
Release
Modification
11.3(7)
This command was introduced.
Usage Guidelines
The port number for the basic RLM connection can be reconfigured for the entire RLM group. Table 35 lists the default RLM port numbers. Table 35
Related Commands
RLM port number. See Table 35 for the port number choices.
Default RLM Port Number
Protocol
Port Number
RLM
3000
ISDN
Port[RLM]+1
Command
Description
clear interface
Resets the hardware logic on an interface.
clear rlm group
Clears all RLM group time stamps to zero.
interface
Defines the IP addresses of the server, configures an interface type, and enters interface configuration mode.
link (RLM)
Specifies the link preference.
retry keepalive
Allows consecutive keepalive failures a certain amount of time before the link is declared down.
server (RLM)
Defines the IP addresses of the server.
show rlm group statistics
Displays the network latency of the RLM group.
show rlm group status
Displays the status of the RLM group.
show rlm group timer
Displays the current RLM group timer values.
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Cisco IOS Voice Commands: P protocol rlm port
Command
Description
shutdown (RLM)
Shuts down all of the links under the RLM group.
timer
Overwrites the default setting of timeout values.
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Cisco IOS Voice Commands: P proxy h323
proxy h323 To enable the proxy feature on your router, use the proxy h323 command in global configuration mode. To disable the proxy feature, use the no form of this command. proxy h323 no proxy h323
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled
Command Modes
Global configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2500 series and Cisco 3600 series.
Usage Guidelines
If the multimedia interface is not enabled using this command or if no gatekeeper is available, starting the proxy allows it to attempt to locate these resources. No calls are accepted until the multimedia interface and the gatekeeper are found.
Examples
The following example turns on the proxy feature: proxy h323
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Cisco IOS Voice Commands: P proxy h323
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Cisco IOS Voice Commands: Q This chapter contains commands to configure and maintain Cisco IOS voice applications. The commands are presented in alphabetical order. Some commands required for configuring voice may be found in other Cisco IOS command references. Use the command reference master index or search online to find these commands. For detailed information on how to configure these applications and features, refer to the Cisco IOS Voice Configuration Guide.
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Cisco IOS Voice Commands: Q q850-cause
q850-cause To map a Q.850 call-disconnect cause code to a different Q.850 call-disconnect cause code, use the q850-cause command in application-map configuration mode. To disable the code-to-code mapping, use the no form of this command. q850-cause code-id q850-cause code-id no q850-cause code-id q850-cause code-id
Syntax Description
code-id
Command Default
No mapping occurs.
Command Modes
Application-map
Command History
Release
Modification
12.4(9)T
This command was introduced.
Usage Guidelines
Q.850 call-disconnect cause code to be mapped. Range: 1 to 127.
Use this command to map a Q.850 call-disconnect cause code to any different Q.850 call-disconnect cause code. Use this command in conjunction with the application and map commands. This command operates only on incoming H.323 call legs that are disconnected by a call-control application.
Examples
The following example maps cause code 34 to cause code 17: Router(config)# application Router(config-app)# map Router(config-app-map)# q850-cause 34 q850-cause 17
Related Commands
Command
Description
application
Enables a specific application on a dial peer.
map
Enables mapping.
map q850-cause
Maps a Q.850 call-disconnect cause code to a tone.
progress_ind
Sets a specific progress indicator in Call Setup, Progress, or Connect messages from an H.323 VoIP gateway.
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Cisco IOS Voice Commands: Q qsig decode
qsig decode To enable decoding for QSIG supplementary services, use the qsig decode command in voice service configuration mode. To reset to the default, use the no form of this command. qsig decode no qsig decode
Syntax Description
This command has no keywords or arguments.
Command Default
QSIG decoding is disabled.
Command Modes
Voice service configuration
Command History
Release
Modification
12.4(4)XC
This command was introduced.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T.
Usage Guidelines
This command decodes application protocol data units (APDUs) for supplementary services. If this command is not enabled, data units are not interpreted and are tunneled through the router.
Examples
The following example enables QSIG decoding: Router(config)# voice service voip Router(conf-voi-serv)# qsig decode
query-interval To configure the interval at which the local border element (BE) queries the neighboring BE, use the query-interval command in Annex G Neighbor BE Configuration mode. To remove the interval, use the no form of this command. query-interval query-interval no query-interval
Syntax Description
query-interval
Defaults
30 minutes
Command Modes
Annex G Neighbor BE configuration
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
Frequency, in minutes, at which this BE should query the specified neighbor BE for descriptors. Default is 30. A value of 0 disables periodic querying.
Usage Guidelines
Use this command to configure the interval at which the local BE queries the neighboring BE. Use this command only if you want a query interval other than 30 minutes.
Examples
The following example sets the query interval to 45 minutes: Router(config-annexg-neigh)# query-interval 45
Related Commands
Command
Description
emulate
Configures the local BE to cache the descriptors received from its neighbors. If caching is enabled, the neighbors are queried at the specified interval for their descriptors.
local
Configures the identifier for the neighbor BE.
session transport
Configures the neighbor’s port number that is used for exchanging Annex G messages.
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Cisco IOS Voice Commands: R This chapter contains commands to configure and maintain Cisco IOS voice applications. The commands are presented in alphabetical order. Some commands required for configuring voice may be found in other Cisco IOS command references. Use the command reference master index or search online to find these commands. For detailed information on how to configure these applications and features, refer to the Cisco IOS Voice Configuration Library.
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Cisco IOS Voice Commands: R radius-server attribute 6
radius-server attribute 6 To provide for the presence of the Service-Type attribute (attribute 6) in RADIUS Access-Accept messages, use the radius-server attribute 6 command in global configuration mode. To make the presence of the Service-Type attribute optional in Access-Accept messages, use the no form of this command. radius-server attribute 6 {mandatory | on-for-login-auth | support-multiple | voice value} no radius-server attribute 6 {mandatory | on-for-login-auth | support-multiple | voice value}
Syntax Description
mandatory
Makes the presence of the Service-Type attribute mandatory in RADIUS Access-Accept messages.
on-for-login-auth
Sends the Service-Type attribute in the authentication packets. Note
The Service-Type attribute is sent by default in RADIUS Accept-Request messages. Therefore, RADIUS tunnel profiles should include “Service-Type=Outbound” as a check item, not just as a reply item. Failure to include Service-Type=Outbound as a check item can result in a security hole.
support-multiple
Supports multiple Service-Type values for each RADIUS profile.
voice value
Selects the Service-Type value for voice calls. The only value that can be entered is 1. The default is 12.
Command Default
If this command is not configured, the absence of the Service-Type attribute is ignored, and the authentication or authorization does not fail. The default for the voice keyword is 12.
Command Modes
Global configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
12.2(13)T
The mandatory keyword was added.
Usage Guidelines
If this command is configured and the Service-Type attribute is absent in the Access-Accept message packets, the authentication or authorization fails. The support-multiple keyword allows for multiple instances of the Service-Type attribute to be present in an Access-Accept packet. The default behavior is to disallow multiple instances, which results in an Access-Accept packet containing multiple instances being treated as though an Access-Reject was received.
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Cisco IOS Voice Commands: R radius-server attribute 6
Examples
The following example shows that the presence of the Service-Type attribute is mandatory in RADIUS Access-Accept messages: Router(config)# radius-server attribute 6 mandatory
The following example shows that attribute 6 is to be sent in authentication packets: Router(config)# radius-server attribute 6 on-for-login-auth
The following example shows that multiple Service-Type values are to be supported for each RADIUS profile: Router(config)# radius-server attribute 6 support-multiple
The following example shows that Service-Type values are to be sent in voice calls: Router(config)# radius-server attribute 6 voice 1
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Cisco IOS Voice Commands: R rai target
rai target To configure the Session Initiation Protocol (SIP) Resource Allocation Indication (RAI) mechanism, use the rai target command in SIP UA configuration mode. To disable SIP RAI configuration, use the no form of this command. rai target target-address resource-group group-index [transport [tcp [tls [scheme {sip | sips}]] | udp]] no rai target target-address
Syntax Description
target-address
IPv4, IPv6, or Domain Name Server (DNS) target address to which the status of the gateway resources are reported. The format of the target address can be one of the following: •
ipv4:ipv4-address
•
ipv6:ipv6-address
•
dns:domain-name
resource-group
Maps the target address with the resource group index.
group-index
Resource group index. The range is from 1 to 5.
transport
(Optional) Specifies the mechanism to transport the RAI information.
tcp
(Optional) Transports the RAI information through Transmission Control Protocol (TCP).
tls
(Optional) Transports the RAI information through Transport Layer Security (TLS).
scheme
(Optional) Specifies the URL scheme for outgoing messages.
sip
(Optional) Selects SIP URL in outgoing OPTIONS message.
sips
(Optional) Selects Secure SIP (SIPS) URL in outgoing OPTIONS message.
udp
(Optional) Transports the RAI information through Unified Datagram Protocol (UDP).
Command Default
The SIP RAI mechanism is disabled.
Command Modes
SIP UA configuration (config-sip-ua)
Command History
Release
Modification
15.1(2)T
This command was introduced.
Usage Guidelines
Use the rai target command to provide the details of SIP along with the index of the resource group that needs to be monitored for reporting over SIP trunk. A maximum of five RAI configurations can be applied for other destination targets or monitoring entities. However, only one RAI configuration is possible for one target address.
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Cisco IOS Voice Commands: R rai target
Examples
The following example shows how to enable reporting of SIP RAI information over TCP to a target address of example.com: Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# rai target dns:example.com resource-group 1
Related Commands
Command
Description
debug rai
Enables debugging for Resource Allocation Indication (RAI).
periodic-report interval
Configures periodic reporting parameters for gateway resource entities.
resource (voice)
Configures parameters for monitoring resources, use the resource command in voice-class configuration mode.
show voice class resource-group
Displays the resource group configuration information for a specific resource group or all resource groups.
voice class resource-group
Enters voice-class configuration mode and assigns an identification tag number for a resource group.
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Cisco IOS Voice Commands: R random-contact
random-contact To populate an outgoing INVITE message with random-contact information (instead of clear-contact information), use the random-contact command in voice service VoIP SIP configuration mode. To disable random-contact information, use the no form of this command. random-contact no random-contact
Syntax Description
This command has no arguments or keywords.
Command Default
Outgoing INVITE messages are populated with clear-contact information.
Command Modes
Voice service VoIP SIP configuration (conf-serv-sip)
Command History
Release
Modification
12.4(22)YB
This command was introduced.
15.0(1)M
This command was integrated into Cisco IOS Release 15.0(1)M.
Usage Guidelines
To populate outbound INVITE messages from the Cisco Unified Border Element with random-contact information instead of clear-contact information, use the random-contact command. This functionality will work only when the Cisco Unified Border Element is configured for Session Initiation Protocol (SIP) registration with random contact using the credentials and registrar commands.
Examples
The following example shows how to populate outbound INVITE messages with random-contact information: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# random-contact
Related Commands
Command
Description
credentials (sip ua)
Sends a SIP registration message from a Cisco Unified Border Element in the UP state.
registrar
Enables SIP gateways to register E.164 numbers on behalf of FXS, EFXS, and SCCP phones with an external SIP proxy or SIP registrar.
voice-class sip random-contact
Populates the outgoing INVITE message with random-contact information at the dial-peer level.
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Cisco IOS Voice Commands: R random-request-uri validate
random-request-uri validate To enable the validation of the called number based on the random value generated during the registration of the number, use the random-request-uri validate command in voice service VoIP SIP configuration mode. To disable validation, use the no form of this command. random-request-uri validate no random-request-uri validate
Syntax Description
This command has no keywords or arguments.
Command Default
Validation is disabled.
Command Modes
Voice service voip sip configuration (conf-serv-sip)
Command History
Release
Modification
12.4(22)YB
This command was introduced.
15.0(1)M
This command was integrated into Cisco IOS Release 15.0(1)M.
Usage Guidelines
The system generates a random string when registering a new number. An INVITE message with the P-Called-Party-ID value can have the Request-URI set to this random number. To enable the system to identify the called-number from the random number in the Request-URI, use the random-request-uri validate command. If the P-Called-Party-ID is not set in the INVITE message, the Request URI for that message must contain the called party information (and cannot contain a random number). Therefore validation is performed only on INVITE messages with a P-Called-Party-ID.
Examples
The following example shows how to enable called-number validation at the global configuration level: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# random-request-uri validate
Related Commands
Command
Description
credentials (sip ua)
Sends a SIP registration message from a Cisco Unified Border Element in the UP state.
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Cisco IOS Voice Commands: R random-request-uri validate
Command
Description
register
Enables SIP gateways to register E.164 numbers on behalf of FXS, EFXS, and SCCP phones with an external SIP proxy or SIP registrar.
voice-class sip random-request-uri validate
Validates the called number based on the random value generated during the registration of the number at the dial-peer configuration level.
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Cisco IOS Voice Commands: R ras retry
ras retry To configure the H.323 Registration, Admission, and Status (RAS) message retry counters, use the ras retry command in voice service h323 configuration mode. To set the counters to the default values, use the no form of this command. ras retry {all | arq | brq | drq | grq | rai | rrq} value no ras retry {all | arq | brq | drq | grq | rai | rrq}
Syntax Description
all
Configures all RAS message counters that do not have explicit values configured individually. If no ras retry all is entered, all values are set to the default except for the individual values that were configured separately.
arq
Configures the admission request (ARQ) message counter.
brq
Configures the bandwidth request (BRQ) message counter.
drq
Configures the disengage request (DRQ) message counter.
grq
Configures the gatekeeper request (GRQ) message counter.
rai
Configures the resource availability indication (RAI) message counter.
rrq
Configures the registration request (RRQ) message counter.
value
Number of times for the gateway to resend messages to the gatekeeper after the timeout period. The timeout period is the period in which a message has not been received by the gateway from the gatekeeper and is configured using the ras timeout command. Valid values are 1 through 30.
Use this command in conjunction with the ras timeout command. The ras timeout command configures the number of seconds for the gateway to wait before resending a RAS message to a gatekeeper. The ras retry command configures the number of times to resend the RAS message after the timeout period expires. The default values for timeouts and retries are acceptable in most networks. You can use these commands if you are experiencing problems in RAS message transmission between gateways and
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Cisco IOS Voice Commands: R ras retry
gatekeepers. For example, if you have gatekeepers that are slow to respond to a type of RAS request, increasing the timeout value and the number of retries increases the call success rate, preventing lost billing information and unnecessary switchover to an alternate gatekeeper.
Examples
The following example shows the GRQ message counter set to 5 and all other RAS message counters set to 10: Router(conf-serv-h323)# ras retry all 10 Router(conf-serv-h323)# ras retry grq 5
Related Commands
Command
Description
ras timeout
Configures the H.323 RAS message timeout values.
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Cisco IOS Voice Commands: R ras retry lrq
ras retry lrq To configure the gatekeeper Registration, Admission, and Status (RAS) message retry counters, use the ras retry lrq command in gatekeeper configuration mode. To set the counters to the default values, use the no form of this command. ras retry lrq value no ras retry lrq
Syntax Description
lrq
Configures the location request (LRQ) message counter.
value
Number of times for the zone gatekeeper (ZGK) to resend messages to the directory gatekeeper (DGK) after the timeout period. The timeout period is the period in which a message has not been received by the ZKG from the DGK and is configured using the ras timeout lrq command. Valid values are 1 through 30.
Command Default
The retry counter is set to1.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.4(4)T
This command was introduced.
Usage Guidelines
Use this command in conjunction with the ras timeout lrq command. The ras timeout lrq command configures the number of seconds for the gateway to wait before resending a RAS message to a gatekeeper. The ras retry lrq command configures the number of times to resend the RAS message after the timeout period expires. The default values for timeouts and retries are acceptable in most networks. You can use these commands if you are experiencing problems in RAS message transmission between gateways and gatekeepers. For example, if you have gatekeepers that are slow to respond to a type of RAS request, increasing the timeout value and the number of retries increases the call success rate, preventing lost billing information and unnecessary switchover to an alternate gatekeeper.
Examples
The following example shows the LRQ message counter set to 5: Router(conf-gk)# ras retry lrq 5
Related Commands
Command
Description
ras timeout lrq
Configures the gatekeeper RAS message timeout values.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R ras rrq dynamic prefixes
ras rrq dynamic prefixes To enable advertisement of dynamic prefixes in additive registration request (RRQ) RAS messages on the gateway, use the ras rrq dynamic prefixes command in voice service h323 configuration mode. To disable advertisement of dynamic prefixes in additive RRQ messages, use the no form of this command. ras rrq dynamic prefixes no ras rrq dynamic prefixes
Syntax Description
This command has no arguments or keywords.
Command Default
In Cisco IOS Release 12.2(15)T, the default was set to enabled. In Cisco IOS Release 12.3(3), the default is set to disabled.
Command Modes
Voice service h323 configuration
Command History
Release
Modification
12.2(15)T
This command was introduced.
12.3(3)
The default is modified to be disabled by default.
12.3(4)T
The default change implemented in Cisco IOS Release 12.3(3) was integrated in Cisco IOS Release 12.3(4)T.
Usage Guidelines
In Cisco IOS Release 12.2(15)T, the default for the ras rrq dynamic prefixes command was set to enabled so that the gateway automatically sent dynamic prefixes in additive RRQ messages to the gatekeeper. Beginning in Cisco IOS Release 12.3(3), the default is set to disabled, and you must specify the command to enable the functionality.
Examples
The following example allows the gateway to send advertisements of dynamic prefixes in additive RRQ messages to the gatekeeper: Router(conf-serv-h323)# ras rrq dynamic prefixes
Related Commands
Command
Description
rrq dynamic-prefixes-accept
Enables processing of additive RRQ messages and dynamic prefixes on the gatekeeper.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R ras rrq ttl
ras rrq ttl To configure the H.323 Registration, Admission, and Status (RAS) registration request (RRQ) time-to-live value, use the ras rrq ttl command in voice service h323 configuration mode. To set the RAS RRQ time-to-live value to the default value, use the no form of this command. ras rrq ttl time-to-live seconds [margin seconds] no ras rrq ttl
Syntax Description
time-to-live seconds
Number of seconds that the gatekeeper should consider the gateway active. Valid values are 15 through 4000. The time-to-live seconds value must be greater than the margin seconds value.
margin seconds
(Optional) The number of seconds that an RRQ message can be transmitted from the gateway before the time-to-live seconds value advertised to the gatekeeper. Valid values are 1 through 60. The margin seconds value times two must be less than or equal to the time-to-live seconds value.
The maximum time-to-live value was changed from 300 to 4000 seconds.
12.3(4)T2
The maximum time-to-live value was changed from 300 to 4000 seconds.
12.3(7)T
The maximum time-to-live value was changed from 300 to 4000 seconds.
Usage Guidelines
Use this command to configure the number of seconds that the gateway should be considered active by the gatekeeper. The gateway transmits this value in the RRQ message to the gatekeeper. The margin time keyword and argument allow the gateway to transmit an early RRQ to the gatekeeper before the time-to-live value advertised to the gatekeeper.
Examples
The following example shows the time-to-live seconds value configured to 300 seconds and the margin seconds value configured to 60 seconds: Router(conf-serv-h323)# ras rrq ttl 300 margin 60
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R ras timeout
ras timeout To configure the H.323 Registration, Admission, and Status (RAS) message timeout values, use the ras timeout command in voice service h323 configuration mode. To set the timers to the default values, use the no form of this command. ras timeout {all | arq | brq | drq | grq | rai | rrq} seconds no ras timeout {all | arq | brq | drq | grq | rai | rrq}
Syntax Description
all
Configures message timeout values for all RAS messages that do not have explicit values configured individually. If no ras timeout all is entered, all values are set to the default except for the individual values that were configured separately.
arq
Configures the admission request (ARQ) message timer.
brq
Configures the bandwidth request (BRQ) message timer.
drq
Configures the disengage request (DRQ) message timer.
grq
Configures the gatekeeper request (GRQ) message timer.
rai
Configures the resource availability indication (RAI) message timer.
rrq
Configures the registration request (RRQ) message timer.
seconds
Number of seconds for the gateway to wait for a message from the gatekeeper before timing out. Valid values are 1 through 45.
Use this command in conjunction with the ras retry command. The ras timeout command configures the number of seconds for the gateway to wait before resending a RAS message to a gatekeeper. The ras retry command configures the number of times to resend the RAS message after the timeout period expires. The default values for timeouts and retries are acceptable in most networks. You can use these commands if you are experiencing problems in RAS message transmission between gateways and gatekeepers. For example, if you have gatekeepers that are slow to respond to a type of RAS request, increasing the timeout value and the number of retries increases the call success rate, preventing lost billing information and unnecessary switchover to an alternate gatekeeper.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R ras timeout
Examples
The following example shows the GRQ message timeout value set to 10 seconds and all other RAS message timeout values set to 7 seconds: Router(conf-serv-h323)# ras timeout grq 10 Router(conf-serv-h323)# ras timeout all 7
Related Commands
Command
Description
ras retry
Configures the H.323 RAS message retry counters.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R ras timeout decisec
ras timeout decisec To configure the H.323 Registration, Admission, and Status (RAS) message timeout values in deciseconds, use the ras timeout decisec command in voice service h323 configuration mode. To set the timers to the default values, use the no form of this command. ras timeout {all | arq | brq | drq | grq | rai | rrq} decisec decisecond no ras timeout {all | arq | brq | drq | grq | rai | rrq} decisec
Syntax Description
all
Configures message timeout values for all RAS messages that do not have explicit values configured individually. If no ras timeout all is entered, all values are set to the default except for the individual values that were configured separately.
arq
Configures the admission request (ARQ) message timer. Default: 3.
brq
Configures the bandwidth request (BRQ) message timer. Default: 3.
drq
Configures the disengage request (DRQ) message timer.Default: 3.
grq
Configures the gatekeeper request (GRQ) message timer. Default: 5.
rai
Configures the resource availability indication (RAI) message timer. Default: 3.
rrq
Configures the registration request (RRQ) message timer. Default: 5.
decisecond
Number of deciseconds for the gateway to wait for a message from the gatekeeper before timing out. Valid values are 1 through 45.
Command Default
Timers are set to their default values.
Command Modes
Voice service h323 configuration
Command History
Release
Modification
12.4(4)T
This command was introduced.
Usage Guidelines
Use this command in conjunction with the ras retry command. The ras timeout decisec command configures the number of deciseconds for the gateway to wait before resending a RAS message to a gatekeeper. The ras retry command configures the number of times to resend the RAS message after the timeout period expires. The default values for timeouts and retries are acceptable in most networks. You can use these commands if you are experiencing problems in RAS message transmission between gateways and gatekeepers. For example, if you have gatekeepers that are slow to respond to a type of RAS request, increasing the timeout value and the number of retries increases the call success rate, preventing lost billing information and unnecessary switchover to an alternate gatekeeper.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R ras timeout decisec
Examples
The following example shows the ARQ message timeout value set to 25 deciseconds and all other RAS message timeout values set to 30 deciseconds: Router(conf-serv-h323)# ras timeout arq decisec 25 Router(conf-serv-h323)# ras timeout all decisec 30
Related Commands
Command
Description
ras retry
Configures the H.323 RAS message retry counters.
ras timeout
Configures the H.323 RAS message timeout values in seconds.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R ras timeout lrq
ras timeout lrq To configure the Gatekeeper Registration, Admission, and Status (RAS) message timeout values, use the ras timeout lrq command in gatekeeper configuration mode. To set the timers to the default values, use the no form of this command. ras timeout lrq seconds no ras timeout lrq
Syntax Description
lrq
Configures the location request (LRQ) message timer.
seconds
Number of seconds for the zone gatekeeper (ZGK) to wait for a message from the directory gatekeeper (DGK) before timing out. Valid values are 1 through 45. The default is 2.
Command Default
Timers are set to their default value
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.4(4)T
This command was introduced.
Usage Guidelines
Use this command in conjunction with the ras retry lrq command. The ras timeout lrq command configures the number of seconds for the zone gatekeeper (ZGK) to wait before resending a RAS message to a directory gatekeeper (DGK). The ras retry lrq command configures the number of times to resend the RAS message after the timeout period expires. The default values for timeouts and retries are acceptable in most networks. You can use these commands if you are experiencing problems in RAS message transmission between gatekeepers. For example, if you have gatekeepers that are slow to respond to a LRQ RAS request, increasing the timeout value and the number of retries increases the call success rate, preventing lost billing information and unnecessary switchover to an alternate gatekeeper.
Examples
The following example shows the LRQ message timeout value set to 4 seconds: Router(conf-gk)# ras timeout lrq 4
Related Commands
Command
Description
ras retry lrq
Configures the gatekeeper RAS message retry counters.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R rbs-zero
rbs-zero To enable 1AESS switch support for T1 lines on the primary serial interface of an access server, use the rbs-zero command in serial interface configuration mode. To disable IAESS switch support, use the no form of this command. rbs-zero [nfas-int nfas-int-range] no rbs-zero [nfas-int nfas-int-range]
Syntax Description
nfas-int nfas-int-range
Command Default
1AESS switch support is disabled.
Command Modes
Serial interface configuration
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command supports the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5800.
Usage Guidelines
(Optional) Non-Facility Associated Signaling (NFAS) interface number. Range is from 0 to 32.
Use this command to configure the primary serial interface of an access server connected to T1 lines to support 1AESS switches for dial-in and dial-out calls. Modem calls of 56K or a lower rate are accepted; 64K calls are rejected. In IAESS mode, the following occurs: •
Modem calls are accepted and digital calls are rejected.
•
The ABCD bit of the 8 bits in the incoming calls is ignored. The ABCD bit of the 8 bits in the outgoing modem calls is set to 0.
In non-1AESS mode, modem and digital calls are accepted.
Examples
The following example enables 1AESS switching support on T1 channel 0: Router(config)# controller Router(config-controller)# Router(config-controller)# Router(config-controller)#
Router(config)# interface serial 1/0:23 Router(config-if)# no ip address Router(config-if)# isdn switch-type primary-ni Router(config-if)# rbs-zero nfas-int 0
Related Commands
Command
Description
interface serial
Enters serial interface configuration mode.
isdn switch-type
Sets the switch type.
pri-group timeslots
Configures the PRI trunk for a designated operation.
show controllers t1
Displays information about the T1 links and the hardware and software driver information for the T1 controller.
show isdn nfas group
Displays all the members of a specified NFAS group or all NFAS groups.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R reason-header override
reason-header override To enable cause code passing from one SIP leg to another, use the reason-header override command in SIP UA configuration mode. To disable reason-header override, use the no form of this command. reason-header override no reason-header override
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values.
Command Modes
SIP UA configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.4(9)T
Usage guidelines were updated to include configuration requirements for SIP-to-SIP configurations.
Usage Guidelines
In an SIP-to-SIP configuration the reason-header override command must be configured to ensure cause code passing from the incoming SIP leg to the outgoing SIP leg.
Examples
The following example, shows the SIP user agent with reason-header override being configured. Router(config)# sip-ua Router(config-sip-ua)# reason-header override
Related Commands
Command
Description
sip-ua
Enables SIP UA configuration commands.
Cisco IOS Voice Command Reference
VR-1579
Cisco IOS Voice Commands: R redial
redial To define speed-dial code for a Feature Speed-dial (FSD) to redial the last number dialed, use the redial command in STC application feature speed-dial configuration mode. To return the code to its default, use the no form of this command. redial keypad-character no redial
Syntax Description
keypad-character
Character string that can be dialed on a telephone keypad (0-9, *, #). Default: #. Before Cisco IOS Release 12.4(20)YA, this is a single character. In Cisco IOS Release 12.5(20)YA and later releases, the string can be any of the following: •
A single character (0-9, *, #)
•
Two digits (00-99)
•
Two to four characters (0-9, *, #) and the leading or ending character must be an asterisk (*) or number sign (#)
In Cisco IOS Release 15.0(1)M and later releases, the string can also be any of the following: •
The length of the keypad-character argument was changed to 1 to 4 characters.
12.4(22)T
This command was integrated into Cisco IOS Release 12.4(22)T.
15.0(1)M
This command was modified.
Cisco IOS Voice Command Reference
VR-1580
Cisco IOS Voice Commands: R redial
Usage Guidelines
This command changes the value of the speed-dial code for Redial from the default (#) to the specified value. In Cisco IOS Release 12.4(20)YA and later releases, if the length of the keypad-character argument is at least two characters and the leading or ending character of the string is an asterisk (*) or a number sign (#), phone users are not required to dial a prefix to access this speed dial. Typically, phone users dial a Feature Speed-dial (FSD) consisting of a prefix plus a speed-dial code, for example *#. If the feature code is 78#, the phone user dials only 78#, without the FSD prefix, to access the corresponding feature. In Cisco IOS Release 15.0(1)M and later releases, if the length of the keypad-character argument is three or four digits, phone users are not required to dial a prefix or any special characters to access this feature. Typically, phone users dial a special feature access code (FAC) consisting of a prefix plus a feature code, for example **2. If the feature code is 788, the phone user dials only 788, without the FAC prefix, to access the corresponding feature. In Cisco IOS Release 12.4(20)YA and later releases, if you attempt to configure this command with a value that is already being used for a feature access code (FAC) or another FSD, you receive a message. If you configure a duplicate code, the system implements the first matching feature in the order of precedence shown in the output of the show stcapp feature codes command. In Cisco IOS Release 12.4(20)YA and later releases, if you attempt to configure this command with a value that precludes or is precluded by a feature code for a FAC or another FSD, you receive a message. If you configure this command with a value that precludes or is precluded by another code, the system always executes the call feature with the shortest code and ignores the longer code. For example, #1 will always preclude #12 and #123. You must configure a new value for the precluded code in order to enable access to that feature. To display a list of all FACs and FSDs, use the show stcapp feature codes command.
Examples
The following example shows how to change the value of the speed-dial code for Redial from the default (#). In this configuration, a phone user must press ** on the keypad to redial the number that was most recently dialed on this line, regardless of what value is configured for the FSD prefix. Router(config)# stcapp feature speed-dial Router(config-stcapp-fsd)# redial ** Router(config-stcapp-fsd)# exit
Related Commands
Command
Description
digit
Designates the number of digits for feature speed-dial codes (FSDs).
prefix (stcapp-fsd)
Defines the prefix for feature speed-dials (FSDs).
show stcapp feature codes
Displays all feature access codes (FACs) and feature access codes (FSDs) that are available for the STC application.
speed dial
Designates a range of speed-dial codes for the STC application.
stcapp feature speed-dial
Enables feature speed-dials (FSDs) in STC application and enters STC application feature speed-dial configuration mode for changing values of the prefix and speed-dial codes from the default.
Cisco IOS Voice Command Reference
VR-1581
Cisco IOS Voice Commands: R redirect contact order
redirect contact order To set the order of contacts in the 300 Multiple Choice message, use the redirect contact order command in SIP configuration mode. To reset the order of contacts to the default, use the no form of this command. redirect contact order [best-match | longest-match] no redirect contact order
Syntax Description
best-match
(Optional) Uses the current system configuration.
longest-match
(Optional) Uses the destination pattern longest match first, and then the second longest match, the third longest match, and so on. This is the default.
Command Default
longest-match
Command Modes
SIP configuration
Command History
Release
Modification
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
Usage Guidelines
This command applies when a 300 Multiple Choice message is sent by a SIP gateway indicating that a call has been redirected and that there are multiple routes to the destination. Enter SIP configuration mode after entering voice service VoIP configuration mode as shown in the following example.
Examples
The following example uses the current system configuration to set the order of contact: Router(config)# voice service voip Router(config-voi-srv)# sip Router(conf-serv-sip)# redirect contact order best-match
Related Commands
Command
Description
sip
Enters SIP configuration mode.
Cisco IOS Voice Command Reference
VR-1582
Cisco IOS Voice Commands: R redirect ip2ip (dial peer)
redirect ip2ip (dial peer) To redirect SIP phone calls to SIP phone calls on a specific VoIP dial peer using the Cisco IOS Voice Gateway, use the redirect ip2ip command in dial peer configuration mode. To disable redirection, use the no form of this command. redirect ip2ip no redirect ip2ip
Syntax Description
This command has no arguments or keywords.
Command Default
Redirection is disabled.
Command Modes
Dial peer configuration
Command History
Release
Modification
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
Usage Guidelines
The redirect ip2ip command must be configured on the inbound dial peer of the gateway. This command enables, on a per dial peer basis, IP-to-IP call redirection for the gateway. To enable global IP-to-IP call redirection for all VoIP dial peers, use voice service configuration mode. To specify IP-to-IP call redirection for a specific VoIP dial peer, configure the dial peer in dial peer configuration mode.
Note
Examples
When IP-to-IP redirection is configured in dial peer configuration mode, the configuration for the specific dial peer is activated only if the dial peer is an inbound dial peer. To enable IP-to-IP redirection globally, use redirect ip2ip (voice service) command.
The following example specifies that on VoIP dial peer 99, IP-to-IP redirection is set: dial-peer voice 99 voip redirect ip2ip
Related Commands
Command
Description
redirect ip2ip (voice service)
Redirects SIP phone calls to SIP phone calls globally on a gateway using the Cisco IOS voice gateway.
Cisco IOS Voice Command Reference
VR-1583
Cisco IOS Voice Commands: R redirect ip2ip (voice service)
redirect ip2ip (voice service) To redirect SIP phone calls to SIP phone calls globally on a gateway using the Cisco IOS Voice Gateway, use the redirect ip2ip command in voice service configuration mode. To disable redirection, use the no form of this command. redirect ip2ip no redirect ip2ip
Syntax Description
This command has no arguments or keywords.
Command Default
Redirection is disabled.
Command Modes
Voice service configuration
Command History
Release
Modification
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
Usage Guidelines
Use this command to enable IP-to-IP call redirection globally on a gateway. Use the redirect ip2ip (dial peer) command to configure IP-to-IP redirection on a specific inbound dial peer.
Examples
The following example specifies that all VoIP dial peers use IP-to-IP redirection: voice service voip redirect ip2ip
Related Commands
Command
Description
redirect ip2ip (dial peer)
Redirects SIP phone calls to SIP phone calls on a specific VoIP dial peer using the Cisco IOS voice gateway.
Cisco IOS Voice Command Reference
VR-1584
Cisco IOS Voice Commands: R redirection (SIP)
redirection (SIP) To enable the handling of 3xx redirect messages, use the redirection command in SIP UA configuration mode. To disable the handling of 3xx redirect messages, use the no form of this command. redirection no redirection
Syntax Description
This command has no arguments or keywords.
Command Default
Redirection is enabled.
Command Modes
SIP UA configuration
Command History
Release
Modification
12.2(13)T
This command was introduced.
Usage Guidelines
The redirection command applies to all Session Initiation Protocol (SIP) VoIP dial peers configured on the gateway. The default mode of SIP gateways is to process incoming 3xx redirect messages according to RFC 2543. However if redirect handling is disabled with the no redirection command, the gateway treats the incoming 3xx responses as 4xx error class responses. To reset the default processing of 3xx messages, use the redirection command.
Examples
The following example disables processing of incoming 3xx redirection messages: Router(config)# sip-ua Router(config-sip-ua)# no redirection
Related Commands
Command
Description
show sip-ua statistics
Displays response, traffic, and retry SIP statistics.
show sip-ua status
Displays SIP UA status.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R refer-ood enable
refer-ood enable To enable out-of-dialog refer (OOD-R) processing, use the refer-ood enable command in SIP user-agent configuration mode. To disable OOD-R, use the no form of this command. refer-ood enable [request-limit] no refer-ood enable
Syntax Description
request-limit
Command Default
OOD-R processing is disabled.
Command Modes
SIP UA configuration (config-sip-ua)
Command History
Release
Modification
12.4(11)XJ
This command was introduced.
12.4(15)T
This command was integrated into Cisco IOS Release 12.4(15)T.
(Optional) Maximum number of concurrent incoming OOD-R requests that the router can process. Range: 1 to 500. Default: 500.
Usage Guidelines
Out of dialog Refer allows applications to establish calls using the SIP gateway or Cisco Unified CME. The application sets up the call and the user does not dial out from their own phone.
Examples
The following example shows how to enable OOD-R: Router(config)# sip-ua Router(config-sip-ua)# refer-ood enable
Related Commands
Command
Description
authenticate (voice register global)
Defines the authenticate mode for SIP phones in a Cisco Unified CME or Cisco Unified SRST system.
credential load
Reloads a credential file into flash memory.
debug voip application Displays all application debug messages.
Cisco IOS Voice Command Reference
VR-1586
Cisco IOS Voice Commands: R register e164
register e164 To configure a gateway to register or deregister a fully-qualified dial-peer E.164 address with a gatekeeper, use the register e164 command in dial peer configuration mode. To deregister the E.164 address, use the no form of this command. register e164 no register e164
Syntax Description
This command has no arguments or keywords.
Command Default
No E.164 addresses are registered until you enter this command.
Command Modes
Dial peer configuration
Command History
Release
Modification
12.0(5)T
This command was introduced.
12.1(5)XM2
The command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T. This command is supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400, and the Cisco AS5850 in this release.
Usage Guidelines
Use this command to register the E.164 address of an analog telephone line attached to a foreign exchange station (FXS) port on a router. The gateway automatically registers fully qualified E.164 addresses. Use the no register e164 command to deregister an address. Use the register e164 command to register a deregistered address. Before you automatically or manually register an E.164 address with a gatekeeper, you must create a dial peer (using the dial-peer command), assign an FXS port to the peer (using the port command), and assign an E.164 address using the destination-pattern command. The E.164 address must be a fully qualified address. For example, +5550112, 5550112, and 4085550112 are fully qualified addresses; 408555.... is not. E.164 addresses are registered only for active interfaces, which are those that are not shut down. If an FXS port or its interface is shut down, the corresponding E.164 address is deregistered.
Tip
You can use the show gateway command to find out whether the gateway is connected to a gatekeeper and whether a fully qualified E.164 address is assigned to the gateway. Use the zone-prefix command to define prefix patterns on the gatekeeper, such as 408555...., that apply to one or more gateways.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R register e164
Examples
The following command sequence places the gateway in dial peer configuration mode, assigns an E.164 address to the interface, and registers that address with the gatekeeper. gateway1(config)# dial-peer gateway1(config-dial-peer)# gateway1(config-dial-peer)# gateway1(config-dial-peer)#
voice 111 pots port 1/0/0 destination-pattern 5550112 register e164
The following commands deregister an address with the gatekeeper. gateway1(config)# dial-peer voice 111 pots gateway1(config-dial-peer)# no register e164
The following example shows that you must have a connection to a gatekeeper and must define a unique E.164 address before you can register an address. gateway1(config)# dial-peer voice 222 pots gateway1(config-dial-peer)# port 1/0/0 gateway1(config-dial-peer)# destination 919555.... gateway1(config-dial-peer)# register e164 ERROR-register-e164:Dial-peer destination-pattern is not a full E.164 number gateway1(config-dial-peer)# no gateway gateway1(config-dial-peer)# dial-peer voice 111 pots gateway1(config-dial-peer)# register e164 ERROR-register-e164:No gatekeeper
Related Commands
Command
Description
destination-pattern
Specifies either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer.
dial-peer (voice)
Enters dial peer configuration mode and specifies the method of voice encapsulation.
port (dial peer)
Associates a dial peer with a specific voice port.
show gateway
Displays the current gateway status.
zone prefix
Adds a prefix to the gatekeeper zone list.
Cisco IOS Voice Command Reference
VR-1588
Cisco IOS Voice Commands: R registered-caller ring
registered-caller ring To configure the Nariwake service registered caller ring cadence, use the registered-caller ring command in dial peer configuration mode. registered-caller ring cadence
Syntax Description
cadence
Command Default
The default Nariwake service registered caller ring cadence is ring 1.
Command Modes
Dial peer configuration
Command History
Release
Modification
12.1.(2)XF
This command was introduced on the Cisco 800 series.
Usage Guidelines
A value of 0, 1, or 2. The default ring cadence for registered callers is 1 and for unregistered callers is 0. The on and off periods of ring 0 (normal ringing signals) and ring 1 (ringing signals for the Nariwake service) are defined in the NTT user manual.
If your ISDN line is provisioned for the I Number or dial-in services, you must also configure a dial peer by using the destination-pattern not-provided command. Either port 1 or port 2 can be configured under this dial peer. The router then forwards the incoming call to voice port 1. (See the “Examples” section below. If more than one dial peer is configured with the destination-pattern not-provided command, the router uses the first configured dial peer for the incoming calls. To display the Nariwake ring cadence setting, use the show run command.
Examples
The following example sets the ring cadence for registered callers to 2. pots country jp dial-peer voice 1 pots registered-caller ring 2
Related Commands
Command
Description
destination-pattern not-provided
Specifies the port to receive the incoming calls that have no called-party number.
Cisco IOS Voice Command Reference
VR-1589
Cisco IOS Voice Commands: R registrar
registrar To enable Session Initiation Protocol (SIP) gateways to register E.164 numbers on behalf of analog telephone voice ports (FXS), IP phone virtual voice ports (EFXS), and Skinny Client Control Protocol (SCCP) phones with an external SIP proxy or SIP registrar, use the registrar command in SIP UA configuration mode. To disable registration of E.164 numbers, use the no form of this command. registrar {dhcp | [registrar-index] registrar-server-address [:port]} [auth-realm realm] [expires seconds] [random-contact] [refresh-ratio ratio-percentage] [scheme {sip | sips}] [tcp] [type] [secondary] no registrar [registrar-index | secondary]
Syntax Description
dhcp
(Optional) Specifies that the domain name of the primary registrar server is retrieved from a DHCP server (cannot be used to configure secondary or multiple registrars).
registrar-index
(Optional) A specific registrar to be configured, allowing configuration of multiple registrars (maximum of six). Range is 1 to 6.
registrar-server-address
The SIP registrar server address to be used for endpoint registration. This value can be entered in one of three formats: dns:address—the Domain Name System (DNS) address of the primary SIP registrar server (the dns: delimiter must be included as the first four characters).
•
ipv4:address—the IP address of the SIP registrar server (the ipv4: delimiter must be included as the first five characters).
•
ipv6:[address]—the IPv6 address of the SIP registrar server (the ipv6: delimiter must be included as the first five characters and the address itself must include opening and closing square brackets).
[:port]
(Optional) The SIP port number (the colon delimiter is required).
auth-realm
(Optional) Specifies the realm for preloaded authorization.
realm
The realm name.
expires seconds
(Optional) Specifies the default registration time, in seconds. Range is 60 to 65535. Default is 3600.
random-contact
(Optional) Specifies the Random String Contact header used to identify the registration session.
refresh-ratio ratio-percentage
(Optional) Specifies the registration refresh ratio, in percentage. Range is 1 to 100. Default is 80.
scheme {sip | sips}
(Optional) Specifies the URL scheme. The options are SIP (sip) or secure SIP (sips), depending on your software installation. The default is sip.
tcp
(Optional) Specifies TCP. If not specified, the default is User Datagram Protocol UDP.
Cisco IOS Voice Command Reference
VR-1590
•
Cisco IOS Voice Commands: R registrar
type
(Optional) The registration type. Note
secondary
The type argument cannot be used with the dhcp option.
(Optional) Specifies a secondary SIP registrar for redundancy if the primary registrar fails. This option is not valid if specifying DHCP or if configuring multiple registrars. Note
You cannot configure any other optional settings once you enter the secondary keyword—specify all other settings first.
Command Default
Registration is disabled.
Command Modes
SIP UA configuration (config-sip-ua)
Command History
Release
Modification
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
12.4(6)T
This command was modified. The tls keyword and the scheme keyword with the string argument were added.
12.4(22)T
This command was modified. Support for IPv6 addresses was added.
12.4(22)YB
This command was modified. The dhcp, random-contact and refresh-ratio keywords were added. Additionally, the aor-domain keyword and the tls option for the tcp keyword were removed.
15.0(1)M
This command was integrated into Cisco IOS Release 15.0(1)M.
15.0(1)XA
This command was modified. The registrar-index argument for support of multiple registrars on SIP trunks was added.
15.1(1)T
This command was integrated into Cisco IOS Release 15.1(1)T.
15.1(2)T
This command was modified. The auth-realm keyword was added.
Usage Guidelines
Use the registrar dhcp or registrar registrar-server-address command to enable the gateway to register E.164 telephone numbers with primary and secondary external SIP registrars. In Cisco IOS Release 15.0(1)XA and later releases, endpoints on Cisco IOS SIP time-division multiplexing (TDM) gateways, Cisco Unified Border Elements (Cisco UBEs), and Cisco Unified Communications Manager Express (Cisco Unified CME) can be registered to multiple registrars using the registrar registrar-index command. By default, Cisco IOS SIP gateways do not generate SIP register messages. Note
When entering an IPv6 address, you must include square brackets around the address value.
Cisco IOS Voice Command Reference
VR-1591
Cisco IOS Voice Commands: R registrar
Examples
The following example shows how to configure registration with a primary and secondary registrar: Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# retry invite 3 Router(config-sip-ua)# retry register 3 Router(config-sip-ua)# timers register 150 Router(config-sip-ua)# registrar ipv4:209.165.201.1 expires 14400 secondary
The following example shows how to configure a device to register with the SIP server address received from the DHCP server. The dhcp keyword is available only for configuration by the primary registrar and cannot be used if configuring multiple registrars. Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# registrar dhcp expires 14400
The following example shows how to configure a primary registrar using an IP address with TCP: Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# retry invite 3 Router(config-sip-ua)# retry register 3 Router(config-sip-ua)# timers register 150 Router(config-sip-ua)# registrar ipv4:209.165.201.3 tcp
The following example shows how to configure a URL scheme with SIP security: Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# retry invite 3 Router(config-sip-ua)# retry register 3 Router(config-sip-ua)# timers register 150 Router(config-sip-ua)# registrar ipv4:209.165.201.7 scheme sips
The following example shows how to configure a secondary registrar using an IPv6 address: Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# registrar ipv6:[3FFE:501:FFFF:5:20F:F7FF:FE0B:2972] expires 14400 secondary
The following example shows how to configure all POTS endpoints to two registrars using DNS addresses: Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# registrar 1 dns:example1.com expires 180 Router(config-sip-ua)# registrar 2 dns:example2.com expires 360
The following example shows how to configure the realm for preloaded authorization using the registrar server address: Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# registrar 2 192.168.140.3:8080 auth-realm example.com expires 180
Cisco IOS Voice Command Reference
VR-1592
Cisco IOS Voice Commands: R registrar
Related Commands
Command
Description
authentication (dial peer)
Enables SIP digest authentication on an individual dial peer.
authentication (SIP UA)
Enables SIP digest authentication.
credentials (SIP UA)
Configures a Cisco UBE to send a SIP registration message when in the UP state.
localhost
Configures global settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages.
retry register
Sets the total number of SIP register messages to send.
show sip-ua register status Displays the status of E.164 numbers that a SIP gateway has registered with an external primary or secondary SIP registrar. timers register
Sets how long the SIP UA waits before sending register requests.
voice-class sip localhost
Configures settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages on an individual dial peer, overriding the global setting.
Cisco IOS Voice Command Reference
VR-1593
Cisco IOS Voice Commands: R registrar server
registrar server To enable the local Session Initiation Protocol (SIP) registrar, use the registrar server command in service SIP configuration mode. To disable the configuration, use the no form of this command. registrar server [expires [max value] [min value]] no registrar server
Syntax Description
expires
(Optional) Configures the registration expiry time.
max value
(Optional) Configures the maximum registration expiry time, in seconds. The range is from 120 to 86400. The default is 3600.
min value
(Optional) Configures the minimum registration expiry time, in seconds. The range is from 60 to 3600. The default is 60.
Command Default
The local SIP registrar is disabled.
Command Modes
Service SIP configuration (conf-serv-sip)
Command History
Release
Modification
15.1(3)T
This command was introduced.
Usage Guidelines
You must enable the local SIP registrar by using the registrar server command before configuring the SIP registration on Cisco Unified Border Element (UBE).
Examples
The following example shows how to enable the local SIP registrar and set the maximum and minimum expiry values to 4000 and 100 seconds respectively: Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# registrar server expires max 4000 min 100
Related Commands
Command
Description
registration passthrough
Configures SIP registration pass-through options at the global level.
voice-class sip registration passthrough
Configures SIP registration pass-through options on a dial peer.
Cisco IOS Voice Command Reference
VR-1594
Cisco IOS Voice Commands: R registration retries
registration retries To set the number of times that Skinny Client Control Protocol (SCCP) tries to register with a Cisco Unified CallManager, use the registration retries command in SCCP Cisco CallManager configuration mode. To reset this number to the default value, use the no form of this command. registration retries retry-attempts no registration retries
Syntax Description
retry-attempts
Command Default
3 registration attempts
Command Modes
SCCP Cisco CallManager configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
Usage Guidelines
Note
Examples
Number of registration attempts. Range is 1 to 32. Default is 3.
Use this command to control the number of registration retries before SCCP confirms that it cannot register with the Cisco Unified CallManager. When SCCP confirms that it cannot register to the current Cisco Unified CallManager (if the number of registration requests sent without an Ack reaches the registration retries value), SCCP tries to register with the next Cisco Unified CallManager.
The optimum setting for this command depends on the platform and your individual network characteristics. Adjust the registration retry attempts to meet your needs.
The following example sets the number of registration retries to 15: Router(config-sccp-ccm)# registration retries 15
Related Commands
Command
Description
ccm group
Creates a Cisco Unified CallManger group and enters SCCP Cisco CallManager configuration mode.
registration timeout
Sets the length of time between registration messages sent from SCCP to the Cisco CallManager.
Cisco IOS Voice Command Reference
VR-1595
Cisco IOS Voice Commands: R registration timeout
registration timeout To set the length of time between registration messages sent from Skinny Client Control Protocol (SCCP) to the Cisco Unified CallManager, use the registration timeout command in SCCP Cisco CallManager configuration mode. To reset the length of time to the default value, use the no form of this command. registration timeout seconds no registration timeout
Syntax Description
seconds
Command Default
3 seconds
Command Modes
SCCP Cisco CallManager configuration
Command History
Release
Modification
12.3(8)T
This command was introduced.
Usage Guidelines
Note
Examples
Time, in seconds, between registration messages. Range is 1 to 180. Default is 3.
Whenever SCCP sends the registration message to the Cisco Unified CallManager, it initiates this timer. Once the timeout occurs, it sends the next registration message unless the number of messages without an Ack reaches the number set by the registration retries command. Use this command to set the Cisco Unified CallManager registration timeout parameter value.
The optimum setting for this command depends on the platform and your individual network characteristics. Adjust the registration timeout value to meet your needs.
The following example sets the length of time between registration messages sent from SCCP to the Cisco Unified CallManager to 12 seconds: Router(config-sccp-ccm)# registration timeout 12
Related Commands
Command
Description
ccm group
Creates a Cisco CallManger group and enters SCCP Cisco CallManager configuration mode.
registration retries
Sets the number of times that SCCP tries to register with the Cisco Unified CallManager.
Cisco IOS Voice Command Reference
VR-1596
Cisco IOS Voice Commands: R registration passthrough
registration passthrough To configure the Session Initiation Protocol (SIP) registration pass-through options, use the registration passthrough command in service SIP configuration mode. To disable the configuration, use the no form of this command. registration passthrough [static] [rate-limit [expires value] [fail-count value]] [registrar-index [index]] no registration passthrough
Syntax Description
static
(Optional) Configures Cisco Unified Border Element (UBE) to use static registrar details for SIP registration. Cisco UBE works in point-to-point mode when the static keyword is used.
(Optional) Sets the expiry value for rate limiting, in seconds. The range is from 60 to 65535. The default value is 3600.
fail-count value
(Optional) Sets the fail count value for rate limiting. The range is from 2 to 20. The default value is 0.
registrar-index
(Optional) Configures the registrar index that is to be used for registration pass-through.
index
(Optional) Registration index value. The range is from 1 to 6.
Command Default
SIP registration pass-through options are not configured.
Command Modes
Service SIP configuration (conf-serv-sip)
Command History
Release
Modification
15.1(3)T
This command was introduced.
Usage Guidelines
You can use the registration passthrough command to configure the following SIP pass-through functionalities: •
Back-to-back registration facility to register phones for call routing.
•
Options to configure the rate-limiting values, such as the expiry time, fail-count, and a list of registrars to be used for registration.
Cisco IOS Voice Command Reference
VR-1597
Cisco IOS Voice Commands: R registration passthrough
Examples
The following example shows how to set the registrar index as 2 for the SIP registration pass-through rate-limiting: Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# registration passthrough static rate-limit registrar-index 2
Sets the SIP registration pass-through rate limiting options on a dial peer.
Cisco IOS Voice Command Reference
VR-1598
Cisco IOS Voice Commands: R rel1xx
rel1xx To enable all Session Initiation Protocol (SIP) provisional responses (other than 100 Trying) to be sent reliably to the remote SIP endpoint, use the rel1xx command in SIP configuration mode. To reset to the default, use the no form of this command. rel1xx {supported value | require value | disable} no rel1xx
Syntax Description
supported value
Supports reliable provisional responses. The value argument may have any value, as long as both the user-agent client (UAC) and user-agent server (UAS) configure it the same. This keyword, with value of 100rel, is the default.
require value
Requires reliable provisional responses. The value argument may have any value, as long as both the UAC and UAS configure it the same.
disable
Disables the use of reliable provisional responses.
Command Default
supported with the 100rel value
Command Modes
SIP configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(11)T
This command was supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
Usage Guidelines
The use of resource reservation with SIP requires that the reliable provisional feature for SIP be enabled either at the VoIP dial-peer level or globally on the router. There are two ways to configure reliable provisional responses: •
Dial peer configuration mode. You can configure reliable provisional responses for the specific dial peer only by using the voice-class sip rel1xx command.
•
SIP configuration mode. You can configure reliable provisional responses globally by using the rel1xx command.
Cisco IOS Voice Command Reference
VR-1599
Cisco IOS Voice Commands: R rel1xx
The voice-class sip rel1xx command in dial peer configuration mode takes precedence over the rel1xx command in global configuration mode with one exception: If the voice-class sip rel1xx command is used with the system keyword, the gateway uses what was configured under the rel1xx command in global configuration mode. Enter SIP configuration mode from voice-service VoIP configuration mode as shown in the following example.
Examples
The following example shows use of the rel1xx command with the value 100rel: Router(config)# voice service voip Router(config-voi-srv)# sip Router(conf-serv-sip)# rel1xx supported 100rel
Related Commands
Command
Description
sip
Enters SIP configuration mode from voice-service VoIP configuration mode.
voice-class sip rel1xx
Provides provisional responses for calls on a dial peer basis.
Cisco IOS Voice Command Reference
VR-1600
Cisco IOS Voice Commands: R remote-party-id
remote-party-id To enable translation of the SIP header Remote-Party-ID, use the remote-party-id command in SIP UA configuration mode. To disable Remote-Party-ID translation, use the no form of this command. remote-party-id no remote-party-id
Syntax Description
This command has no arguments or keywords.
Command Default
Remote-Party-ID translation is enabled
Command Modes
SIP UA configuration
Command History
Release
Modification
12.2(13)T
This command was introduced.
Usage Guidelines
Examples
When the remote-party-id command is enabled, one of the following calling information treatments occurs: •
If a Remote-Party-ID header is present in the incoming INVITE message, the calling name and number extracted from the Remote-Party-ID header are sent as the calling name and number in the outgoing Setup message. This is the default behavior. Use the remote-party-id command to enable this option.
•
When no Remote-Party-ID header is available, no translation occurs so the calling name and number are extracted from the From header and are sent as the calling name and number in the outgoing Setup message. This treatment also occurs when the feature is disabled.
The following example shows the Remote-Party-ID translation being enabled: Router(config-sip-ua)# remote-party-id
Related Commands
Command
Description
debug ccsip events
Enables tracing of SIP SPI events.
debug ccsip messages
Enables SIP SPI message tracing.
debug isdn q931
Displays call setup and teardown of ISDN connections.
debug voice ccapi in out
Enables tracing the execution path through the call control API.
Cisco IOS Voice Command Reference
VR-1601
Cisco IOS Voice Commands: R req-qos
req-qos To specify the desired quality of service to be used in reaching a specified dial peer, use the req-qos command in dial peer configuration mode. To restore the default value for this command, use the no form of this command. req-qos {best-effort | controlled-load | guaranteed-delay} [{audio bandwidth | video bandwidth} default | max bandwidth-value] no req-qos
Syntax Description
best-effort
Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation.
controlled-load
Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to assure that preferential service is received even when the bandwidth is overloaded.
guaranteed-delay
Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded.
audio bandwidth
(Optional) Specifies amount of bandwidth to be requested for audio streams.
default
Sets the default bandwidth to be requested for audio or video streams. •
Audio streams—Range is 1 to 64 kbps; default value is 64 kbps.
•
Video streams—Range is 1 to 5000 kbps; default value is no maximum
max bandwidth-value
Sets the maximum bandwidth to be requested for audio streams. Range is 1 to 64 kbps; default value is no maximum.
video bandwidth
(Optional) Specifies the amount of bandwidth to be requested for video streams.
default bandwidth-value
Sets the default bandwidth to be requested for video streams. Range is 1 to 5000 kbps; default value is 384 kbps.
max bandwidth-value
(Optional) Sets the maximum bandwidth to be requested for video streams. .
Defaults
best-effort
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series routers.
12.3(4)T
Keywords added to support audio and video streams.
Cisco IOS Voice Command Reference
VR-1602
Cisco IOS Voice Commands: R req-qos
Usage Guidelines
Use the req-qos command to request a specific quality of service to be used in reaching a dial peer. Like acc-qos, when you issue this command, the Cisco IOS software reserves a certain amount of bandwidth so that the selected quality of service can be provided. Cisco IOS software uses Resource Reservation Protocol (RSVP) to request quality of service guarantees from the network. This command is applicable only to VoIP dial peers.
Examples
The following example configures guaranteed-delay as the requested quality of service to a dial peer: dial-peer voice 10 voip req-qos guaranteed-delay
The following example configures guaranteed-delay and requests a default bandwidth level of 768 kbps for video streams: dial-peer voice 20 voip req-qos guaranteed-delay video bandwidth default 768
Related Commands
Command
Description
acc-qos
Defines the acceptable QoS for any inbound and outbound call on a VoIP dial peer.
Cisco IOS Voice Command Reference
VR-1603
Cisco IOS Voice Commands: R request
request To use SIP profiles to add, copy, modify, or remove Session Initiation Protocol (SIP) or Session Description Protocol (SDP) header value in a SIP request message, use the request command in voice class configuration mode. To disable the configuration, use the no form of this command. request method {sdp-header | sip-header} header-name {add | copy | modify | remove} string no request method {sdp-header | sip-header} header-name {add | copy | modify | remove} string
Syntax Description
method
Type of message to be added, modified, or removed. It can be one of the following values: •
ack—SIP acknowledgment message.
•
any—Any SIP message.
•
bye—SIP BYE message.
•
cancel—SIP CANCEL message.
•
comet—SIP COMET message.
•
info—SIP INFO message.
•
invite—The first SIP INVITE message.
•
notify—SIP NOTIFY message.
•
options—SIP OPTIONS message.
•
prack—SIP PRACK message.
•
publish—SIP PUBLISH message.
•
refer—SIP REFER message.
•
register—SIP REGISTER message.
•
reinvite—SIP REINVITE message.
•
subscribe—SIP SUBSCRIBE message.
•
update—SIP UPDATE message.
sdp-header
Specifies an SDP header.
sip-header
Specifies a SIP header.
header-name
SDP or SIP header name.
add
Adds a header.
copy
Copies a header.
modify
Modifies a header.
remove
Removes a header.
string
String to be added, copied, modified, or removed as a header. Note
Cisco IOS Voice Command Reference
VR-1604
If you use the copy keyword, you must provide a matching pattern followed by the variable name for the string argument.
Cisco IOS Voice Commands: R request
Command Default
SIP profiles are not modified to add, copy, modify, or remove SIP or SDP header values.
Command Modes
Voice class configuration (config-class)
Command History
Release
Modification
15.1(3)T
This command was introduced.
Usage Guidelines
If there are interoperability issues with Cisco UBE, the Cisco UBE will not work with the default SIP signaling. Hence, you must modify the SIP profiles to add, copy, modify, or remove SIP or SDP header values, and therefore enable Cisco UBE to work with SIP signaling. Use the request command to modify SIP profiles for a request message. You can add, copy, modify, or remove SIP or SDP header values in an outgoing SIP request message.
Examples
The following example shows how to copy a SIP header value in a SIP request message: Router(config)# voice class sip-profiles 10 Router(config-class)# request invite sip-header contact copy "(.*)" u01
Related Commands
Command
Description
response
Modifies a SIP profile to add, copy, modify, or remove a SIP or SDP header value from a SIP response message.
Cisco IOS Voice Command Reference
VR-1605
Cisco IOS Voice Commands: R request peer-header
request peer-header To use SIP profiles to copy a peer header from an outgoing Session Initiation Protocol (SIP) request message, use the request peer-header command in voice class configuration mode. To disable the configuration, use the no form of this command. request method peer-header sip {sip-req-uri | header-name} copy pattern variable no request method peer-header sip {sip-req-uri | header-name} copy pattern variable
Syntax Description
method
Type of message to be copied. You can specify any of the following values:
Command Default
ack—SIP acknowledgment message.
•
any—SIP message.
•
bye—SIP BYE message.
•
cancel—SIP CANCEL message.
•
comet—SIP COMET message.
•
info—SIP INFO message.
•
invite—First SIP INVITE message.
•
notify—Specifies SIP NOTIFY message.
•
options—SIP OPTIONS message.
•
prack—SIP PRACK message.
•
publish—SIP PUBLISH message.
•
refer—SIP REFER message.
•
register—SIP REGISTER message.
•
reinvite—SIP REINVITE message.
•
subscribe—SIP SUBSCRIBE message.
•
update—SIP UPDATE message.
sip
Specifies that the SIP header must be copied from the peer call leg.
sip-req-uri
Specifies the SIP request Uniform Resource Identifier (URI) to be copied from the peer call leg.
header-name
Header name from which the values must be copied.
copy
Copies a header.
pattern
Match pattern.
variable
Variable to which the pattern value must be copied. The range is from u01 to u99.
No SIP profiles are modified to copy a peer header in an outgoing SIP request message.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R request peer-header
Command Modes
Voice class configuration (config-class)
Command History
Release
Modification
15.1(3)T
This command was introduced.
Usage Guidelines
If there are interoperability issues with Cisco UBE, then the Cisco UBE will not be able to work with the default SIP signaling. Hence, you must modify the SIP profiles to add, copy, modify, or remove SIP or SDP header values, and therefore enable Cisco UBE to work with SIP signaling. Configure the request peer-header command to use SIP profiles to copy a peer header from an outgoing SIP request message.
Examples
The following example shows how to copy a peer header in an outgoing SIP request message: Router(config)# voice class sip-profiles 10 Router(config-class)# request invite peer-header sip contact copy "(.*)" u01
Related Commands
Command
Description
response peer-header
Uses SIP profiles to copy a peer header from an outgoing SIP response message.
Cisco IOS Voice Command Reference
VR-1607
Cisco IOS Voice Commands: R request (XML transport)
request (XML transport) To set the XML transport mode request handling parameters, use the request command in XML transport configuration mode. To disable the XML transport request parameter setting, use the no form of this command request {outstanding number | timeout seconds} no request
Syntax Description
outstanding
Maximum number of outstanding requests.
number
The valid range for the number of outstanding requests is from 1 to 10. The default is 1.
timeout
Response timeout at the transport level.
seconds
Specifies the number of seconds a request is active before it times out. Valid rangeis from 0 to 60 seconds. The default value is 0 (no timeout).
Command Default
The default for outstanding is 1 and the default for timeout is 0 (no timeout).
Command Modes
XML transport configuration
Command History
Release
Modification
12.4(6)T
This command was introduced.
Usage Guidelines
Use this command to set the request timeout. A value of 0 seconds specifies no timeout. This timeout applies to the request being processed and not outstanding requests as described below. The specified timeout limits the amount of time between the request being dequeued by the application and the completion of the processing of that request. Use this command to specify the number of outstanding requests allowed per application for the specified transport mode. The outstanding requests are those requests that are queued at the application for processing but have not yet been processed.
Examples
The following example shows how to enter XML transport configuration mode, set the XML transport request timeout to 10 seconds, and exit XML transport configuration mode: Router(config)# ixi transport http Router(conf-xml-trans)# request timeout 10
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R request (XML transport)
Related Commands
Command
Description
ixi transport http
Enters XML transport configuration mode.
ixi application mib
Enters XML application configuration mode.
response size (XML transport) Set the XML transport fragment size.
Cisco IOS Voice Command Reference
VR-1609
Cisco IOS Voice Commands: R reset
reset To reset a set of digital signal processors (DSPs), use the reset command in global configuration mode. reset number
Syntax Description
number
Command Default
No default behavior or values.
Command Modes
Global configuration
Command History
12.0(5)XE
This command was introduced on the Cisco 7200 series.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Examples
Number of DSPs to be reset. Range is from 0 to 30.
The following example displays the reset command configuration for DSP 1: reset 1 01:24:54:%DSPRM-5-UPDOWN: DSP 1 in slot 1, changed state to up
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Cisco IOS Voice Commands: R reset timer expires
reset timer expires To globally configure Cisco Unified Communications Manager Express (Cisco Unified CME), a Cisco IOS voice gateway, or a Cisco Unified Border Element (Cisco UBE) to reset the expires timer upon receipt of a Session Initiation Protocol (SIP) 183 Session In Progress message, use the reset timer expires command in voice service SIP configuration mode. To globally disable resetting of the expires timer upon receipt of SIP 183 messages, use the no form of this command. reset timer expires 183 no reset timer expires 183
Syntax Description
183
Command Default
The expires timer is not reset after receipt of SIP 183 Session In Progress messages and a session or call that is not connected within the default expiration time (three minutes) is dropped.
Command Modes
Voice service SIP configuration (conf-serv-sip)
Command History
Release
Modification
15.0(1)XA
This command was introduced.
15.1(1)T
This command was integrated into Cisco IOS Release 15.1(1)T.
Usage Guidelines
Specifies resetting of the expires timer upon receipt of SIP 183 Session In Progress messages.
In some scenarios, early media cut-through calls (such as emergency calls) rely on SIP 183 with session description protocol (SDP) Session In Progress messages to keep the session or call alive until receiving a FINAL SIP 200 OK message, which indicates that the call is connected. In these scenarios, the call can time out and be dropped if it does not get connected within the default expiration time (three minutes). Note
The expires timer default is three minutes. However, you can configure the expiration time to a maximum of 30 minutes using the timers expires command in SIP user agent (UA) configuration mode.
To prevent early media cut-through calls from being dropped because they reach the expires timer limit, use the reset timer expires command in voice service SIP configuration mode to globally enable all dial peers on Cisco Unified CME, Cisco IOS voice gateways, or Cisco UBEs to reset the expires timer upon receipt of any SIP 183 message. To configure the reset timer expiration setting for an individual dial peer, use the voice-class sip reset timer expires command in dial peer voice configuration mode. To disable the expires timer reset on receipt of SIP 183 messages function, use the no reset timer expires command in voice service SIP configuration mode.
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Cisco IOS Voice Commands: R reset timer expires
Examples
The following example shows how to globally configure all dial peers on Cisco Unified CME, a Cisco IOS voice gateway, or a Cisco UBE to reset the expires timer each time a SIP 183 message is received: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# reset timer expires 183
Related Commands
Command
Description
timers expires
Specifies how long a SIP INVITE request remains valid before it times out if no appropriate response is received for keeping the session alive.
voice-class sip reset timer expires
Configures an individual dial peer on Cisco Unified CME, a Cisco IOS voice gateway, or a Cisco UBE to reset the expires timer upon receipt of a SIP 183 message.
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Cisco IOS Voice Commands: R resource (voice)
resource (voice) To configure parameters for monitoring resources, use the resource command in voice-class configuration mode. To disable the configuration for monitoring resources, use the no form of this command. resource {cpu {1-min-avg | 5-sec-avg} | ds0 | dsp | mem {io-mem | proc-mem | total-mem}} [threshold high threshold-value low threshold-value] no resource {cpu | ds0 | dsp | mem}
Syntax Description
cpu
Reports the CPU utilization information.
1-min-avg
Collects the CPU data for an average of one minute.
5-sec-avg
Collects the CPU data for an average of five seconds.
ds0
Reports utilization information for the DS0 port.
dsp
Reports utilization information for the digital signal processor (DSP) channel.
mem
Reports the memory utilization information.
io-mem
Reports the input/output memory utilization information.
proc-mem
Reports the process memory utilization information.
total-mem
Reports the complete memory utilization information.
threshold
Configures the high and low threshold values for the critical resources.
high
(Optional) Configures the resource high watermark value.
low
(Optional) Configures the resource low watermark value.
threshold-value
Threshold value, in percentage.
Command Default
Critical gateway resources are not monitored.
Command Modes
Voice-class configuration mode (config-class)
Command History
Release
Modification
15.1(2)T
This command was introduced.
Usage Guidelines
Use the resource command to configure parameters for critical resources such as CPU, memory, DS0, and DSP to report the utilization status to external entities using the gateway resources for call handling. You can use the voice class resource-group command to enter voice-class configuration mode and configure resource groups. Each resource group has a unique number that identifies a group of resources to be monitored.
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Cisco IOS Voice Commands: R resource (voice)
When you configure the high watermark values for any of the monitoring resources, be sure not to use more resources than available on the gateway. The high and low watermark values for threshold only indicate that the gateway might run out of resources soon. However, the gateway must still be able to trigger threshold-based reporting to the routing/monitoring entity. When you configure the low watermark value for the threshold, be sure not to underutilize the gateway resources.
Examples
The following example shows how to configure CPU to report the utilization information to the external entities: Router> enable Router# configure terminal Router(config)# voice class resource-group 1 Router(config-class)# resource cpu 1-min-avg threshold high 10 low 2
Related Commands
Command
Description
debug rai
Enables debugging for Resource Allocation Indication (RAI).
periodic-report interval
Configures periodic reporting parameters for gateway resource entities.
rai target
Configures the SIP RAI mechanism.
show voice class resource-group
Displays the resource group configuration information for a specific resource group or all resource groups.
voice class resource-group
Enters voice-class configuration mode and assigns an identification tag number for a resource group.
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Cisco IOS Voice Commands: R resource threshold
resource threshold To configure a gateway to report H.323 resource availability to its gatekeeper, use the resource threshold command in gateway configuration mode. To disable gateway resource-level reporting, use the no form of this command. resource threshold [all] [high percentage-value] [low percentage-value] no resource threshold
Syntax Description
all
(Optional) High- and low-parameter settings are applied to all monitored H.323 resources. This is the default condition.
high percentage-value
(Optional) Resource utilization level that triggers a Resource Availability Indicator (RAI) message that indicates that H.323 resource use is high. Enter a number between 1 and 100 that represents the high-resource utilization percentage. A value of 100 specifies high-resource usage when any H.323 resource is unavailable. Default is 90 percent.
low percentage-value
(Optional) Resource utilization level that triggers an RAI message that indicates H.323 resource usage has dropped below the high-usage level. Enter a number between 1 and 100 that represents the acceptable resource utilization percentage. After the gateway sends a high-utilization message, it waits to send the resource recovery message until the resource use drops below the value defined by the low parameter. Default is 90 percent.
Command Default
Reports low resources when 90 percent of resources are in use and reports resource availability when resource use drops below 90 percent.
Command Modes
Gateway configuration
Command History
Release
Modification
12.0(5)T
This command was introduced on the Cisco AS5300.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T. This command is supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release
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Cisco IOS Voice Commands: R resource threshold
Usage Guidelines
This command defines the resource load levels that trigger RAI messages. To view the monitored resources, enter the show gateway command. The monitored H.323 resources include digital signal processor (DSP) channels and DS0s. Use the show call resource voice stats command to see the total amount of resources available for H.323 calls.
Note
The DS0 resources that are monitored for H.323 calls are limited to the ones that are associated with a voice POTS dial peer. See the dial peer configuration commands for details on how to associate a dial peer with a PRI or channel-associated signaling (CAS) group. When any monitored H.323 resources exceed the threshold level defined by the high parameter, the gateway sends an RAI message to the gatekeeper with the AlmostOutOfResources field flagged. This message reports high resource usage. When all gateway H.323 resources drop below the level defined by the low parameter, the gateway sends the RAI message to the gatekeeper with the AlmostOutOfResources field cleared. When a gatekeeper can choose between multiple gateways for call completion, the gatekeeper uses internal priority settings and gateway resource statistics to determine which gateway to use. When all other factors are equal, a gateway that has available resources is chosen over a gateway that has reported limited resources.
Examples
The following example defines the H.323 resource limits for a gateway. gateway1(config-gateway)# resource threshold high 70 low 60
Related Commands
Command
Description
show call resource voice stats
Displays resource statistics for an H.323 gateway.
show call resource voice threshold
Displays the threshold configuration settings and status for an H.323 gateway.
show gateway
Displays the current gateway status.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R resource-pool (mediacard)
resource-pool (mediacard) To create a Digital Signal Processor (DSP) resource pool on ad-hoc conferencing and transcoding port adapters, use the resource-pool command in mediacard configuration mode. To remove the DSP resource pool and release the associated DSP resources, use the no form of this command. resource-pool identifier dsps number no resource-pool identifier dsps number
Syntax Description
identifier
Identifies the DSP resource to be configured. Valid values consist of alphanumeric characters, plus “_” and “-”.
dsps
Digital signal processor.
number
Specifies the number of DSPs to be allocated for the specified resource pool. Valid values are from 1 to 4.
Command Default
No default behavior or values
Command Modes
Mediacard configuration
Command History
Release
Modification
12.3(8)XY
This command was introduced on the Communication Media Module.
12.3(14)T
This command was integrated into Cisco IOS Release 12.3(14)T.
12.4(3)
This command was integrated into Cisco IOS Release 12.4(3).
Usage Guidelines
The DSP resource pool identifier should be unique across the same Communication Media Module (CMM). Removing a resource pool may cause the profile using that resource pool to be disabled if it is the last resource pool in the profile.
Examples
The following example shows how to create a DSP resource pool: resource-pool headquarters_location1 dsps 2
Related Commands
Command
Description
debug mediacard
Displays debugging information for DSPRM.
show mediacard
Displays information about the selected media card.
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Cisco IOS Voice Commands: R response
response To use SIP profiles to add, copy, modify, or remove Session Initiation Protocol (SIP) or Session Description Protocol (SDP) header value in a SIP response message, use the response command in voice class configuration mode. To disable the configuration, use the no form of this command. response option {sdp-header | sip-header} header-name {add | copy | modify | remove} string no response option {sdp-header | sip-header} header-name {add | copy | modify | remove} string
Syntax Description
option
Response code to be added, copied, modified, or removed. You can specify one of the following values: •
code—Response code value. It can be one of the following values: – 100 – 180 to 183 – 200 – 102 – 300 to 302 – 305 – 380 – 400 to 423 – 480 to 489 – 491 – 493 – 500 to 505 – 515 – 580 – 600 – 603 – 604 – 606
•
sdp-header
Specifies SDP header.
sip-header
Specifies SIP header.
header-name
SDP or SIP header name.
add
Adds a header.
copy
Copies a header.
modify
Modifies a header.
remove
Removes a header.
string
String to be added as a header.
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any—Adds, copies, modifies, or removes any response message.
Cisco IOS Voice Commands: R response
Command Default
No SIP profile is modified to add, copy, modify, or remove a SIP header value.
Command Modes
Voice class configuration (config-class)
Command History
Release
Modification
15.1(3)T
This command was introduced.
Usage Guidelines
If there are interoperability issues with Cisco UBE, the Cisco UBE will not be able to work with the default SIP signaling. Hence, you must modify the SIP profiles to add, copy, modify, or remove SIP header values, to enable Cisco UBE to work with SIP signaling. Use the response command to modify SIP profiles for a response message. You can add, copy, modify, or remove SIP or SDP header values in an outgoing SIP response message.
Examples
The following example shows how to copy a SIP header value in a SIP response message: Router(config)# voice class sip-profiles 10 Router(config-class)# response 409 sip-header to copy string1
Related Commands
Command
Description
request
Modifies a SIP profile to add, copy, modify, or remove a SIP or SDP header value from an outgoing SIP request message.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R response (XML application)
response (XML application) To set XML application response parameters, use the response command in XML application configuration mode. To disable response parameter settings, use the no form of this command. response {formatted | timeout {-1 | seconds}} no response {formatted | timeout {-1 | seconds}}
Syntax Description
formatted
Response parameters in formatted human readable XML.
timeout
Application specified response timeout.
-1
Enter -1 to indicate no application specified timeout. This is the default timeout setting.
seconds
Number of seconds a response is active before it times out. Valid range includes 0 to 60 seconds.
Command Default
The default for the timeout keyword is -1 indicating not application specified timeout.
Command Modes
XML application configuration
Command History
Release
Modification
12.4(6)T
This command was introduced.
Usage Guidelines
The response timeout specified in this command, if other than -1 which is the default, overwrites the timeout value specified in the request (XML transport) command that sets the timeout at the transport level. The same http transport layer could have multiple applications active at the same time. You can set the timeout for each application individually or have all of the applications to use the same timeout value set at transport layer using the request (XML transport) command in XML transport configuration mode.
Examples
The following example shows how to enter XML application configuration mode, set XML response parameters in formatted human readable XML, and exit XML application configuration mode: Router(config)# ixi application mib Router(conf-xml-app)# response formatted
Related Commands
Command
Description
ixi application mib
Enters XML application configuration mode.
request (XML transport)
Set the XML transport mode request handling parameters.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R response peer-header
response peer-header To use SIP profiles to copy a peer header value in a SIP response message, use the response peer-header command in voice class configuration mode. To disable the configuration, use the no form of this command. response {code | any} peer-header sip {sip-req-uri | header-name} copy pattern variable no response option peer-header sip {sip-req-uri | header-name} copy pattern variable
Syntax Description
code
Response code to be copied. You can specify one of the following values: – 100 – 180 to 183 – 200 – 102 – 300 to 302 – 305 – 380 – 400 to 423 – 480 to 489 – 491 – 493 – 500 to 505 – 515 – 580 – 600 – 603 – 604 – 606 •
Command Default
any—Adds, copies, modifies, or removes any response message.
any
Adds, copies, modifies, or removes any response message.
sip
Specifies that the SIP header must be copied from the peer call leg.
sip-req-uri
Specifies the SIP request Uniform Resource Identifier (URI) to be copied from the peer call leg.
header-name
Header name from which the peer header values must be copied.
copy
Copies a header.
pattern
Match pattern.
variable
The destination variable name. The range is from u01 to u99.
No SIP profile is modified.
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Cisco IOS Voice Commands: R response peer-header
Command Modes
Voice class configuration (config-class)
Command History
Release
Modification
15.1(3)T
This command was introduced.
Usage Guidelines
If there are interoperability issues with Cisco UBE, the Cisco UBE will not be able to work with the default SIP signaling. Hence, you must modify the SIP profiles to add, copy, modify, or remove SIP or SDP header values, to enable Cisco UBE to work with SIP signaling. Use the response peer-header command to copy a peer header value in a SIP response message.
Examples
The following example shows how to copy a peer header value in a SIP response message: Router(config)# voice class sip-profiles 10 Router(config-class)# response 200 peer-header sip contact copy "(.*)" u01
Related Commands
Command
Description
request peer-header
Uses SIP profiles to copy a peer header value in a SIP request message.
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Cisco IOS Voice Commands: R response size (XML transport)
response size (XML transport) To set the response transport fragment size, use the response size command in XML transport configuration mode. To disable the response transport fragment size setting, use the no form of this command. response size kBps no response size
Syntax Description
kBps
Command Modes
XML transport configuration
Command History
Release
Modification
12.4(6)T
This command was introduced.
Size of the fragment in the response buffer in kilobytes. Valid range is 1 to 64 kB. The default is 4 kB.
Usage Guidelines
The fragment size is constrained by the transport type. The CLI help provides input guidelines.
Examples
The following example shows how to enter XML transport configuration mode, set XML transport fragment size to 32 Kbytes, and exit XML transport configuration mode: Router(config)# ixi transport http Router(conf-xml-trans)# response size 32
Related Commands
Command
Description
ixi transport http
Enters XML transport configuration mode.
ixi application mib
Enter XML application configuration mode.
request (XML transport)
Sets XML transport request handling parameters.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R response-timeout
response-timeout To configure the maximum time to wait for a response from a server, use the response-timeout command in settlement configuration mode. To reset to the default, use the no form of this command. response-timeout seconds no response-timeout seconds
Syntax Description
seconds
Command Default
1 second
Command Modes
Settlement configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Response waiting time, in seconds. Default is 1.
Usage Guidelines
If no response is received within the response-timeout time limit, the current connection ends, and the router attempts to contact the next service point.
Examples
The following example sets response timeout to 1 second. settlement 0 response-timeout 1
Related Commands
Command
Description
connection-timeout
Configures the time for which a connection is maintained after completion of a communication exchange.
customer-id
Identifies a carrier or ISP with a settlement provider.
device-id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
max-connection
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
retry-delay
Sets the time between attempts to connect with the settlement provider.
retry-limit
Sets the maximum number of attempts to connect to the provider.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R response-timeout
Command
Description
session-timeout
Sets the interval for closing the connection when there is no input or output traffic.
settlement
Enters settlement mode and specifies the attributes specific to a settlement provider.
show settlement
Displays the configuration for all settlement server transactions.
shutdown/no shutdown
Deactivates the settlement provider/activates the settlement provider.
type
Configures an SAA-RTR operation type.
url
Specifies the Internet service provider address.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R retries (auto-config application)
retries (auto-config application) To set the number of download retry attempts for an auto-configuration application, use the retries command in auto-config application configuration mode. To reset to the default, use the no form of this command. retries number no retries
Syntax Description
number
Command Default
The default value is 2.
Command Modes
Auto-config application configuration
Command History
Release
Modification
12.3(8)XY
This command was introduced on the Communication Media Module.
12.3(14)T
This command was integrated into Cisco IOS Release 12.3(14)T.
Examples
Specifies the download retry attempts. Valid range is 1 to 3.
The following example shows the retries command used to set the number of retries for an auto-configuration application to 3: Router(auto-config-app)# retries 3
Related Commands
Command
Description
auto-config
Enables auto-configuration or enters auto-config application configuration mode for the SCCP application.
show auto-config
Displays the current status of auto-configuration applications.
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Cisco IOS Voice Commands: R retry bye
retry bye To configure the number of times that a BYE request is retransmitted to the other user agent, use the retry bye command in SIP UA configuration mode. To reset to the default, use the no form of this command. retry bye number no retry bye number
Syntax Description
number
Command Default
10 retries
Command Modes
SIP UA configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400 and Cisco AS5850 in this release.
Number of BYE retries. Range is from 1 to 10. The default is 10.
Usage Guidelines
To reset this command to the default value, you can also use the default command.
Examples
The following example sets the number of BYE retries to 5. sip-ua retry bye 5
Related Commands
Command
Description
default
Resets the value of a command to its default.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
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Cisco IOS Voice Commands: R retry bye
Command
Description
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times that the PRACK request is retransmitted to the other user agent.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
sip-ua
Enables the SIP user-agent configuration commands, with which you configure the user agent.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R retry cancel
retry cancel To configure the number of times that a CANCEL request is retransmitted to the other user agent, use the retry cancel command in SIP UA configuration mode. To reset to the default, use the no form of this command. retry cancel number no retry cancel number
Syntax Description
number
Command Default
10 retries
Command Modes
SIP UA configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400 and Cisco AS5850 in this release.
Number of CANCEL retries. Range is from 1 to 10. Default is 10.
Usage Guidelines
To reset this command to the default value, you can also use the default command.
Examples
The following example sets the number of cancel retries to 5. sip-ua retry cancel 5
Related Commands
Command
Description
default
Resets the value of a command to its default.
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R retry cancel
Command
Description
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times that the PRACK request is retransmitted to the other user agent.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
sip-ua
Enables the sip ua configuration commands, with which you configure the user agent.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R retry comet
retry comet To configure the number of times that a COMET request is retransmitted to the other user agent, use the retry comet command in SIP UA configuration mode. To reset to the default, use the no form of this command. retry comet number no retry comet
Syntax Description
number
Command Default
10 retries
Command Modes
SIP UA configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
Usage Guidelines
Number of COMET retries. Range is from 1 to 10. Default is 10.
COMET, or conditions met, indicates if preconditions for a given call or session have been met. This command is applicable only with calls (other than best-effort) that involve quality of service (QoS). Use the default number of 10 retries, when possible. Lower values, such as 1, can lead to an increased chance of the message not being received by the other user agent.
Examples
The following example configures a COMET request to be retransmitted 8 times: Router(config)# sip-ua Router(config-sip-ua)# retry comet 8
Related Commands
Command
Description
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R retry comet
Command
Description
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times that the PRACK request is retransmitted to the other user agent.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
show sip-ua retry
Displays the SIP retry attempts.
show sip-ua statistics
Displays response, traffic, timer, and retry statistics.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R retry interval
retry interval To define the time between border element attempts delivery of unacknowledged call-detail-record (CDR) information, use the retry interval command in Annex G neighbor usage configuration mode. To reset to the default, use the no form of this command. retry interval seconds no retry interval
Syntax Description
seconds
Command Default
900 seconds
Command Modes
Annex G neighbor usage configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
Retry interval between delivery attempts, in seconds. Range is from 1 to 3600 (1 hour). The default is 900.
Usage Guidelines
Use this command to set the interval during which the border element attempts delivery of unacknowledged call-detail-record (CDR) information.
Examples
The following example sets the retry interval to 2700 seconds (45 minutes): Router(config-nxg-neigh-usg)# retry interval 2700
Related Commands
Command
Description
access-policy
Requires that a neighbor be explicitly configured.
inbound ttl
Sets the inbound time-to-live value.
outbound retry-interval
Defines the retry period for attempting to establish the outbound relationship between border elements.
retry window
Defines the total time for which a border element attempts delivery.
service-relationship
Establishes a service relationship between two border elements.
shutdown
Enables or disables the border element.
usage-indication
Enters the mode used to configure optional usage indicators.
Cisco IOS Voice Command Reference
VR-1633
Cisco IOS Voice Commands: R retry invite
retry invite To configure the number of times that a Session Initiation Protocol (SIP) INVITE request is retransmitted to the other user agent, use the retry invite command in SIP UA configuration mode. To reset to the default, use the no form of this command. retry invite number no retry invite number
Syntax Description
number
Command Default
6 retries
Command Modes
SIP UA configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Usage Guidelines
Number of INVITE retries. Range is from 1 to 10. Default is 6.
To reset this command to the default value, you can also use the default command. When configuring SIP using SIP user-agent configuration commands such as the retry invite command, the use of the default values for the commands causes the rotary function to not take effect. The rotary function is when you set up more than one VoIP dial peer for the same destination pattern, and the dial peers are assigned to different targets. Assign different targets so that if the call cannot be set up with the first dial peer (preference one), the next dial peer can be tried. To use the rotary function within SIP, set the retry value for the SIP retry invite command to 4 or less.
Examples
The following example sets the number of invite retries to 5. sip-ua retry invite 5
Cisco IOS Voice Command Reference
VR-1634
Cisco IOS Voice Commands: R retry invite
Related Commands
Command
Description
default
Resets the value of a command to its default.
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times that the PRACK request is retransmitted to the other user agent.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
sip-ua
Enables the UA configuration commands, with which you configure the user agent.
Cisco IOS Voice Command Reference
VR-1635
Cisco IOS Voice Commands: R retry keepalive (SIP)
retry keepalive (SIP) To set the retry count for keepalive retransmission, use the retry keepalive command in SIP UA configuration mode. To restore the retry count to the default value for keepalive retransmission, use the no form of this command. retry keepalive count no retry keepalive count
Syntax Description
count
Command Default
The default value for the retry keepalive retransmission is 6.
Command Modes
SIP UA configuration
Command History
Release
Modification
12.4(6)T
This command was introduced.
Retry keepalive retransmission value in the range from 1 to 10. The default value is 6.
Usage Guidelines
Sets the keepalive retransmissions retry count.
Examples
The following example sets the retry for the keepalive retransmissions to 8: sip-ua retry keepalive 8
Related Commands
Command
Description
busyout monitor keepalive
Selects a voice port or ports to be busied out in cases of a keepalive failure.
keepalive target
Identifies a SIP server that will receive keepalive packets from the SIP gateway.
keepalive trigger
Sets the trigger to the number of Options message requests that must consecutively receive responses from the SIP servers in order to unbusy the voice ports when in the down state.
timers keepalive
Sets the time interval between sending Options message requests when the SIP server is active or down.
Cisco IOS Voice Command Reference
VR-1636
Cisco IOS Voice Commands: R retry notify
retry notify To configure the number of times that the notify message is retransmitted to the user agent that initiated the transfer or Refer request, use the retry notify command in SIP UA configuration mode. To reset to the default, use the no form of this command. retry notify number no retry notify
Syntax Description
number
Command Default
10 retries
Command Modes
SIP UA configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB2
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Cisco IOS XE Release 2.5
This command was integrated into Cisco IOS XE Release 2.5.
Usage Guidelines
Number of notify message retries. Range is from 1 to 10. Default is 10.
A notify message informs the user agent that initiated the transfer or refer request of the outcome of the Session Initiation Protocol (SIP) transaction. Use the default number of 10 when possible. Lower values such as 1 can lead to an increased chance of the message not being received by the other user agent.
Examples
The following example configures a notify message to be retransmitted 10 times: Router(config)# sip-ua Router(config-sip-ua)# retry notify 10
Cisco IOS Voice Command Reference
VR-1637
Cisco IOS Voice Commands: R retry notify
Related Commands
Command
Description
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry invite
Configures the number of times that a Session Initiation Protocol (SIP) INVITE request is retransmitted to the other user agent.
retry prack
Configures the number of times that the PRACK request is retransmitted to the other user agent.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
show sip-ua retry
Displays the SIP retry attempts.
show sip-ua statistics
Displays response, traffic, timer, and retry statistics.
timers notify
Sets the amount of time that the user agent should wait before retransmitting the Notify message.
Cisco IOS Voice Command Reference
VR-1638
Cisco IOS Voice Commands: R retry prack
retry prack To configure the number of times that the PRACK request is retransmitted to the other user agent, use the retry prack command in SIP UA configuration mode. To reset to the default, use the no form of this command. retry prack number no retry prack
Syntax Description
number
Command Default
10 retries
Command Modes
SIP UA configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 platforms is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
Number of PRACK retries. Range is from 1 to 10. Default is 10.
Usage Guidelines
PRACK allows reliable exchanges of Session Initiation Protocol (SIP) provisional responses between SIP endpoints. Use the default number of 10 when possible. Lower values such as 1 can lead to an increased chance of the message not being received by the other user agent.
Examples
The following example configures a PRACK request to be retransmitted 9 times: Router(config)# sip-ua Router(config-sip-ua)# retry prack 9
Related Commands
Command
Description
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
Cisco IOS Voice Command Reference
VR-1639
Cisco IOS Voice Commands: R retry prack
Command
Description
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
show sip-ua retry
Displays the SIP retry attempts.
show sip-ua statistics
Displays response, traffic, timer, and retry statistics.
Cisco IOS Voice Command Reference
VR-1640
Cisco IOS Voice Commands: R retry refer
retry refer To configure the number of times that the Refer request is retransmitted, use the retry refer command in SIP UA configuration mode. To reset to the default, use the no form of this command. retry refer number no retry refer
Syntax Description
number
Command Default
10 retries
Command Modes
SIP UA configuration
Command History
Release
Modification
12.2(11)YT
This command was introduced.
12.2(15)T
This command is supported on the Cisco 1700 series, Cisco 2600 series, Cisco 3600 series, and the Cisco 7200 series routers in this release.
Usage Guidelines
Number of Refer request retries. Range is from 1 to 10. Default is 10.
A Session Initiation Protocol (SIP) Refer request is sent by the originating gateway to the receiving gateway and initiates call forward and call transfer capabilities. When configuring the retry refer command, use the default number of 10 when possible. Lower values such as 1 can lead to an increased chance of the message not being received by the receiving gateway.
Examples
The following example configures a Refer request to be retransmitted 10 times: Router(config)# sip-ua Router(config-sip-ua)# retry refer 10
Related Commands
Command
Description
show sip-ua retry
Displays the SIP retry attempts.
show sip-ua statistics
Displays response, traffic, timer, and retry statistics.
Cisco IOS Voice Command Reference
VR-1641
Cisco IOS Voice Commands: R retry register
retry register To set the total number of Session Initiation Protocol (SIP) register messages that the gateway should send, use the retry register command in SIP user-agent configuration mode. To reset this number to the default, use the no form of this command. retry register retries no retry register
Syntax Description
retries
Command Default
The gateway sends ten retries.
Command Modes
SIP user-agent configuration
Command History
Release
Modification
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
12.4(22)T
Support for IPv6 was added.
Total number of register messages that the gateway should send. The range is from 1 to 10, and the default is 10 retries.
Usage Guidelines
Use the default number of 10 when possible. Lower values such as 1 can lead to an increased chance of the message not being received by the other user agent.
Examples
The following example specifies that the gateway sends nine register messages: sip-ua retry invite 9 retry register 9 timers register 150
Related Commands
Command
Description
registrar
Enables SIP gateways to register E.164 numbers on behalf of analog telephone voice ports (FXS), IP phone virtual voice ports (EFXS), and SCCP phones with an external SIP proxy or SIP registrar.
timers register
Sets how long the SIP user agent waits before sending register requests.
Cisco IOS Voice Command Reference
VR-1642
Cisco IOS Voice Commands: R retry rel1xx
retry rel1xx To configure the number of times that the reliable 1xx response is retransmitted to the other user agent, use the retry rel1xx command in SIP UA configuration mode. To reset to the default, use the no form of this command. retry rel1xx number no retry rel1xx
Syntax Description
number
Command Default
6 retries
Command Modes
SIP UA configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
Number of reliable 1xx retries. Range is from 1 to 10. Default is 6.
Usage Guidelines
Use the default number of 6 when possible. Lower values such as 1 can lead to an increased chance of the message not being received by the other user agent.
Examples
The following example configures the reliable 1xx response to be retransmitted 7 times: Router(config)# sip-ua Router(config-sip-ua)# retry rel1xx 7
Related Commands
Command
Description
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
Cisco IOS Voice Command Reference
VR-1643
Cisco IOS Voice Commands: R retry rel1xx
Command
Description
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times the PRACK request is retransmitted.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
show sip-ua retry
Displays the SIP retry attempts.
show sip-ua statistics
Displays response, traffic, timer, and retry statistics.
Cisco IOS Voice Command Reference
VR-1644
Cisco IOS Voice Commands: R retry response
retry response To configure the number of times that the response message is retransmitted to the other user agent, use the retry response command in SIP UA configuration mode. To reset to the default, use the no form of this command. retry response number no retry response
Syntax Description
number
Command Default
6 retries
Command Modes
SIP UA configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Number of response retries. Range is from 1 to 10. Default is 6.
Usage Guidelines
To reset this command to the default value, you can also use the default command.
Examples
The following example sets the number of response retries to 5. sip-ua retry response 5
Related Commands
Command
Description
default
Resets the value of a command to its default.
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
Cisco IOS Voice Command Reference
VR-1645
Cisco IOS Voice Commands: R retry response
Command
Description
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times the PRACK request is retransmitted.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
sip-ua
Enables the sip-ua configuration commands, with which you configure the user agent.
Cisco IOS Voice Command Reference
VR-1646
Cisco IOS Voice Commands: R retry subscribe
retry subscribe To configure the number of times that a SIP SUBSCRIBE message is retransmitted to the other user agent, use the retry subscribe command in SIP UA configuration mode. To reset to the default, use the no form of this command. retry subscribe number no retry subscribe number
Syntax Description
number
Command Default
10 retries
Command Modes
SIP UA configuration
Command History
Release
Modification
12.3(4)T
This command was introduced.
Number of SUBSCRIBE retries. Range is 1 to 10. Default is 10.
Usage Guidelines
Use the retry timer command to configure retry intervals for this command. The default value for retry timer is 1000 ms, and the range is 10 to 100. Setting the timer to lower values can cause the application to get a failure response more quickly.
Examples
The following example sets the number of subscribe retries to 5: sip-ua retry subscribe 5
Related Commands
Command
Description
retry notify
Configures the number of times that the Notify message is resent to the user agent that initiated the Invite request.
retry timer
Configures the retry interval for resending SIP messages.
show sip-ua retry
Displays SIP user agent retry statistics.
Cisco IOS Voice Command Reference
VR-1647
Cisco IOS Voice Commands: R retry window
retry window To define the total time for which a border element attempts delivery, use the retry window command in Annex G neighbor usage configuration mode. To reset to the default, use the no form of this command. retry window window-value no retry window
Syntax Description
window-value
Command Default
1440 minutes (24 hours)
Command Modes
Annex G neighbor usage configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
Window value, in minutes. Range is from 1 to 65535. Default is 1440 minutes (24 hours).
Usage Guidelines
Use this command to set the total time during which a border element attempts delivery of unacknowledged call-detail-record (CDR) information.
Examples
The following example sets the retry window to 15 minutes: Router(config-nxg-neigh-usg)# retry window 15
Related Commands
Command
Description
access-policy
Requires that a neighbor be explicitly configured.
inbound ttl
Sets the inbound time-to-live value.
outbound retry-interval
Defines the retry period for attempting to establish the outbound relationship between border elements.
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
Cisco IOS Voice Command Reference
VR-1648
Cisco IOS Voice Commands: R retry window
Command
Description
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times that the PRACK request is retransmitted to the other user agent.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
service-relationship
Establishes a service relationship between two border elements.
shutdown
Enables or disables the border element.
usage-indication
Enters the submode used to configure optional usage indicators.
Cisco IOS Voice Command Reference
VR-1649
Cisco IOS Voice Commands: R retry-delay
retry-delay To set the time between attempts to connect with the settlement provider, use the retry-delay command in settlement configuration mode. To reset to the default, use the no form of this command. retry-delay seconds no retry-delay
Syntax Description
seconds
Command Default
2 seconds
Command Modes
Settlement configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Interval, in seconds, between attempts to connect with the settlement provider. Range is from 1 to 600.
Usage Guidelines
After exhausting all service points for the provider, the router is delayed for the specified length of time before resuming connection attempts.
Examples
The following example sets a retry value of 15 seconds: settlement 0 relay-delay 15
Related Commands
Command
Description
connection-timeout
Configures the time for which a connection is maintained after completion of a communication exchange.
customer-id
Identifies a carrier or ISP with a settlement provider.
device-id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
max-connection
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
response-timeout
Configures the maximum time to wait for a response from a server.
retry-limit
Sets the maximum number of attempts to connect to the provider.
Cisco IOS Voice Command Reference
VR-1650
Cisco IOS Voice Commands: R retry-delay
Command
Description
session-timeout
Sets the interval for closing the connection when there is no input or output traffic.
settlement
Enters settlement configuration mode and specifies the attributes specific to a settlement provider.
show settlement
Displays the configuration for all settlement server transactions.
shutdown/no shutdown
Deactivates the settlement provider/activates the settlement provider.
type
Configures an SAA-RTR operation type.
Cisco IOS Voice Command Reference
VR-1651
Cisco IOS Voice Commands: R retry-limit
retry-limit To set the maximum number of attempts to connect to the provider, use the retry-limit command in settlement configuration mode. To reset to the default, use the no form of this command. retry-limit number no retry-limit number
Syntax Description
number
Command Default
1 retry
Command Modes
Settlement configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Maximum number of connection attempts in addition to the first attempt. Default is 1.
Usage Guidelines
If no connection is established after the configured number of retries has been attempted, the router ceases connection attempts. The retry limit number does not count the initial connection attempt. A retry limit of one (default) results in a total of two connection attempts to every service point.
Examples
The following example sets the number of retries to 1: settlement 0 retry-limit 1
Related Commands
Command
Description
connection-timeout
Configures the time for which a connection is maintained after a communication exchange is complete.
customer-id
Identifies a carrier or ISP with a settlement provider.
device-id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
max-connection
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
response-timeout
Configures the maximum time to wait for a response from a server.
Cisco IOS Voice Command Reference
VR-1652
Cisco IOS Voice Commands: R retry-limit
Command
Description
retry-delay
Sets the time between attempts to connect with the settlement provider.
session-timeout
Sets the interval for closing the connection when there is no input or output traffic.
settlement
Enters settlement mode and specifies the attributes specific to a settlement provider.
show settlement
Displays the configuration for all settlement server transactions.
shutdown
Brings up the settlement provider.
type
Configures an SAA-RTR operation type.
Cisco IOS Voice Command Reference
VR-1653
Cisco IOS Voice Commands: R ring
ring To set up a distinctive ring for your connected telephones, fax machines, or modems, use the ring command in interface configuration mode. To disable the ring, use the no form of this command. ring cadence-number no ring cadence-number
Syntax Description
cadence-number
Number that determines the ringing cadence. Range is from 0 to 2: •
Type 0 is a primary ringing cadence—default ringing cadence for the country your router is in.
•
Type 1 is a distinctive ring—0.8 seconds on, 0.4 seconds off, 0.8 seconds on, 0.4 seconds off.
•
Type 2 is a distinctive ring—0.4 seconds on, 0.2 seconds off, 0.4 seconds on, 0.2 seconds off, 0.8 seconds on, 4 seconds off.
Command Default
0
Command Modes
Interface configuration
Command History
Release
Modification
12.0(3)T
This command was introduced on the Cisco 800 series.
Usage Guidelines
This command applies to Cisco 800 series routers. You can specify this command when creating a dial peer. This command does not work if it is not specified within the context of a dial peer. For information on creating a dial peer, see to the Cisco 800 Series Routers Software Configuration Guide.
Examples
The following example specifies the type 1 distinctive ring: ring 1
Related Commands
Command
Description
destination-pattern
Specifies the prefix, the full E.164 telephone number, or an ISDN directory number to be used for a dial peer.
dial-peer voice
Enters dial peer configuration mode, defines the type of dial peer, and defines the tag number associated with a dial peer.
no call-waiting
Disables call waiting.
Cisco IOS Voice Command Reference
VR-1654
Cisco IOS Voice Commands: R ring
Command
Description
port (dial peer)
Enables an interface on a PA-4R-DTR port adapter to operate as a concentrator port.
pots distinctive-ring-guard-time
Specifies a delay during which a telephone port can be rung after a previous call is disconnected (for Cisco 800 series routers).
show dial-peer voice
Displays configuration information and call statistics for dial peers.
Cisco IOS Voice Command Reference
VR-1655
Cisco IOS Voice Commands: R ring cadence
ring cadence To specify the ring cadence for a Foreign Exchange Station (FXS) voice port, use the ring cadence command in voice-port configuration mode. To reset to the default, use the no form of this command. ring cadence {pattern-number | define pulse interval} no ring cadence To specify the ring pattern for external calls, use the ring cadence external command. It is supported only in STCAPP. ring cadence external patternXX | define To specify the ring cadence for internal calls, use the existing ring cadence command. ring cadence patternXX | define The syntax for the ring cadence external command is the same as for the ring cadence command.
Syntax Description
pattern-number
define
Predefined ring cadence patterns. Each pattern specifies a ring-pulse time and a ring-interval time. pattern01—2 seconds on, 4 seconds off
•
pattern02—1 second on, 4 seconds off
•
pattern03—1.5 seconds on, 3.5 seconds off
•
pattern04—1 second on, 2 seconds off
•
pattern05—1 second on, 5 seconds off
•
pattern06—1 second on, 3 seconds off
•
pattern07—0.8 second on, 3.2 seconds off
•
pattern08—1.5 seconds on, 3 seconds off
•
pattern09—1.2 seconds on, 3.7 seconds off
•
pattern09—1.2 seconds on, 4.7 seconds off
•
pattern11—0.4 second on, 0.2 second off, 0.4 second on, 2 seconds off
•
pattern12—0.4 second on, 0.2 second off, 0.4 second on, 2.6 seconds off
User-definable ring cadence pattern. Each number pair specifies one ring-pulse time and one ring-interval time. You must enter numbers in pairs, and you can enter from 1 to 6 pairs. The second number in the last pair that you enter specifies the interval between rings.
Cisco IOS Voice Command Reference
VR-1656
•
Cisco IOS Voice Commands: R ring cadence
pulse
Number (1 or 2 digits) specifying ring-pulse (on) time in hundreds of milliseconds. Range is from 1 to 50, for pulses of 100 to 5000 ms. For example: 1 = 100 ms; 10 = 1 s, 40 = 4 s.
interval
Number (1 or 2 digits) specifying ring-interval (off) time in hundreds of milliseconds. Range is from 1 to 50, for pulses of 100 to 5000 ms. For example: 1 = 100 ms; 10 = 1 s, 40 = 4 s.
Command Default
Ring cadence defaults to the pattern that you specify with the cptone command.
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810.
12.0(7)XK
This command was implemented on the Cisco 2600 series and Cisco 3600 series. The patternXX keyword was added.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
15.0(1)M
This command was modified. The external keyword was added to specify the ring pattern of external calls.
Usage Guidelines
The patternXX keyword provides preset ring cadence patterns for use on any platform. The define keyword allows you to create a custom ring cadence. On the Cisco 2600 and Cisco 3600 series routers, only one or two pairs of digits can be entered under the define keyword.
Examples
The following example sets the ring cadence to 1 second on and 2 seconds off on voice port 1/0/0: voice-port 1/0/0 ring cadence pattern04
Related Commands
Command
Description
cptone
Specifies the default tone, ring, and cadence settings according to country.
ring frequency
Specifies the ring frequency for a specified FXS voice port.
Cisco IOS Voice Command Reference
VR-1657
Cisco IOS Voice Commands: R ring frequency
ring frequency To specify the ring frequency for a specified Foreign Exchange Station (FXS) voice port, use the ring frequency command in voice-port configuration mode. To reset to the default, use the no form of this command. ring frequency hertz no ring frequency hertz
Syntax Description
hertz
Ring frequency, in hertz, used in the FXS interface. Valid entries are as follows: •
Cisco 3600 series: 25 and 50. Default is 25.
Command Default
Cisco 3600 series routers: 25 Hz
Command Modes
Voice-port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco MC3810.
Usage Guidelines
Use this command to select a specific ring frequency for an FXS voice port. Use the no form of this command to reset the default value. The ring frequency you select must match the connected equipment. If set incorrectly, the attached phone might not ring or might buzz. In addition, the ring frequency is usually country-dependent. You should take into account the appropriate ring frequency for your area before configuring this command. This command does not affect ringback, which is the ringing a user hears when placing a remote call.
Examples
The following example sets the ring frequency on the voice port to 25 Hz: voice-port 1/0/0 ring frequency 25
Related Commands
Command
Description
ring cadence
Specifies the ring cadence for an FXS voice port.
ring number
Specifies the number of rings for a specified FXO voice port.
Cisco IOS Voice Command Reference
VR-1658
Cisco IOS Voice Commands: R ring number
ring number To specify the number of rings for a specified Foreign Exchange Office (FXO) voice port, use the ring number command in voice port configuration mode. To reset to the default, use the no form of this command. ring number number no ring number number
Syntax Description
number
Command Default
1 ring
Command Modes
Voice port configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 3600 series.
Usage Guidelines
Number of rings detected before answering the call. Range is from 1 to 10. The default is 1.
Use this command to set the maximum number of rings to be detected before answering a call over an FXO voice port. Use the no form of this command to reset the default value, which is one ring. Normally, this command should be set to the default so that incoming calls are answered quickly. If you have other equipment available on the line to answer incoming calls, you might want to set the value higher to give the equipment sufficient time to respond. In that case, the FXO interface would answer if the equipment online did not answer the incoming call in the configured number of rings. This command is not applicable to Foreign Exchange Station (FXS) or E&M interfaces because they do not receive ringing on incoming calls.
Examples
The following example sets 5 as the maximum number of rings to be detected before closing a connection over this voice port: voice-port 1/0/0 ring number 5
Related Commands
Command
Description
ring frequency
Specifies the ring frequency for a specified FXS voice port.
Cisco IOS Voice Command Reference
VR-1659
Cisco IOS Voice Commands: R ringing-timeout
ringing-timeout To define the timeout period for the SCCP telephony control (STC) application feature call back, use the ringing-timeout command in STC application feature callback configuration mode. To return to the default timeout period, use the no form of this command. ringing-timeout seconds no ringing-timeout
This command was integrated into Cisco IOS Release 12.4(22)T.
Usage Guidelines
Period of time in seconds. Range: 5 to 60. Default: 30.
This command changes the timeout period of the ringing timer from the default of 30 seconds to the specified value. The ringing timer specifies the number of seconds during which the calling device that is in a Callback on Busy condition can receive a Callback Ringing and after which, if the calling device does not answer, the CallBack on Busy condition is cancelled.
Examples
The following example shows how to change the timeout period of the ringing timer for CallBack on Busy from the default (30) to a new value (45). Router(config)# stcapp feature callback Router(config-stcapp-callback)# ringing-timer 45 Router(config-stcapp-callback)#
Related Commands
Command
Description
activation-code
Defines the callback activation key sequence for CallBack on Busy.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R roaming (dial peer)
roaming (dial peer) To enable roaming capability for a dial peer, use the roaming command in dial peer configuration mode. To disable roaming capability, use the no form of this command. roaming no roaming
Syntax Description
This command has no arguments or keywords.
Command Default
No roaming
Command Modes
Dial peer configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
Usage Guidelines
Use this command to enable roaming capability of a dial peer if that dial peer can terminate roaming calls. If a dial peer is dedicated to local calls only, disable roaming capability. The roaming dial peer must work with a roaming service provider. If the dial peer allows a roaming user to go through and the service provider is not roaming-enabled, the call fails.
Examples
The following example enables roaming capability for a dial peer: dial-peer voice 10 voip roaming
Related Commands
Command
Description
roaming (settlement)
Enables the roaming capability for a settlement provider.
settle-call
Limits the dial peer to using only the specific clearinghouse identified by the specified provider-number.
settlement roam-pattern
Configures a pattern to match against when determining roaming.
Cisco IOS Voice Command Reference
VR-1661
Cisco IOS Voice Commands: R roaming (settlement)
roaming (settlement) To enable roaming capability for a settlement provider, use the roaming command in settlement configuration mode. To disable roaming capability, use the no form of this command. roaming no roaming
Syntax Description
This command has no arguments or keywords.
Command Default
No roaming
Command Modes
Settlement configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
Usage Guidelines
Enable roaming capability of a settlement provider if that provider can authenticate a roaming user and route roaming calls. A roaming call is successful only if both the settlement provider and the outbound dial peer for that call are roaming-enabled.
Examples
The following example enables roaming capability for a settlement provider: settlement 0 roaming
Related Commands
Command
Description
roaming (dial peer)
Enables the roaming capability for the dial peer.
settle-call
Limits the dial peer to using only the specific clearinghouse identified by the specified provider-number.
settlement roam-pattern
Configures a pattern to match against when determining roaming.
Cisco IOS Voice Command Reference
VR-1662
Cisco IOS Voice Commands: R rrq dynamic-prefixes-accept
rrq dynamic-prefixes-accept To enable processing of additive registration request (RRQ) RAS messages and dynamic prefixes on the gatekeeper, use the rrq dynamic-prefixes-accept command in gatekeeper configuration mode. To disable processing of additive RRQ messages and dynamic prefixes, use the no form of this command. rrq dynamic-prefixes-accept no rrq dynamic-prefixes-accept
Syntax Description
This command has no arguments or keywords.
Command Default
In Cisco IOS Release 12.2(15)T, the default was set to enabled. In Cisco IOS Release 12.3(3), the default is set to disabled.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.2(15)T
This command was introduced.
12.3(3)
The default is modified to be disabled by default.
12.3(4)T
The default change implemented in Cisco IOS Release 12.3(3) was integrated in Cisco IOS Release 12.3(4)T.
Usage Guidelines
In Cisco IOS Release 12.2(15)T, the default for the rrq dynamic-prefixes-accept command was set to enabled so that the gatekeeper automatically received dynamic prefixes in additive RRQ messages from the gateway. Beginning in Cisco IOS Release 12.3(3), the default is set to disabled, and you must specify the command to enable the functionality.
Examples
The following example allows the gatekeeper to process additive RRQ messages and dynamic prefixes from the gateway: Router(config-gk)# rrq dynamic-prefixes-accept
Related Commands
Command
Description
ras rrq dynamic prefixes
Enables advertisement of dynamic prefixes in additive RRQ messages on the gateway.
Cisco IOS Voice Command Reference
VR-1663
Cisco IOS Voice Commands: R rsvp
rsvp To enable RSVP support on a transcoding or MTP device, use the rsvp command in DSP farm profile configuration mode. To disable RSVP support, use the no form of this command. rsvp no rsvp
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled
Command Modes
DSP farm profile configuration
Command History
Release
Modification
12.4(6)T
This command was introduced.
Usage Guidelines
Note
Examples
This command enables a transcoder or MTP device to register as RSVP-capable with Cisco Unified CallManager. The SCCP device acts as an RSVP agent under the control of Cisco Unified CallManager. To support RSVP, you must also enable the codec pass-through command.
This command is not supported in conferencing profiles.
The following example enables RSVP support on the transcoding device defined by profile 200: Router(config)# dspfarm profile 200 transcode Router(config-dspfarm-profile)# rsvp Router(config-dspfarm-profile)# codec pass-through
Related Commands
Command
Description
codec (DSP Farm profile)
Specifies the codecs supported by a DSP farm profile.
debug call rsvp-sync events
Displays events that occur during RSVP setup.
dspfarm profile
Enters DSP farm profile configuration mode and defines a profile for DSP farm services.
show sccp connections rsvp
Displays information about active SCCP connections that use RSVP.
Cisco IOS Voice Command Reference
VR-1664
Cisco IOS Voice Commands: R rtcp keepalive
rtcp keepalive To configure RTP Control Protocol (RTCP) keepalive report generation and generate RTCP keepalive packets, use the rtcp keepalive command in voice service configuration mode. To disable the configuration, use the no form of this command. rtcp keepalive no rtcp keepalive
Syntax Description
This command has no arguments or keywords.
Defaults
The command is disabled by default.
Command Modes
Voice service configuration (config)
Command History
Release
Modification
15.1(2)T
This command was introduced.
Usage Guidelines
Use this command to configure RTCP keepalive report generation and generate RTCP keepalive packets. The no form of the command restores the default behavior.
Examples
The following example shows how to configure RTCP keepalive report generation and generate RTCP keepalive packets: Router> enable Router# configure terminal Router(config) voice service voip Router(conf-voi-serv)# rtcp keepalive
Related Commands
Command
Description
debug voip rtcp
Enables debugging for RTCP packets.
debug voip rtp
Enables debugging for RTP packets.
debug ip rtp protocol
Enables debugging for RTP protocol.
ip rtcp report interval Configures the average reporting interval between subsequent RTCP report transmissions.
Cisco IOS Voice Command Reference
VR-1665
Cisco IOS Voice Commands: R rtp payload-type
rtp payload-type To identify the payload type of a Real-Time Transport Protocol (RTP) packet, use the rtp payload-type command in dial peer voice configuration mode. To remove the RTP payload type, use the no form of this command. rtp payload-type {cisco-cas-payload number | cisco-clear-channel number | cisco-codec-aacld number | cisco-codec-fax-ack number | cisco-codec-fax-ind number | cisco-codec-gsmamrnb number | cisco-codec-ilbc number | cisco-codec-isac number | cisco-codec-video-h263+ number | cisco-codec-video-h264 number | cisco-fax-relay number | cisco-pcm-switch-over-alaw number | cisco-pcm-switch-over-ulaw number | cisco-rtp-dtmf-relay number | lmr-tone number | nse number | nte number | nte-tone number} [comfort-noise {13 | 19}] no rtp payload-type {cisco-cas-payload number | cisco-clear-channel number | cisco-codec-fax-ack number | cisco-codec-fax-ind number | cisco-codec-gsmamrnb number | cisco-codec-ilbc number | cisco-codec-video-h263+ number | cisco-codec-video-h264 number | cisco-fax-relay number | cisco-pcm-switch-over-alaw number | cisco-pcm-switch-over-ulaw number | cisco-rtp-dtmf-relay number | lmr-tone number | nse number | nte number | nte-tone number} [comfort-noise {13 | 19}]
Cisco codec fax acknowledge. Range: 96 to 127. Default: 97.
cisco-codec-fax-ind number
Cisco codec fax indication. Range: 96 to 127. Default: 96.
cisco-codec-gsmamrnb number Cisco Global System for Mobile Adaptive Multi-Rate Narrow Band (GSMAMR-NB) codec. Range: 96 to 127. Default: 117. cisco-codec-ilbc number
Cisco internet Low Bitrate Codec (iLBC) codec. Range: 96 to 127. Default: 116.
cisco-codec-isac number
Cisco internet Speech Audio Codec (iSAC) codec. Range: 96 to 127. Default: 124.
cisco-codec-video-h263+ number
RTP video codec H.263+ payload type. Range: 96 to 127. Default: 118.
cisco-codec-video-h264 number RTP video codec H.264 payload type. Range: 96 to 127. Default: 119. cisco-fax-relay number
LMR payload type. Range: 96 to 127. Default: 0. The default value is set by the no rtp payload-type lmr-tone command.
Cisco IOS Voice Command Reference
VR-1666
Cisco IOS Voice Commands: R rtp payload-type
nse number
A named signaling event (NSE). Range: 96 to 117. Default: 100.
nte number
A named telephone event (NTE). Range: 96 to 127. Default: 101.
nte-tone number
RFC-2833 tone payload type. Range 96 to 127. Default: 101.
comfort-noise {13 | 19}
(Optional) RTP payload type of comfort noise. The July 2001 draft entitled RTP Payload for Comfort Noise, from the Internet Engineering Task Force (IETF) Audio/Video Transport (AVT) working group, designates 13 as the payload type for comfort noise. If you are connecting to a gateway that complies with the RTP Payload for Comfort Noise draft, use 13. Use 19 only if you are connecting to older Cisco gateways that use DSPware before version 3.4.32. Note
This command option is not available on the Cisco AS5400 running NextPort digital signal processors (DSPs). This command option is available on the Cisco AS5400 only if the platform has a high-density packet voice/fax feature card (AS5X-FC) with one or more AS5X-PVDM2-64 DSP modules installed. This support was added in Cisco IOS Release 12.4(4)XC, and integrated into Release 12.4(9)T, and later 12.4T releases.
Command Default
No RTP payload type is configured.
Command Modes
Dial peer voice configuration (config-dial-peer)
Command History
Release
Modification
12.2(2)T
This command was introduced.
12.2(2)XB
This command was modified. The nte and comfort-noise keywords were added.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T.
12.4(4)XC
This command was modified. The cisco-codec-gsmamrnb keyword was added.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T.
12.4(11)T
This command was modified. The cisco-codec-ilbc, cisco-codec-video-h263+, and cisco-codec-video-h264 keywords were added.
12.4(15)XY
This command was modified. The lmr-tone and nte-tone keywords were added.
12.4(20)T
This command was integrated into Cisco IOS Release 12.4(20)T.
IOS Release XE 2.5
This command was integrated into Cisco IOS XE Release 2.5.
15.1(1)T
This command was modified. The cisco-codec-isac keyword was added.
Cisco IOS Voice Command Reference
VR-1667
Cisco IOS Voice Commands: R rtp payload-type
Usage Guidelines
Use this command to identify the payload type of an RTP. Use this command after the dtmf-relay command is used to choose the NTE method of DTMF relay for a Session Initiation Protocol (SIP) call. Configured payload types of NSE and NTE exclude certain values that have been previously hard-coded with Cisco-proprietary meanings. Do not use the following numbers, which have preassigned values: 96, 97, 100, 117, 121 to 123, and 125 to 127. Use of these values results in an error message when the command is entered. You must first reassign the value in use to a different unassigned number, for example: rtp payload-type cisco-codec-ilbc 100 ERROR: value 100 in use! rtp payload-type nse 105 rtp payload-type cisco-codec-ilbc 100
Examples
The following example shows how to identify the RTP payload type as GSMAMR-NB115: Router(config-dial-peer)# rtp payload-type cisco-codec-gsmamrnb 115
The following example shows how to identify the RTP payload type as NTE 99: Router(config-dial-peer)# rtp payload-type nte 99
The following example shows how to identify the RTP payload type for the iLBC as 100: Router(config-dial-peer)# rtp payload-type cisco-codec-ilbc 100
Related Commands
Command
Description
dtmf-relay
Specifies how an H.323 or SIP gateway relays DTMF tones between telephony interfaces and an IP network.
Cisco IOS Voice Command Reference
VR-1668
Cisco IOS Voice Commands: R rtp send-recv
rtp send-recv To configure a Cisco IOS Session Initiation Protocol (SIP) gateway to establish a bidirectional voice path as soon as it receives a SIP 183 PROGRESS message with Session Description Protocol (SDP), use the rtp send-recv command in voice service SIP configuration mode. To configure the gateway to establish a backward-only media cut-through voice path upon receipt of a 183 PROGRESS message with SDP that persists until the call progresses to the connect state, use the no form of this command. rtp send-recv no rtp send-recv
Syntax Description
This command has no arguments or keywords.
Command Default
A bidirectional voice path is established upon receipt of a 183 PROGRESS message with SDP.
Command Modes
Voice service SIP configuration (conf-serv-sip)
Command History
Release
Modification
12.4(15)XZ
This command was introduced.
12.4(20)T
This command was integrated into Cisco IOS Release 12.4(20)T.
Usage Guidelines
The default behavior on a Cisco IOS SIP gateway is to establish a bidirectional voice path from the moment it receives a SIP 183 PROGRESS message with SDP. However, this can result in clipping on some voice platforms if both parties send audio at the same time, such as during a call setup process when interactive voice response (IVR) and a caller both speak simultaneously. To establish the voice path in the backward direction only until the call is connected, use the no rtp send-recv command in voice service SIP configuration mode. A backward-only voice path operates only during the connection attempt—once a call is connected, the voice path automatically converts to bidirectional sending and receiving of Real-Time Transport Protocol (RTP) packets and RTP control packets (RTCPs). However, if the no rtp send-recv command is configured on a SIP gateway, no inband or RFC 2833-based dual tone multifrequency (DTMF) digits can be sent in the forward direction until after the call is connected and the bidirectional voice path is established.
Examples
The following example enables RTP backward-only media cut-through on a Cisco IOS SIP gateway: Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# no rtp send-recv
Cisco IOS Voice Command Reference
VR-1669
Cisco IOS Voice Commands: R rtp-ssrc multiplex
rtp-ssrc multiplex To multiplex Real-Time Transport Control Protocol (RTCP) packets with RTP packets and to send multiple synchronization source in RTP headers (SSRCs) in a RTP session, use the rtp-ssrc multiplex command in voice service or dial peer voice configuration mode. To disable the configuration, use the no form of this command. Syntax Available Under Voice Service Configuration Mode
rtp-ssrc multiplex no rtp-ssrc multiplex Syntax Available Under Dial Peer Voice Configuration Mode
rtp-ssrc multiplex [system] no rtp-ssrc multiplex [system]
Syntax Description
system
Command Default
Under voice service configuration mode, the rtp-ssrc multiplex command is not enabled and hence there is no interoperation with Cisco TelePresence System (CTS).
Uses the system value. This is the default value.
At the dial-peer level, the rtp-ssrc multiplex command uses the global configuration level settings.
Command Modes
Voice service configuration (conf-voi-serv) Dial peer voice configuration (config-dial-peer)
Command History
Release
Modification
12.4(15)XY
This command was introduced.
12.4(20)T
This command was integrated into Cisco IOS Release 12.4(20)T.
Usage Guidelines
The rtc-ssrc multiplex command is used for the interoperation with CTS.
Examples
The following example shows how to multiplex RTCP packets with RTP packets and send multiple SSRCs in a RTP session: Router# configure terminal Router(config)# dial-peer voice 234 voip Router(config-dial-peer)# rtp-ssrc multiplex system
Cisco IOS Voice Command Reference
VR-1670
Cisco IOS Voice Commands: R rtsp client session history duration
rtsp client session history duration To specify how long to keep Real Time Streaming Protocol (RTSP) client history records in memory, use the rtsp client session history duration command in global configuration mode. To reset to the default, use the no form of this command. rtsp client session history duration minutes no rtsp client session history duration
Syntax Description
minutes
Command Default
10 minutes
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco AS5300.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1750 and Cisco 1751. This release does not support any other Cisco platforms.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Examples
Duration, in minutes, to keep the record. Range is from 1 to 10000. Default is 10.
The following example sets the duration for the RTSP session history to 500 minutes: rtsp client session history duration 500
Related Commands
Command
Description
call application voice load
Allows reload of an application that was loaded via the MGCP scripting package.
rtsp client session history records
Specifies the number of RTSP client session history records kept during the session.
show call application voice
Displays all TCL or MGCP scripts that are loaded.
show rtsp client session
Displays cumulative information about the RTSP session records.
Cisco IOS Voice Command Reference
VR-1671
Cisco IOS Voice Commands: R rtsp client rtpsetup enable
rtsp client rtpsetup enable To configure a router to send the IP address in a Real Time Streaming Protocol (RTSP) setup message, use the rtsp client rtpsetup enable command in global configuration mode. To disable the configuration, use the no form of this command. rtsp client rtpsetup enable no rtsp client rtpsetup enable
Syntax Description
This command has no arguments or keywords.
Command Default
This command is disabled.
Command Modes
Global configuration (config)
Command History
Release
Modification
15.0(1)M
This command was introduced in a release earlier than Cisco IOS Release 15.0(1)M.
Examples
The following example shows how to configure a router to send the IP address in an RTSP setup message: Router# configure terminal Router(config)# rtsp client rtpsetup enable
Related Commands
Command
Description
rtsp client session history duration
Specifies how long to keep RTSP client history records in memory.
rtsp client timeout connect
Sets the number of seconds allowed for the router to establish a TCP connection to an RTSP server.
Cisco IOS Voice Command Reference
VR-1672
Cisco IOS Voice Commands: R rtsp client session history records
rtsp client session history records To configure the number of records to keep in the Real Time Streaming Protocol (RTSP) client session history, use the rtsp client session history records command in global configuration mode. To reset to the default, use the no form of this command. rtsp client session history records number no rtsp client session history records number
Syntax Description
number
Command Default
50 records
Command Modes
Global configuration
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco AS5300.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1750 and Cisco 1751. This release does not support any other Cisco platforms.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Examples
Number of records to retain in a session history. Range is from 1 to 100000. Default is 50.
The following example specifies that a total of 500 records are to be kept in the RTSP client history: rtsp client session history records 500
Related Commands
Command
Description
call application voice load
Allows reload of an application that was loaded via the MGCP scripting package.
rtsp client session history duration
Specifies the how long the RTSP is kept during the session.
show call application voice
Displays all Tcl or MGCP scripts that are loaded.
Cisco IOS Voice Command Reference
VR-1673
Cisco IOS Voice Commands: R rtsp client timeout connect
rtsp client timeout connect To set the number of seconds allowed for the router to establish a TCP connection to a Real -Time Streaming Protocol (RTSP) server, use the rtsp client timeout connect command in global configuration mode. To reset to the default, use the no form of this command. rtsp client timeout connect seconds no rtsp client timeout connect
Syntax Description
seconds
Command Default
3 seconds
Command Modes
Global configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
How long, in seconds, the router waits to connect to the server before timing out. Range is 1 to 20.
Usage Guidelines
This command determines when the router abandons its attempt to connect to an RTSP server and declares a timeout error, if a connection cannot be established after the specified number of seconds.
Examples
The following example sets the connection timeout to 10 seconds: rtsp client timeout connect 10
Related Commands
Command
Description
rtsp client session history records
Sets the maximum number of records to store in the RTSP client session history.
rtsp client timeout message
Sets the number of seconds that the router waits for a response from an RTSP server.
Cisco IOS Voice Command Reference
VR-1674
Cisco IOS Voice Commands: R rtsp client timeout message
rtsp client timeout message To set the number of seconds that the router waits for a response from a Real -Time Streaming Protocol (RTSP) server, use the rtsp client timeout message command in global configuration mode. To reset to the default, use the no form of this command. rtsp client timeout message seconds no rtsp client timeout message
Syntax Description
seconds
Command Default
3 seconds
Command Modes
Global configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
How long, in seconds, the router waits for a response from the server after making a request. Range is 1 to 20.
Usage Guidelines
This command sets how long the router waits for the RTSP server to respond to a request before declaring a timeout error.
Examples
The following example sets the request timeout to 10 seconds: rtsp client timeout message 10
Related Commands
Command
Description
rtsp client session history records
Sets the maximum number of records to store in the RTSP client session history.
rtsp client timeout connect
Sets the number of seconds allowed for the router to establish a TCP connection to an RTSP server.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R rule (ENUM configuration)
rule (ENUM configuration) To define a rule for an ENUM match table, use the rule command in ENUM configuration mode. To delete the rule, use the no form of this command. rule rule-number preference /match-pattern/ /replacement-rule /domain-name no rule rule-number preference /match-pattern/ /replacement-rule /domain-name
Syntax Description
rule-number
Assigns an identification number to the rule. Range is from 1 to 2147483647.
preference
Assigns a preference value to the rule. Range is from 1 to 2147483647. Lower values have higher preference.
/match-pattern
Stream editor (SED) expression used to match incoming call information. The slash “/” is a delimiter in the pattern.
/replacement-rule
SED expression used to replace match-pattern in the call information. The slash “/” is a delimiter in the pattern.
/domain-name
Domain name to be used while the query to the DNS server is sent.
Command Default
No default behavior or values
Command Modes
ENUM configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
Usage Guidelines
Table 36 shows examples of match patterns, input strings, and result strings for the rule (voice translation-rule) command. Table 36
Match Patterns, Input Strings and Result Strings
Match Pattern
Replacement Pattern Input String
Result String
Description
/^.*/
//
4085550100
—
Any string to null string.
/^456\(.*\) /
/555\1/
5550100
5550100
Match from the beginning of the input string.
/\(^...\)45 6\(...\)/
/\1555\2/
408555010
4085550100
Match from the middle of the input string.
/\(.*\)0100 /
/\0199/
4085550100
4085550199
Match from the end of the input string.
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Cisco IOS Voice Commands: R rule (ENUM configuration)
Table 36
Match Patterns, Input Strings and Result Strings (continued)
Match Pattern
Replacement Pattern Input String
Result String
Description
/^1#\(.*\)/
/\1/
1#2345
2345
Replace match string with null string.
/^408...\(8 333\)/
/555\1/
4085550100
5550100
Match multiple patterns.
Rules are entered in any order, but their preference number determines the sequence in which they are used for matching against the input string, which is a called number. A lower preference number is used before a higher preference number. If a match is found, the input string is modified according to the replacement rule, and the E.164 domain name is attached to the modified number. This longer number is sent to a Domain Name System (DNS) server to determine a destination for the call. The server returns one or more URLs as possible destinations. The originating gateway tries to place the call using each URL in order of preference. If a call cannot be completed using any of the URLs, the call is disconnected.
Examples
The following example defines ENUM rule number 3 with preference 2. The beginning of the call string is checked for digits 9011; when a match is found, 9011 is replaced with 1408 and the call is sent out as an e164.arpa number. Router(config)# voice enum-match-table number Router(config-enum)# rule 3 2 /^9011\(.*\)//+1408\1/ arpa
Related Commands
Command
Description
show voice enum-match-table
Displays the configuration of a voice ENUM match table.
test enum
Tests the ENUM rule.
voice enum-match-table
Initiates the definition of a voice ENUM match table.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R rule (voice translation-rule)
rule (voice translation-rule) To define a translation rule, use the rule command in voice translation-rule configuration mode. To delete the translation rule, use the no form of this command. Match and Replace Rule
Priority of the translation rule. Range is from 1 to 15.
/match-pattern/
Stream editor (SED) expression used to match incoming call information. The slash ‘/’ is a delimiter in the pattern.
/replace-pattern/
SED expression used to replace the match pattern in the call information. The slash ‘/’ is a delimiter in the pattern.
type match-type replace-type
(Optional) Number type of the call. Valid values for the match-type argument are as follows: •
abbreviated—Abbreviated representation of the complete number as supported by this network.
•
any—Any type of called number.
•
international—Number called to reach a subscriber in another country.
•
national—Number called to reach a subscriber in the same country, but outside the local network.
•
network—Administrative or service number specific to the serving network.
•
reserved—Reserved for extension.subscriber—Number called to reach a subscriber in the same local network.
•
unknown—Number of a type that is unknown by the network.
Valid values for the replace-type argument are as follows:
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•
abbreviated—Abbreviated representation of the complete number as supported by this network.
•
international—Number called to reach a subscriber in another country.
•
national—Number called to reach a subscriber in the same country, but outside the local network.
Cisco IOS Voice Commands: R rule (voice translation-rule)
type match-type replace-type (continued)
plan match-type replace-type
•
network—Administrative or service number specific to the serving network.
•
reserved—Reserved for extension.
•
subscriber—Number called to reach a subscriber in the same local network.
•
unknown—Number of a type that is unknown by the network.
(Optional) Numbering plan of the call. Valid values for the match-type argument are as follows: •
any—Any type of dialed number.
•
data
•
ermes
•
isdn
•
national—Number called to reach a subscriber in the same country, but outside the local network.
•
private
•
reserved—Reserved for extension.
•
telex
•
unknown—Number of a type that is unknown by the network.
Valid values for the replace-type argument are as follows:
reject
•
data
•
ermes
•
isdn
•
national—Number called to reach a subscriber in the same country, but outside the local network.
•
private
•
reserved—Reserved for extension.
•
telex
•
unknown—Number of a type that is unknown by the network.
The match pattern of a translation rule is used for call-reject purposes.
Command Default
No default behavior or values
Command Modes
Voice translation-rule configuration
Command History
Release
Modification
12.2(11)T
This command was introduced with a new syntax in voice-translation-rule configuration mode.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R rule (voice translation-rule)
Usage Guidelines
Note
Use this command in conjunction after the voice translation-rule command. An earlier version of this command uses the same name but is used after the translation-rule command and has a slightly different command syntax. In the older version, you cannot use the square brackets when you are entering command syntax. They appear in the syntax only to indicate optional parameters, but are not accepted as delimiters in actual command entries. In the newer version, you can use the square brackets as delimiters. Going forward, we recommend that you use this newer version to define rules for call matching. Eventually, the translation-rule command will not be supported.
A translation rule applies to a calling party number (automatic number identification [ANI]) or a called party number (dialed number identification service [DNIS]) for incoming, outgoing, and redirected calls within Cisco H.323 voice-enabled gateways. Number translation occurs several times during the call routing process. In both the originating and terminating gateways, the incoming call is translated before an inbound dial peer is matched, before an outbound dial peer is matched, and before a call request is set up. Your dial plan should account for these translation steps when translation rules are defined. Table 37 shows examples of match patterns, input strings, and result strings for the rule (voice translation-rule) command. Table 37
Match Patterns, Input Strings and Result Strings
Match Pattern
Replacement Pattern Input String
Result String
Description
/^.*/
//
4085550100
—
Any string to null string.
//
//
4085550100
4085550100
Match any string but no replacement. Use this to manipulate the call plan or call type.
/\(^...\)45 6\(...\)/
/\1555\2/
4084560177
4085550177
Match from the middle of the input string.
/\(.*\)0120 /
/\10155/
4081110120
4081110155
Match from the end of the input string.
/^1#\(.*\)/
/\1/
1#2345
2345
Replace match string with null string.
/^408...\(8 333\)/
/555\1/
4087770100
5550100
Match multiple patterns.
/1234/
/00&00/
5550100
55500010000
Match the substring.
/1234/
/00\000/
5550100
55500010000
Match the substring (same as &).
The software verifies that a replacement pattern is in a valid E.164 format that can include the permitted special characters. If the format is not valid, the expression is treated as an unrecognized command. The number type and calling plan are optional parameters for matching a call. If either parameter is defined, the call is checked against the match pattern and the selected type or plan value. If the call matches all the conditions, the call is accepted for additional processing, such as number translation. Several rules may be grouped together into a translation rule, which gives a name to the rule set. A translation rule may contain up to 15 rules. All calls that refer to this translation rule are translated against this set of criteria. The precedence value of each rule may be used in a different order than that in which they were typed into the set. Each rule’s precedence value specifies the priority order in which the rules are to be used. For example, rule 3 may be entered before rule 1, but the software uses rule 1 before rule 3.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R rule (voice translation-rule)
The software supports up to 128 translation rules. A translation profile collects and identifies a set of these translation rules for translating called, calling, and redirected numbers. A translation profile is referenced by trunk groups, source IP groups, voice ports, dial peers, and interfaces for handling call translation.
Examples
The following example applies a translation rule. If a called number starts with 5550105 or 70105, translation rule 21 uses the rule command to forward the number to 14085550105 instead. Router(config)# voice translation-rule 21 Router(cfg-translation-rule)# rule 1 /^5550105/ /14085550105/ Router(cfg-translation-rule)# rule 2 /^70105/ /14085550105/
In the next example, if a called number is either 14085550105 or 014085550105, after the execution of translation rule 345, the forwarding digits are 50105. If the match type is configured and the type is not “unknown,” dial-peer matching is required to match the input string numbering type. Router(config)# voice translation-rule 345 Router(cfg-translation-rule)# rule 1 /^14085550105/ /50105/ plan any national Router(cfg-translation-rule)# rule 2 /^014085550105/ /50105/ plan any national
Related Commands
Command
Description
show voice translation-rule
Displays the parameters of a translation rule.
voice translation-rule
Initiates the voice translation-rule definition.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: R rule (voice translation-rule)
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S This chapter contains commands to configure and maintain Cisco IOS voice applications. The commands are presented in alphabetical order. Some commands required for configuring voice may be found in other Cisco IOS command references. Use the master index of commands or search online to find these commands. For detailed information on how to configure these applications and features, refer to the Cisco IOS Voice Configuration Library.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S sccp
sccp To enable the Skinny Client Control Protocol (SCCP) protocol and its related applications (transcoding and conferencing), use the sccp command in global configuration mode. To disable the protocol, use the no form of this command. sccp no sccp
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled
Command Modes
Global configuration
Command History
Release
Modification
12.1(5)YH
This command was introduced on the Cisco VG200.
12.2(13)T
This command was implemented on the Cisco 2600 series, Cisco 3620, Cisco 3640, Cisco 3660, and Cisco 3700 series.
Usage Guidelines
The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs) or high-density voice (HDV) transcoding/conferencing DSP farms (NM-HDV-FARMs) to provide digital-signal-processor (DSP) resources. SCCP and its related applications (transcoding and conferencing) become enabled only if digital-signal-processor (DSP) resources for these applications are configured, DSP-farm service is enabled, and the Cisco CallManager registration process is completed. The no form of this command disables SCCP and its applications by unregistering from the active Cisco CallManager, dropping existing connections, and freeing allocated resources.
Examples
The following example sets related values and then enables SCCP: Router(config)# Router(config)# Router(config)# Router(config)# Router(config)# Router(config)#
sccp sccp sccp sccp sccp end
Cisco IOS Voice Command Reference
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ccm 10.10.10.1 priority 1 local fastEthernet 0/0 switchback timeout guard 180 ip precedence 5
Cisco IOS Voice Commands: S sccp
Related Commands
Command
Description
dspfarm (DSP farm)
Enables DSP-farm service.
show dspfarm
Displays summary information about DSP resources.
show sccp
Displays the SCCP configuration information and current status.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S sccp blf-speed-dial retry-interval
sccp blf-speed-dial retry-interval To set the retry timeout for Busy Lamp Field (BLF) notification for speed-dial numbers on SCCP phones registered to an external Cisco Unified CME router, use the sccp blf-speed-dial retry-interval command in presence configuration mode. To reset to the default, use the no form of this command. sccp blf-speed-dial retry-interval seconds limit number no sccp blf-speed-dial retry-interval
Syntax Description
seconds
Retry timeout in seconds. Range: 60 to 3600. Default: 60.
limit number
Maximum number of retries. Range: 10 to 100. Default: 10.
Command Default
Retry timeout is 60 seconds; retry limit is 10.
Command Modes
Presence configuration (config-presence)
Command History
Cisco IOS Release
Modification
12.4(11)XJ
This command was introduced.
12.4(15)T
This command was integrated into Cisco IOS Release 12.4(15)T.
Usage Guidelines
This command specifies how frequently the router attempts to subscribe for the line status of an external directory number when the BLF speed-dial feature is configured on a SCCP phone. This retry mechanism is used when either the presentity does not exist or the router receives a terminated NOTIFY from the external presence server. If the subscribe request toward the external server fails after the configured number of retries, the subscribe request from the phone is rejected.
Examples
The following example shows the BLF speed-dial retry interval set to 100 seconds and the limit to 25: Router(config)# presence Router(config-presence)# sccp blf-speed-dial retry-interval 100 limit 25
Related Commands
Command
Description
allow subscribe
Allows internal watchers to monitor external presence entities (directory numbers).
blf-speed-dial
Enables BLF monitoring for a speed-dial number on a phone registered to Cisco Unified CME.
server
Specifies the IP address of a presence server for sending presence requests from internal watchers to external presence entities.
show presence global
Displays configuration information about the presence service.
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Cisco IOS Voice Commands: S sccp ccm
sccp ccm To add a Cisco Unified Communications Manager server to the list of available servers and set various parameters—including IP address or Domain Name System (DNS) name, port number, and version number—use the sccp ccm command in global configuration mode. To remove a particular server from the list, use the no form of this command. NM-HDV or NM-HDV-FARM Voice Network Modules
IPv4 address of the Cisco Unified Communications Manager server.
ipv6-address
IPv6 address of the Cisco Unified Communications Manager server.
dns
DNS name.
identifier identifier-number
Specifies the number that identifies the Cisco Unified Communications Manager server. The range is 1 to 65535.
priority priority
Specifies the priority of this Cisco Unified Communications Manager server relative to other connected servers. The range is 1 (highest) to 4 (lowest). Note
port port-number
This keyword is required only for NM-HDV and NM-HDV-FARM modules. Do not use this keyword if you are using the NM-HDV2 or NM-HD-1V/2V/2VE; set the priority using the associate ccm command in the Cisco Unified Communications Manager group.
(Optional) Specifies the TCP port number. The range is 1025 to 65535. The default is 2000.
version version-number (Optional) Cisco Unified Communications Manager version. Valid versions are 3.0, 3.1, 3.2, 3.3, 4.0, 4.1, 5.0.1, 6.0, and 7.0+. There is no default value. trustpoint
(Optional) Specifies the trustpoint for Cisco Unified Communications Manager certificate.
This command was modified. The identifier keyword and additional values for Cisco Unified Communications Manager versions were added.
12.4(11)XW
This command was modified. The 6.0 keyword was added to the list of version values.
12.4(20)T
This command was integrated into Cisco IOS Release 12.4(20)T.
12.4(22)T
This command was modified. Support for IPv6 was added. The version keyword and version-number argument were changed from being optional to being required, and the 7.0+ keyword was added.
15.0(1)M
This command was modified in a release earlier than Cisco IOS Release 15.0(1)M. The trustpoint keyword and the label argument were added.
You can configure up to four Cisco Unified Communications Manager servers—a primary and up to three backups—to support digital signal processor (DSP) farm services. To add the Cisco Unified Communications Manager server to a Cisco Unified Communications Manager group, use the associate ccm command. IPv6 support is provided for registration with Cisco Unified CM version 7.0 and later. To enable Ad Hoc or Meet-Me hardware conferencing in Cisco Unified CME, you must first set the version keyword to 4.0 or a later version. Beginning with Cisco IOS Release 12.4(22)T users manually configuring the sccp ccm command must provide the version. Existing router configurations are not impacted because automatic upgrade and downgrade are supported.
Examples
The following example shows how to add the Cisco Unified Communications Manager server with IP address 10.0.0.0 to the list of available servers: Router(config)# sccp ccm 10.0.0.0 identifier 3 port 1025 version 4.0
The following example shows how to add the Cisco Unified CallManager server whose IPv6 address is 2001:DB8:C18:1::102: Router(config)# sccp ccm 2001:DB8:C18:1::102 identifier 2 version 7.0
Related Commands
Command
Description
associate ccm
Associates a Cisco Unified Communications Manager server with a Cisco Unified Communications Manager group and establishes its priority within the group.
sccp
Enables SCCP and its associated transcoding and conferencing applications.
sccp ccm group
Creates a Cisco Unified Communications Manager group and enters SCCP Cisco Unified Communications Manager configuration mode.
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Cisco IOS Voice Commands: S sccp ccm
Command
Description
sccp local
Selects the local interface that SCCP applications use to register with Cisco Unified Communications Manager.
show sccp
Displays SCCP configuration information and current status.
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Cisco IOS Voice Commands: S sccp ccm group
sccp ccm group To create a Cisco Unified Communications Manager group and enter SCCP Cisco CallManager configuration mode, use the sccp ccm group command in global configuration mode. To remove a particular Cisco Unified Communications Manager group, use the no form of this command. sccp ccm group group-number no sccp ccm group group-number
Syntax Description
group-number
Command Default
No groups are defined, so all servers are configured individually.
Command Modes
Global configuration (config)
Command History
Release
Modification
12.3(8)T
This command was introduced.
12.4(22)T
This command was modified. Support for IPv6 was added.
15.0(1)M
This command was modified. The number of group number range was increased to 50.
Number that identifies the Cisco Unified Communications Manager group. Range is 1 to 50.
Usage Guidelines
Use this command to group Cisco Unified Communications Manager servers that are defined using the sccp ccm command. You can associate designated DSP farm profiles using the associate profile command so that the DSP services are controlled by the Cisco Unified Communications Manager servers in the group.
Examples
The following example enters SCCP Cisco CallManager configuration mode and associates Cisco Unified Communications Manager 25 with Cisco Unified Communications Manager group 10: Router(config)# sccp ccm group 10 Router(config-sccp-ccm)# associate ccm 25 priority 2
Related Commands
Command
Description
associate ccm
Associates a Cisco Unified Communications Manager server with a Cisco Unified Communications Manager group and establishes its priority within the group.
associate profile
Associates a DSP farm profile with a Cisco Unified Communications Manager group.
bind interface
Binds an interface with a Cisco Unified Communications Manager group.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S sccp ccm group
Command
Description
connect interval
Specifies the amount of time that a DSP farm profile waits before attempting to connect to a Cisco Unified Communications Manager when the current Cisco Unified Communications Manager fails to connect.
connect retries
Specifies the number of times that a DSP farm attempts to connect to a Cisco Unified Communications Manager when the current Cisco Unified Communications Manager connections fails.
sccp ccm
Adds a Cisco Unified Communications Manager server to the list of available servers.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S sccp codec mask
sccp codec mask To mask a codec type so that it is not used by Cisco CallManager, use the sccp codec mask command in global configuration mode. To unmask a codec, use the no form of this command. sccp codec codec mask no sccp codec codec mask
Syntax Description
codec
Codec to mask. Values are the following: •
g711alaw
•
g711ulaw
•
g729abr8
•
g729ar8
•
g729br8
•
g729r8
Command Default
No codecs are masked.
Command Modes
Global configuration
Command History
Release
Modification
12.1(5)YH4
This command was introduced.
12.2(13)T
This command was integrated into Cisco IOS Release 12.2(13)T.
12.4(11)XJ2
The gsmefr and gsmfr keywords were removed as configurable codec options for all platforms with the exception of the gsmfr codec on the Cisco AS5400 and AS5350 with MSAv6 dsps.
12.4(15)T
This command was integrated into Cisco IOS Release 12.4(15)T.
Usage Guidelines
Note
This command prevents the voice gateway from reporting codec types that are masked so that Cisco CallManager only selects codec types that are supported by the endpoints.
You must enable this command before Skinny Client Control Protocol (SCCP) is enabled. If the sccp codec mask command is used when SCCP is active, you must disable the SCCP using the no sccp command and then re-enable sccp for the sccp codec mask command to take effect.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S sccp codec mask
Examples
The following example shows how to mask codec type G.711 ulaw and G.729r8: sccp codec g711ulaw mask sccp codec g729r8 mask
Related Commands
Command
Description
sccp
Enables SCCP and related applications.
sccp ccm
Adds a Cisco CallManager server to the list of available servers and sets various parameters.
sccp local
Selects the local interface that SCCP applications use to register with Cisco CallManager.
show sccp
Displays SCCP configuration information and current status.
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Cisco IOS Voice Commands: S sccp ip precedence
sccp ip precedence To set the IP precedence value to be used by Skinny Client Control Protocol (SCCP), use the sccp ip precedence command in global configuration mode. To reset to the default, use the no form of this command. sccp ip precedence value no sccp ip precedence
Syntax Description
value
Command Default
5
Command Modes
Global configuration
Command History
Release
Modification
12.1(5)YH
This command was introduced on the Cisco VG200.
12.2(13)T
This command was implemented on the Cisco 2600 series, Cisco 3620, Cisco 3640, Cisco 3660, and Cisco 3700 series.
IP precedence value. Range is from 1 (lowest) to 7 (highest).
Usage Guidelines
The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs) or high-density voice (HDV) transcoding/conferencing DSP farms (NM-HDV-FARMs) to provide digital-signal-processor (DSP) resources.
Examples
The following example sets IP precedence to the highest possible value: Router# sccp ip precedence 1
Related Commands
Command
Description
dspfarm (DSP farm)
Enables DSP-farm service.
sccp
Enables SCCP and its associated transcoding and conferencing applications.
show sccp
Displays the SCCP configuration information and current status.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S sccp local
sccp local To select the local interface that Skinny Client Control Protocol (SCCP) applications (transcoding and conferencing) use to register with Cisco CallManager, use the sccp local command in global configuration mode. To deselect the interface, use the no form of this command. sccp local interface-type interface-number [port port-number] no sccp local interface-type interface-number
Syntax Description
interface-type
Interface type that the SCCP application uses to register with Cisco CallManager. The type can be an interface address or a virtual-interface address such as Ethernet.
interface-number
Interface number that the SCCP application uses to register with Cisco CallManager.
port port-number
(Optional) Port number used by the selected interface. Range is 1025 to 65535. Default is 2000.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
Modification
12.1(5)YH
This command was introduced.
12.2(13)T
This command was integrated into Cisco IOS Release 12.2(13)T.
12.3(14)T
The port keyword and port-number argument were added.
Usage Guidelines
Note
Examples
The router must be equipped with one or more voice network modules that provide DSP resources.
If the default port is used by another application, the SCCP application fails to register to Cisco CallManager. Use the port keyword with the port-number argument to specify a different port for SCCP to use for registering with Cisco CallManager.
The following example selects a Fast Ethernet interface for SCCP applications to use to register with Cisco CallManager: sccp local FastEthernet 0/0
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Cisco IOS Voice Commands: S sccp local
Related Commands
Command
Description
dsp services dspfarm
Enables DSP-farm services.
sccp
Enables SCCP and its associated transcoding and conferencing applications.
show sccp
Displays the SCCP configuration information and current status.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S sccp plar
sccp plar To enter SCCP PLAR configuration mode, use the sccp plar command in global configuration mode. To disable private line automatic ringdown (PLAR) on all ports, use the no form of this command. sccp plar no sccp plar
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled (PLAR is not enabled on any port).
Command Modes
Global configuration
Command History
Release
Modification
12.4(6)T
This command was introduced.
Usage Guidelines
This command is used for enabling PLAR features on analog FXS endpoints that use Skinny Client Control Protocol (SCCP) for call control. Use the voiceport command to enable a specific analog voice port for PLAR.
Examples
The following example sets PLAR on voice ports 2/0, 2/1, and 2/3: Router(config)# sccp plar Router(config-sccp-plar)# voiceport 2/0 dial 3660 digit 1234 wait-connect 500 interval 200 Router(config-sccp-plar)# voiceport 2/1 dial 3264 digit 678,,,9*0,,#123 interval 100 Router(config-sccp-plar)# voiceport 2/3 dial 3478 digit 34567 wait-connect 500
Related Commands
Command
Description
dial peer voice
Enters dial peer configuration mode and defines a dial peer.
voiceport
Enables a PLAR connection for an analog phone.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S sccp switchback timeout guard
sccp switchback timeout guard To set the Skinny Client Control Protocol (SCCP) switchback guard timer, use the sccp switchback timeout guard command in global configuration mode. To reset to the default, use the no form of this command. sccp switchback timeout guard seconds no sccp switchback timeout guard
Syntax Description
seconds
Command Default
1200 seconds
Command Modes
Global configuration
Command History
Release
Modification
12.1(5)YH
This command was introduced on the Cisco VG200.
12.2(13)T
This command was implemented on the Cisco 2600 series, Cisco 3620, Cisco 3640, Cisco 3660, and Cisco 3700 series.
Usage Guidelines
Guard timer value, in seconds. Range is from 180 to 7200. Default is 1200.
The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs) or high-density voice (HDV) transcoding/conferencing DSP farms (NM-HDV-FARMs) to provide digital-signal-processor (DSP) resources. You can use the guard timer value for the switchback algorithm that follows the Graceful Timer method.
Examples
The following example sets the switchback guard timer value to 180 seconds (3 minutes): Router# sccp switchback timeout guard 180
Related Commands
Command
Description
dspfarm (DSP farm)
Enables DSP-farm service.
sccp
Enables SCCP and its associated transcoding and conferencing applications.
show sccp
Displays the SCCP configuration information and current status.
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Cisco IOS Voice Commands: S scenario-cause
scenario-cause To configure new Q.850 call-disconnect cause codes for use if an H.323 call fails, use the scenario-cause command in H.323-voice-service configuration mode. To revert to the defaults, use the no form of this command. scenario-cause {arj-default | timeout {arq | t301 | t303 | t310} code-id} no scenario-cause {arj-default | timeout {arq | t301 | t303 | t310}}
Syntax Description
arj-default
Q.850 call-disconnect cause code for use if a call fails for reasons that are assigned to the Admission Reject (ARJ) default cause code.
timeout arq
Q.850 call-disconnect cause code for use if the H.323 gatekeeper Automatic Repeat Request (ARQ) timer expires.
timeout t301
Q.850 call-disconnect cause code for use when the H.225 alerting (T301) timer expires. .
timeout t303
Q.850 call-disconnect cause code for use when the H.225 setup (T303) timer expires.
timeout t310
Q.850 call-disconnect cause code for use when the H.225 call-proceeding (T310) timer expires.
code-id
Q.850 code id number. Range: 1 to 127.
Command Default
No mapping occurs.
Command Modes
H.323-voice-service
Command History
Release
Modification
12.4(9)T
This command was introduced.
Usage Guidelines
Use this command to configure new Q.850 call-disconnect cause codes for use if an H.323 voice call fails during setup.
Examples
The following example causes a gateway to send the default ARJ cause code of 24 rather than the previous default of 63 when a call fails for reasons that are associated with the ARJ default cause code: Router(config)# voice service voip Router(conf-voi-serv)# h323 Router(conf-serv-h323)# scenario-cause arj-default 24
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Cisco IOS Voice Commands: S scenario-cause
Related Commands
Command
Description
h225 timeout call-proceeding
Sets the call-proceeding (T310, or call-setup to call-disconnect) disconnect timer.
map q850-cause
Maps a Q.850 call-disconnect cause code to a tone.
q850-cause
Maps a Q.850 call-disconnect cause code to a different Q.850 call-disconnect cause code.
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Cisco IOS Voice Commands: S secondary
secondary To set the backup location for storing call detail records (CDRs) if the primary location becomes unavailable, use the secondary command in gateway accounting file configuration mode. To reset to the default, use the no form of this command. secondary {ftp path/filename username username password password | ifs device:filename} no secondary {ftp | ifs}
Syntax Description
ftp path/filename
Name and location of the backup file on an external FTP server. Filename is limited to 25 characters.
ifs device:filename
Name and location of the backup file in flash memory or other internal file system on this router. Values depend on storage devices available on the router, for example flash or slot0. Filename is limited to 25 characters.
This command was integrated into Cisco IOS Release 12.4(20)T.
Usage Guidelines
This command defines the backup location where accounting records are sent if the file transfer to the primary device fails. The file accounting process retries the primary device, defined with the primary command, up to the number of times defined by the maximum retry-count command before automatically switching over to the secondary device. The secondary device is attempted only after the primary device fails after the defined number of retries. If the secondary device also fails, the system logs an error and the file accounting process stops. To manually switch back to the primary device when it becomes available, use the file-acct reset command. The system does not automatically switch back to the primary device. A syslog warning message is generated if flash becomes full. The filename you assign is appended with the gateway hostname and time stamp at the time the file is created to make the filename unique. For example, if you specify the filename cdrtest1 on a router with the hostname cme-2821, a file is created with the name cdrtest1.cme-2821.2007_10_28T22_21_41.000, where 2007_10_28T22_21_41.000 is the time that the file was created.
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Cisco IOS Voice Commands: S secondary
Limit the filename you assign with this command to 25 characters, otherwise it could be truncated when the accounting file is created because the full filename, including the appended hostname and timestamp, is limited to 63 characters.
Examples
The following example shows the backup location of the accounting file is set to flash:cdrtest2: gw-accounting file primary ftp server1/cdrtest1 username bob password temp secondary ifs flash:cdrtest2 maximum buffer-size 25 maximum retry-count 3 maximum fileclose-timer 720 cdr-format compact
Related Commands
Command
Description
file-acct reset
Manually switches back to the primary device for file accounting.
maximum retry-count Sets the maximum number of times the router attempts to connect to the primary file device before switching to the secondary device. primary
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Sets the primary location for storing the CDRs generated for file accounting.
Cisco IOS Voice Commands: S security
security To enable authentication and authorization on a gatekeeper, use the security command in gatekeeper configuration mode. To disable security, use the no form of this command. security {any | h323-id | e164} {password default password | password separator character} no security {any | h323-id | e164} {password default password | password separator character}
Syntax Description
any
Uses the first alias of an incoming registration, admission, and status (RAS) protocol registration, regardless of its type, to identify the user to RADIUS/TACACS+.
h323-id
Uses the first H.323 ID type alias to identify the user to RADIUS/TACACS+.
e164
Uses the first E.164 address type alias to identify the user to RADIUS/TACACS+.
password default password
Default password that the gatekeeper associates with endpoints when authenticating them with an authentication server. The password must be identical to the password on the authentication server.
password separator character
Character that endpoints use to separate the H.323-ID from the piggybacked password in the registration. Specifying this character allows each endpoint to supply a user-specific password. The separator character and password are stripped from the string before it is treated as an H.323-ID alias to be registered. Note that passwords may only be piggybacked in the H.323-ID, not the E.164 address, because the E.164 address allows a limited set of mostly numeric characters. If the endpoint does not wish to register an H.323-ID, it can still supply an H.323-ID consisting of just the separator character and password. This H.323-ID consisting of just the separator character and password are understood to be a password mechanism and no H.323-ID is registered.
Command Default
No default
Command Modes
Gatekeeper configuration
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco 2600 series and Cisco 3600 series.
Usage Guidelines
Use this command to enable identification of registered aliases by RADIUS/TACACS+. If the alias does not exist in RADIUS/TACACS+, the endpoint is not allowed to register. A RADIUS/TACACS+ server and encryption key must have been configured in Cisco IOS software for security to work.
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Cisco IOS Voice Commands: S security
Only the first alias of the proper type is identified. If no alias of the proper type is found, the registration is rejected. This command does not allow you to define the password mechanism unless the security type (h323-id or e164 or any) has been defined. Although the no security password command undefines the password mechanism, it leaves the security type unchanged, so security is still enabled. However, the no security command disables security entirely, including removing any existing password definitions.
Examples
The following example enables identification of registrations using the first H.323 ID found in any registration: security h323id
The following example enables security, authenticating all users by using their H.323-IDs and a password of qwerty2x: security h323-id security password qwerty2x
The next example enables security, authenticating all users by using their H.323-IDs and the password entered by the user in the H.323-ID alias he or she registers: security h323-id security password separator !
Now if a user registers with an H.323-ID of joe!024aqx, the gatekeeper authenticates user joe with password 024aqx, and if that is successful, registers the user with the H.323-ID of joe. If the exclamation point is not found, the user is authenticated with the default password, or a null password if no default has been configured. The following example enables security, authenticating all users by using their E.164 IDs and the password entered by the user in the H.323-ID alias he or she registers: security e164 security password separator !
Now if a user registers with an E.164 address of 5551212 and an H.323-ID of !hs8473q6, the gatekeeper authenticates user 5551212 and password hs8473q6. Because the H.323-ID string supplied by the user begins with the separator character, no H.323-ID is registered, and the user is known only by the E.164 address.
Related Commands
Command
Description
accounting (gatekeeper)
Enables the accounting security feature on the gatekeeper.
radius-server host
Specifies a RADIUS server host.
radius-server key
Sets the authentication and encryption key for all RADIUS communications between the router and the RADIUS daemon.
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Cisco IOS Voice Commands: S security acl
security acl To configure access-list based filtering on the gatekeeper, use the security acl command in gatekeeper configuration mode. To disable, use the no form of this command. security acl {answerarq| lrq} access-list-number no security acl {answerarq| lrq}
Syntax Description
answerarq
Filters incoming answer admission requests (AnswerARQ) using IP access-lists.
lrq
Filters incoming location requests (LRQs) using IP access-list.
access-list-number
Number of an access list that was configured using the access-list command. This is a decimal number from 1 to 99 or from 1300 to 1999. Only standard IP access lists numbered 1 through 99 are supported for the Tokenless Call Authorization feature.
Command Default
No default behavior or values.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.3(5)
This command was introduced.
Usage Guidelines
The security acl command configures the gatekeeper to use IP access lists for security. Use this command in conjunction with the access-list command to configure access-list based AnswerARQ and LRQ requests filtering on a gatekeeper. The gatekeeper will process only those requests which have been sent by sources that are permitted access by the specified IP access list. Requests sent by sources which have been denied by the specified IP access lists, will be rejected.
Examples
The following example shows how to configure a gatekeeper to use a previously configured IP access list with an IP access list number of 30 for call authorization: Router(config-gk)# security acl answerarq 30
The following example shows how to configure a gatekeeper to use a previously configured IP access list with an IP access list number of 20 for LRQ filtering: Router(config-gk)# security acl lrq 20
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Cisco IOS Voice Commands: S security acl
Related Commands
Command
Description
access-list
Configures the access list mechanism for filtering frames by protocol type or vendor code.
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Cisco IOS Voice Commands: S security izct
security izct To configure the gatekeeper to include the destination E.164 alias in the IZC token hash, use the security izct command in gatekeeper configuration mode. To not include destination E.16 alias in IZC token hash, use the no form of this command. security izct password {password [hash {dest-alias | src-alias | dest-csa | src-csa | dest-epid | src-epid}]} no security izct {password [hash {dest-alias | src-alias | dest-csa | src-csa | dest-epid | src-epid}]}
Syntax Description
password password
Specifies the password that the gatekeeper associates with endpoints when authenticating them with an authentication server. The password must be identical to the password on the authentication server.
hash
Specifies the options to be used in hash generation.
dest-alias
Specifies that the destination alias be included in hash generation.
src-alias
Specifies that the source alias be included in hash generation.
dest-csa
Specifies that the destination csa be included in hash generation.
src-csa
Specifies that the source alias be included in hash generation.
dest-epid
Specifies that the destination epid be included in hash generation.
src-epid
Specifies that the source epid be included in hash generation.
Command Default
Destination E.16 alias are not included in IZC token hash.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.3(5)
This command was introduced.
12.4(15)XZ
The dest-alias, src-alias, dest-csa, src-csa, dest-epid, and src-epid keywords were added.
12.4(20)T
This command was integrated into Cisco IOS Release 12.4(20)T.
Usage Guidelines
Configure the security izct command on the gatekeeper that generates the InterZone Clear Token (IZCT) hash to prevent rogue endpoints from sending an ARQ message with one called number and then changing the called number when they send the SETUP message to the terminating endpoint. When this command is configured, modification of the called number after the IZCT hash is generated by the trunking gateway will not be allowed. The IZCT token generated is valid only for 30 seconds and the IZCT hash token generated by terminating gatekeeper (TGK) can be used for multiple calls. The call is rejected if any intermediate entity, such as a Cisco Gatekeeper Transaction Message Protocol (GKTMP) server (on the originating gatekeeper) or the originating gateway (using number translation rules), tries to modify the called number after the token is prepared during address resolution.
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Cisco IOS Voice Commands: S security izct
•
The hash keyword at originating gateway (OGW) and TGK do not need to match.
•
More than one hash keyword can be configured for the security izct command.
The security izct command must be configured at OGK or TGK in order to enable the feature. When configuring an OGW to an OGK to a TGK and to a TGW. The security izct command is optional at the OGK, and required at the TGK. If hash parameter is not specified at the TGK, then dest-alias (default) will be used for hash token computation. The no version of this command the requires the keyword argument combinations as defined in the preceding command syntax table.
Examples
The following example prevents modification of the called number after the IZCT hash is generated by the trunking gateway: Router(config-gk)# security izct password example hash dest-alias
Related Commands
Command
Description
accounting (gatekeeper)
Enables the accounting security feature on the gatekeeper.
radius-server host
Specifies a RADIUS server host.
radius-server key
Sets the authentication and encryption key for all RADIUS communications between the router and the RADIUS daemon.
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Cisco IOS Voice Commands: S security mode
security mode To set the security mode for a specific dial peer using Skinny Client Control Protocol (SCCP) Telephony Control Application (STCAPP) services in a secure Cisco Unified CME network, use the security mode command in dial peer configuration mode. To return to the default, use the no form of this command. security mode {authenticated | none | encrypted | system} no security mode
Syntax Description
authenticated
Sets the security mode to authenticated and enables SCCP signaling between the voice gateway and Cisco Unified CME to take place through the secure TLS connection on TCP port 2443.
none
SCCP signaling is not secure.
encrypted
Sets the security mode to encrypted and enables SCCP signaling between the voice gateway and Cisco Unified CME to take place through Secure Real-Time Transport Protocol (SRTP).
system
Enables the security mode specified at the global level by the stcapp security mode command.
Command Default
Security mode specified at the global level is enabled.
Command Modes
Dial peer configuration (config-dialpeer)
Command History
Release
Modification
12.4(11)XW1
This command was introduced.
12.4(20)T
This command was integrated into Cisco IOS Release 12.4(20)T.
Usage Guidelines
Use this command to specify security mode on the voice gateway for Cisco Unified CME phone authentication and encryption. Set the SCCP signaling security mode globally using the stcapp security mode command in global configuration mode. If you use both the stcapp security mode and the security mode commands, the dial-peer level command, security mode, overrides the global setting.
Examples
The following example selects secure SCCP signaling in authenticated mode: Router(config)# dial-peer voice 1 pots Router(config-dialpeer)# security mode authenticated
The following example selects encrypted secure SCCP signaling and encryption through SRTP: Router(config)# dial-peer voice 2 pots Router(config-dialpeer)# security mode encrypted
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Cisco IOS Voice Commands: S security mode
Related Commands
Command
Description
stcapp security mode
Enables security for STCAPP endpoints and specifies the security mode to be used for setting up the TLS connection.
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Cisco IOS Voice Commands: S sequence-numbers
sequence-numbers To enable the generation of sequence numbers in each frame generated by the digital signal processor (DSP) for Voice over Frame Relay applications, use the sequence-numbers command in dial peer configuration mode. To disable the generation of sequence numbers, use the no form of this command. sequence-numbers no sequence-numbers
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled
Command Modes
Dial peer configuration
Command History
Release
Modification
12.0(3)XG
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.0(4)T
This command was integrated into Cisco IOS Release 12.0(4)T.
Usage Guidelines
Sequence numbers on voice packets allow the digital signal processor (DSP) at the playout side to detect lost packets, duplicate packets, or out-of-sequence packets. This helps the DSP to mask out occasional drop-outs in voice transmission at the cost of one extra byte per packet. The benefit of using sequence numbers versus the cost in bandwidth of adding an extra byte to each voice packet on the Frame Relay network must be weighed to determine whether to disable this function for your application. Another factor to consider is that this command does not affect codecs that require a sequence number, such as G.726. If you are using a codec that requires a sequence number, the DSP generates one regardless of the configuration of this command.
Examples
The following example disables generation of sequence numbers for VoFR frames for VoFR dial peer 200: dial-peer voice 200 vofr no sequence-numbers
Related Commands
Command
Description
called-number (dial peer)
Enables an incoming VoFR call leg to get bridged to the correct POTS call leg when using a static FRF.11 trunk connection.
codec (dial peer)
Specifies the voice coder rate of speech for a Voice over Frame Relay dial peer.
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Cisco IOS Voice Commands: S sequence-numbers
Command
Description
cptone
Specifies a regional analog voice interface-related tone, ring, and cadence setting.
destination-pattern
Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.
dtmf-relay (Voice over Frame Relay)
Enables the generation of FRF.11 Annex A frames for a dial peer.
session protocol (Voice over Frame Relay)
Establishes a session protocol for calls between the local and remote routers via the packet network.
session target
Specifies a network-specific address for a specified dial peer or destination gatekeeper.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
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Cisco IOS Voice Commands: S server (auto-config application)
server (auto-config application) To configure the IP address or name of the TFTP server for an auto-configuration application, use the server command in auto-config application configuration mode. To remove the IP address or name, use the no form of this command. server ip-address | domain-name [ip-address | domain-name] [ip-address | domain-name] no server
Syntax Description
ip-address
Specifies the IP address of the TFTP server.
domain-name
Specifies the domain name of the TFTP server.
Command Default
No default behavior or values.
Command Modes
Auto-config application configuration
Command History
Release
Modification
12.3(8)XY
This command was introduced on the Communication Media Module.
12.3(14)T
This command was integrated into Cisco IOS Release 12.3(14)T.
Examples
The following example shows the server command used to configure two TFTP servers for an auto-configuration application: Router(auto-config-app)# server 172.18.240.45 172.18.240.55
Related Commands
Command
Description
auto-config
Enables auto-configuration or enters auto-config application configuration mode for the Skinny Client Control Protocol (SCCP) application.
show auto-config
Displays the current status of auto-config applications.
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Cisco IOS Voice Commands: S server (presence)
server (presence) To specify the IP address of a presence server for sending presence requests from internal watchers to external presence entities, use the server command in presence configuration mode. To remove the server, use the no form of this command. server ip-address no server
Syntax Description
ip-address
Command Default
A remote presence server is not used.
Command Modes
Presence configuration (config-presence)
Command History
Release
Modification
12.4(11)XJ
This command was introduced.
12.4(15)T
This command was integrated into Cisco IOS Release 12.4(15)T.
IP address of the remote presence server.
Usage Guidelines
This command specifies the IP address of a presence server that handles presence requests when the watcher and presence entity (presentity) are not colocated. The router acts as the presence server and processes all presence requests and status notifications when a watcher and presentity are both internal. If a subscription request is for an external presentity, the request is sent to the remote server specified by this command.
Examples
The following example shows a presence server with IP address 10.10.10.1: Router(config)# presence Router(config-presence)# allow subscribe Router(config-presence)# server 10.10.10.1
Related Commands
Command
Description
allow subscribe
Allows internal watchers to monitor external presence entities (directory numbers).
allow watch
Allows a directory number on a phone registered to Cisco Unified CME to be watched in a presence service.
max-subscription
Sets the maximum number of concurrent watch sessions that are allowed.
show presence global
Displays configuration information about the presence service.
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Cisco IOS Voice Commands: S server (presence)
Command
Description
show presence subscription
Displays information about active presence subscriptions.
watcher all
Allows external watchers to monitor internal presence entities (directory numbers).
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Cisco IOS Voice Commands: S server (RLM)
server (RLM) To identify an RLM server, use the server RLM configuration command. To remove the identification, use the no form of this command server name-tag no server name-tag
Syntax Description
name-tag
Command Default
Disabled
Command Modes
RLM configuration
Command History
Release
Modification
11.3(7)
This command was introduced.
Name to identify the server configuration so that multiple entries of server configuration can be entered.
Usage Guidelines
Each server can have multiple entries of IP addresses or aliases.
Examples
The following example identifies the RLM server and defines the associated IP addresses: rlm group 1 server r1-server link address 10.1.4.1 source Loopback1 weight 4 link address 10.1.4.2 source Loopback2 weight 3
Related Commands
Command
Description
clear interface
Resets the hardware logic on an interface.
clear rlm group
Clears all RLM group time stamps to zero.
interface
Defines the IP addresses of the server, configures an interface type, and enters interface configuration mode.
link (RLM)
Specifies the link preference.
protocol rlm port
Reconfigures the port number for the basic RLM connection for the whole rlm-group.
retry keepalive
Allows consecutive keepalive failures a certain amount of time before the link is declared down.
show rlm group statistics
Displays the network latency of the RLM group.
show rlm group status
Displays the status of the RLM group.
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Cisco IOS Voice Commands: S server (RLM)
Command
Description
show rlm group timer
Displays the current RLM group timer values.
shutdown (RLM)
Shuts down all of the links under the RLM group.
timer
Overwrites the default setting of timeout values.
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Cisco IOS Voice Commands: S server absent reject
server absent reject To configure the gatekeeper to reject new registrations or calls when the connection to the Gatekeeper Transaction Message Protocol (GKTMP) server is down, use the server absent reject command in gatekeeper configuration mode. To disable, use the no form of this command. server absent reject {arq | rrq} no server absent reject {arq | rrq}
Syntax Description
arq
Reject call admission request (ARQ) messages.
rrq
Reject registration request (RRQ) messages.
Command Default
By default, registrations and calls are not rejected.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.2(11)T
This command was introduced on the Cisco 3660 and Cisco MC3810.
Usage Guidelines
Note
Examples
This command configures the gatekeeper to reject new registrations or calls when it is unable to reach the GKTMP server because the TCP connection between the gatekeeper and GKTMP server is down. If multiple GKTMP servers are configured, the gatekeeper tries all of them and rejects registrations or calls only if none of the servers respond. You can also use this feature for security or service denial if a connection with the server is required to complete a registration.
This command assumes that RRQ and ARQ triggers are used between the gatekeeper and GKTMP server.
The following example directs the gatekeeper to reject registrations when it cannot connect to the GKTMP server: Router# show gatekeeper configuration . . . h323id tet gw-type-prefix 1#* default-technology gw-type-prefix 9#* gw ipaddr 1.1.1.1 1720 no shutdown server absent reject rrq . . .
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Cisco IOS Voice Commands: S server flow-control
server flow-control To enable flow control on the Cisco IOS gatekeeper (GK) and reset all thresholds to default, use the server flow-control command in gatekeeper configuration mode. To disable GK flow control, use the no form of this command. server flow-control [onset value] [abatement value] [qcount value] no server flow-control
Syntax Description
onset value
(Optional) Percentage of the server timeout value that is used to mark the server as usable or unusable. Range is from 1 to 100. The default is 80.
abatement value
(Optional) Percentage of the server timeout value that is used to mark the server as unusable or usable. Range is from 1 to 100. The default is 50. Note
qcount value
The abatement value must be less than the onset value.
(Optional) Threshold length of the outbound queue on the GK. The queue contains messages waiting to be transmitted to the server. The TCP socket between the GK and Gatekeeper Transaction Message Protocol (GKTMP) server queues messages if it has too many to transmit. If the count of outbound queue length on the server reaches the qcount value, the server is marked unusable. Range is from 1 to 1000. The default is 400.
Command Default
The gatekeeper will send a maximum of 1000 RRQ messages.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T.
Usage Guidelines
Suppose the server timeout value is 3 seconds, the onset value is 50, and the abatement value is 40. When the average response time from the server to the Gatekeeper Transaction Message Protocol (GKTMP) reaches 1.5 seconds (the onset percentage of the server timeout value), the server is marked as unusable. During the period that the server is marked as unusable, REQUEST ALV messages are still sent to the unusable server. When the response time is lowered to 1.2 seconds (the abatement percentage of the timeout value), the server is marked usable again, and the GKTMP resumes sending messages to the server. When the server flow-control command is configured on its own the default is value 400. If you change one parameter using the server flow-control command, all other parameters revert to the default values. For example, if the onset is configured at 70 percent and you use the server flow-control command to set the abatement level, the onset resets to the default (80 percent).
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Cisco IOS Voice Commands: S server flow-control
Examples
The following example uses the command with the default values: Router# server flow-control
The following example enables the GKTMP Interface Resiliency Enhancement feature with an onset level of 50: Router# server flow-control onset 50 *Mar
8 20:05:34.081: gk_srv_handle_flowcontrol: Flow control enabled
Router# show running-config Building configuration... Current configuration : 1065 bytes ! version 12.2 no service single-slot-reload-enable service timestamps debug datetime msec service timestamps log uptime no service password-encryption ! hostname snet-3660-3 ! . . . gatekeeper zone local snet-3660-3 cisco.com zone remote snet-3660-2 cisco.com 209.165.200.225 1719 zone prefix snet-3660-2 408* lrq forward-queries no use-proxy snet-3660-3 default inbound-to terminal no use-proxy snet-3660-3 default outbound-from terminal no shutdown server registration-port 8000 server flow-control onset 50 ! . . . end
The following example enables the GKTMP Interface Resiliency Enhancement feature: Router# show gatekeeper status Gatekeeper State: UP Load Balancing: DISABLED Flow Control: ENABLED Zone Name: snet-3660-3 Accounting: DISABLED Endpoint Throttling: DISABLED Security: DISABLED Maximum Remote Bandwidth: unlimited Current Remote Bandwidth: 0 kbps Current Remote Bandwidth (w/ Alt GKs): 0 kbps
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VR-1700
Cisco IOS Voice Commands: S server flow-control
The following example shows the server statistics, including timeout encountered, average response time, and the server status: Router# show gatekeeper server GATEKEEPER SERVERS STATUS ========================= Gatekeeper Server listening port: 8250 Gatekeeper Server timeout value: 30 (100ms) GateKeeper GKTMP version: 3.1 Gatekeeper-ID: Gatekeeper1 -----------------------RRQ Priority: 5 Server-ID: Server43 Server IP address: 209.165.200.254:40118 Server type: dynamically registered Connection Status: active Trigger Information: Trigger unconditionally Server Statistics: REQUEST RRQ Sent=0 RESPONSE RRQ Received = 0 RESPONSE RCF Received = 0 RESPONSE RRJ Received = 0 Timeout encountered=0 Average response time(ms)=0 Server Usable=TRUE
Related Commands
Command
Description
timer server timeout
Specifies the timeout value for a response from a back-end GKTMP server.
Cisco IOS Voice Command Reference
VR-1701
Cisco IOS Voice Commands: S server registration-port
server registration-port To configure the listener port for the server to establish a connection with the gatekeeper, use the server registration-port command in gatekeeper configuration mode. To force the gatekeeper to close the listening socket so that no more new registration takes place, use the no form of this command. server registration-port port-number no server registration-port port-number
Syntax Description
Command Default
Note
port-number
Port number on which the gatekeeper listens for external server connections. Range is from 1 to 65535. There is no default.
No registration port is configured.
If the gatekeeper is to communicate with network servers, a registration port must be configured on it.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the following platforms: Cisco 2500 series, Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, and Cisco MC3810.
12.2(11)T
This command was implemented on the Cisco 3700 series.
Usage Guidelines
Note
Examples
Use this command to configure a server registration port to poll for servers that want to establish connections with the gatekeeper.
The no form of this command forces the gatekeeper on this router to close the listen socket, so it cannot accept more registrations. However, existing connections between the gatekeeper and servers are left open.
The following example establishes a listener port for a server connection with a gatekeeper: Router(config)# gatekeeper Router(config-gk)# server registration-port 20000
Cisco IOS Voice Command Reference
VR-1702
Cisco IOS Voice Commands: S server registration-port
Related Commands
Command
Description
server trigger
Configure static server triggers for specific RAS messages to be forwarded to a specified server.
show gatekeeper servers
Displays the triggers configured on the gatekeeper.
Cisco IOS Voice Command Reference
VR-1703
Cisco IOS Voice Commands: S server routing
server routing To specify the type of circuit messages sent to the Gatekeeper Transaction Message Protocol (GKTMP) server, use the server routing command in gatekeeper configuration mode. To return to the default, use the no form of this command. server routing {both | carrier | trunk-group} no server routing {both | carrier | trunk-group}
Syntax Description
both
Sends both types of information in GKTMP messages.
carrier
Sends only carrier information in GKTMP messages. This is the default.
trunk-group
Sends only trunk-group information in GKTMP messages.
Command Default
Carrier
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
Usage Guidelines
Use this command to route carrier and trunk-group messages from the gatekeeper to the GKTMP server. The carrier keyword sends the “I” and “J” tags in the GKTMP messages. The trunk-group keyword sends the “P” and “Q” tags in the GKTMP messages. The both keyword sends both sets of tags.
Examples
The following example enables trunk-group information to be sent in GKTMP messages from the gatekeeper: Router(config)# gatekeeper Router(config-gk)# server routing trunk-group
Related Commands
Command
Description
show gatekeeper servers
Displays the triggers configured on the gatekeeper.
Cisco IOS Voice Command Reference
VR-1704
Cisco IOS Voice Commands: S server trigger arq
server trigger arq To configure the admission request (ARQ) trigger statically on the gatekeeper, use the server trigger arq command in gatekeeper configuration mode. Submode commands are available after the server trigger arq command is entered. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the all form of this command. server trigger arq gkid priority server-id server-ip-address server-port no server trigger arq gkid priority server-id server-ip-address server-port no server trigger all
Syntax Description
all
Deletes all CLI-configured triggers.
gkid
Local gatekeeper identifier.
priority
Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.
server-id
ID number of the external application.
server-ip-address
IP address of the server.
server-port
Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.
Submode Commands
After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line. info-only
Use to indicate to the Cisco IOS gatekeeper that messages that meet the specified trigger parameters should be sent to the GKTMP server application as notifications only and that the gatekeeper should not wait for a response from the Gatekeeper Transaction Message Protocol (GKTMP) server application.
shutdown
Use to temporarily disables a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.
destination-info {e164 | email-id | h323-id} value
Use to send ARQ RAS messages containing a specified destination to the GKTMP server application. Configure one of the following conditions •
e164—Destination is an E.164 address.
•
email-id—Destination is an e-mail ID.
•
h323-id—Destination is an H.323 ID.
•
value—Value against which to compare the destination address in the RAS messages. For E.164 addresses, the following wildcards can be used: – A trailing series of periods, each of which represents a single
character. – A trailing asterisk, which represents one or more characters.
Cisco IOS Voice Command Reference
VR-1705
Cisco IOS Voice Commands: S server trigger arq
redirect-reason reason-number
Use to send ARQ RAS messages containing a specific redirect reason to the GKTMP server application. •
reason-number—Range is from 0 to 65535. Currently-used values are: – 0—Unknown reason. – 1—Call forwarding busy or called DTE busy. – 2—Call forwarded; no reply. – 4—Call deflection. – 9—Called DTE out of order. – 10—Call forwarding by the called DTE. – 15—Call forwarding unconditionally.
Command Default
No trigger servers are set.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(11)T
This command was implemented on the Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810. The irr trigger was added.
Usage Guidelines
Use this command and its optional submode commands to configure the admission request (ARQ) static server trigger. The gatekeeper checks incoming gateway ARQ messages for the configured trigger information. If an incoming ARQ message contains the specified trigger information, the gatekeeper sends the ARQ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the ARQ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application. If no submode commands are configured for the ARQ messages, the gatekeeper sends all ARQ messages to the GKTMP server application. If the gatekeeper receives an ARQ trigger registration message that contains several trigger conditions, the conditions are treated as “OR” conditions. In other words, if an incoming ARQ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server. If the gatekeeper receives two ARQ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two ARQ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming ARQ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one ARQ trigger registration message with the same priority but for different GKTMP servers, the gatekeeper retains all of the registrations. The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.
Cisco IOS Voice Command Reference
VR-1706
Cisco IOS Voice Commands: S server trigger arq
Examples
The following example configures a trigger registration on gatekeeper “sj.xyz.com” to send all ARQ messages to GKTMP server “Server-123”: Router(config-gk)# server trigger arq sj.xyz.com 1 Server-123 1.14.93.130 1751 Router(config-gk_arqtrigger)# exit
The following example configures an ARQ trigger registration on gatekeeper “alpha”, which sends to GKTMP server “Server-west” any ARQ message that contains H.323 ID “3660-gw1”, e-mail ID “joe.xyz.com”, or a redirect reason 1. All other ARQ messages are not sent to the GKTMP server application. Router(config-gk)# server trigger arq alpha 1 Server-west 10.10.10.10 1751 Router(config-gk-arqtrigger)# destination-info h323-id 3660-gw1 Router(config-gk-arqtrigger)# destination-info email-id joe.xyz.com Router(config-gk-arqtrigger)# redirect-reason 1 Router(config-gk-arqtrigger# exit
If the ARQ registration message defined above for gatekeeper “alpha” is configured and the gatekeeper receives the following trigger registration: Router(config-gk)# server trigger arq alpha 2 Server-west 10.10.10.10 1751 Router(config-gk_arqtrigger)# destination-info e164 1800.... Router(config-gk_arqtrigger)# exit
Then gatekeeper “alpha” checks all incoming ARQ messages for the destination H.323 ID, e-mail ID, or redirect reason before checking for the E.164 address 1800 (for example, 18005551212). If any one of those conditions is met, the gatekeeper sends the ARQ message to the GKTMP server “Server-west”. If the second gatekeeper “alpha” ARQ trigger registration had been defined with a priority 1 instead of priority 2, the second server trigger definition would have overridden the first one. In other words, the gatekeeper “alpha” would send to GKTMP server “Server-west” only those ARQ messages that contain a destination E.164 address that starts with 1800. All other ARQ messages would not be sent to the GKTMP server.
Related Commands
Command
Description
server registration-port
Configures the server listening port on the gatekeeper.
show gatekeeper servers
Displays the triggers configured on the gatekeeper.
Cisco IOS Voice Command Reference
VR-1707
Cisco IOS Voice Commands: S server trigger brq
server trigger brq To configure the bandwidth request (BRQ) trigger statically on the gatekeeper, use the server trigger brq command in gatekeeper configuration mode. Submode commands are available after entering the server trigger brq command. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the all form of the command. server trigger brq gkid priority server-id server-ip-address server-port no server trigger brq gkid priority server-id server-ip-address server-port no server trigger all
Syntax Description
all
Deletes all CLI-configured triggers.
gkid
Local gatekeeper identifier.
priority
Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.
server-id
ID number of the external application.
server-ip-address
IP address of the server.
server-port
Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.
Submode Commands
After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line. info-only
Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol (GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.
redirect-reason reason-number
Use to send BRQ RAS messages containing a specific redirect reason to the GKTMP server application. •
reason-number—Range is from 0 to 65535. Currently used values are as follows: – 0—Unknown reason. – 1—Call forwarding busy or called DTE busy. – 2—Call forwarded; no reply. – 4—Call deflection. – 9—Called DTE out of order. – 10—Call forwarding by the called DTE. – 15—Call forwarding unconditionally.
shutdown
Cisco IOS Voice Command Reference
VR-1708
Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.
Cisco IOS Voice Commands: S server trigger brq
Command Default
No trigger servers are set.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(11)T
This the command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810. The irr trigger was added.
Usage Guidelines
Use this command and its optional submode commands to configure the bandwidth request (BRQ) static server trigger. The gatekeeper checks incoming gateway BRQ messages for the configured trigger information. If an incoming BRQ message contains the specified trigger information, the gatekeeper sends the BRQ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the BRQ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application. If no submode commands are configured for the BRQ messages, the gatekeeper sends all BRQ messages to the GKTMP server application. If the gatekeeper receives BRQ trigger registration message that contains several trigger conditions, the conditions are treated as “OR” conditions. In other words, if an incoming BRQ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server. If the gatekeeper receives two BRQ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two BRQ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming BRQ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one BRQ trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations. The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.
Examples
The following example configures a trigger registration on gatekeeper “sj.xyz.com” to send all BRQ messages to GKTMP server “Server-123”: Router(config-gk)# server trigger brq sj.xyz.com 1 Server-123 1.14.93.130 1751 Router(config-gk_brqtrigger)# exit
The following example configures BRQ trigger registration on gatekeeper “alpha”, which sends to GKTMP server “Server-west” any BRQ message containing redirect reason 1 or redirect reason 2. All other BRQ messages are not sent to the GKTMP server application. Router(config-gk)# server trigger brq alpha 1 Server-west 10.10.10.10 1751 Router(config-gk-brqtrigger)# redirect-reason 1 Router(config-gk-brqtrigger)# redirect-reason 2 Router(config-gk-brqtrigger# exit
Cisco IOS Voice Command Reference
VR-1709
Cisco IOS Voice Commands: S server trigger brq
If the BRQ registration message defined above for gatekeeper “alpha” is configured and the gatekeeper receives the following trigger registration: Router(config-gk)# server trigger brq alpha 2 Server-west 10.10.10.10 1751 Router(config-gk_brqtrigger)# redirect-reason 10 Router(config-gk_brqtrigger)# exit
Then gatekeeper “alpha” checks all incoming BRQ messages for redirect reasons 1 or 2 before checking for redirect reason 10. If any one of those conditions is met, the gatekeeper sends the BRQ message to the GKTMP server “Server-west”. If the second gatekeeper “alpha” BRQ trigger registration had been defined with a priority 1 instead of priority 2, then the second server trigger definition would have overridden the first one. In other words, the gatekeeper “alpha” would send to GKTMP server “Server-west” only those BRQ messages that contain a redirect reason 10. All other BRQ messages would not be sent to the GKTMP server.
Related Commands
Command
Description
server registration-port
Configures the server listening port on the gatekeeper.
show gatekeeper servers
Displays the triggers configured on the gatekeeper.
Cisco IOS Voice Command Reference
VR-1710
Cisco IOS Voice Commands: S server trigger drq
server trigger drq To configure the disengage request (DRQ) trigger statically on the gatekeeper, use the server trigger drq command in gatekeeper configuration mode. Submode commands are available after entering the server trigger drq command. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the all form of the command. server trigger drq gkid priority server-id server-ip-address server-port no server trigger drq gkid priority server-id server-ip-address server-port no server trigger all
Syntax Description
all
Deletes all CLI-configured triggers.
gkid
Local gatekeeper identifier.
priority
Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.
server-id
ID number of the external application.
server-ip-address
IP address of the server.
server-port
Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.
Submode Commands
After the command is entered, the software enters a submode that permits you to configure additional filters on the Reliability, Availability, and Serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line. info-only
Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol (GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.
destination-info {e164 | email-id | h323-id} value
Use to send automatic repeat request (ARQ) RAS messages containing a specified destination to the GKTMP server application. Configure one of the following conditions: •
e164—Destination is an E.164 address.
•
email-id—Destination is an e-mail ID.
•
h323-id—Destination is an H.323 ID.
•
value—Value against which to compare the destination address in the RAS messages. For E.164 addresses, the following wildcards can be used: – A trailing series of periods, each of which represents a single
character. – A trailing asterisk, which represents one or more characters.
Cisco IOS Voice Command Reference
VR-1711
Cisco IOS Voice Commands: S server trigger drq
call-info-type { fax | modem | voice}
shutdown
Use to send ARQ RAS messages containing a specified call info type to the GKTMP server application. The following types can be used: •
fax
•
modem
•
voice
Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.
Command Default
No trigger servers are set.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(11)T
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810.
12.4(4)T
The call-info-type submode command was added.
Usage Guidelines
Use this command and its optional submode commands to configure the disengage request (DRQ) static server trigger. The gatekeeper checks incoming gateway DRQ messages for the configured trigger information. If an incoming DRQ message contains the specified trigger information, the gatekeeper sends the DRQ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the DRQ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application. If no submode commands are configured for the DRQ messages, the gatekeeper sends all DRQ messages to the GKTMP server application. If the gatekeeper receives a DRQ trigger registration message that contains several trigger conditions, the conditions are treated as “OR” conditions. In other words, if an incoming DRQ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server. If the gatekeeper receives two DRQ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two DRQ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming DRQ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one DRQ trigger registration message with the same priority but for different GKTMP servers, the gatekeeper retains all of the registrations. The no form of the command removes the trigger definition from the Cisco IOS gatekeeper together with all statically configured conditions under that trigger.
Cisco IOS Voice Command Reference
VR-1712
Cisco IOS Voice Commands: S server trigger drq
Examples
The following example configures a trigger registration on gatekeeper “sj.xyz.com” to send all DRQ messages to GKTMP server “Server-123”: Router(config-gk)# server trigger drq sj.xyz.com 1 Server-123 1.14.93.130 1751 Router(config-gk_drqtrigger)# exit
The following example configures DRQ trigger registration on gatekeeper “alpha”, which sends to GKTMP server “Server-west” any DRQ message containing an H.323 ID “3660-gw1” or e-mail ID “joe.xyz.com”. All other DRQ messages are not sent to the GKTMP server application. Router(config-gk)# server trigger drq alpha 1 Server-west 10.10.10.10 1751 Router(config-gk-drqtrigger)# destination-info h323-id 3660-gw1 Router(config-gk-drqtrigger)# destination-info email-id joe.xyz.com Router(config-gk-drqtrigger# exit
If the DRQ registration message defined above for gatekeeper “alpha” is configured and the gatekeeper receives the following trigger registration: Router(config-gk)# server trigger drq alpha 2 Server-west 10.10.10.10 1751 Router(config-gk_drqtrigger)# destination-info e164 1800.... Router(config-gk_drqtrigger)# exit
then gatekeeper “alpha” checks all incoming DRQ messages for the destination H.323 ID or e-mail ID before checking for the E.164 address 1800 (for example, 18005551212). If any one of those conditions is met, the gatekeeper sends the DRQ message to the GKTMP server “Server-west”. If the second gatekeeper “alpha” DRQ trigger registration had been defined with a priority 1 instead of priority 2, then the second trigger registration would have overridden the first one. In other words, the gatekeeper “alpha” would send to GKTMP server Server-west only those DRQ messages that contain a destination E.164 address starting with 1800. All other DRQ messages would not be sent to the GKTMP server.
Related Commands
Command
Description
server registration-port
Configures the server listening port on the gatekeeper.
show gatekeeper servers
Displays the triggers configured on the gatekeeper.
Cisco IOS Voice Command Reference
VR-1713
Cisco IOS Voice Commands: S server trigger irr
server trigger irr To configure the information request response (IRR) trigger statically on the gatekeeper, use the server trigger irr command in gatekeeper configuration mode. Submode commands are available after entering the server trigger irr command. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the all form of the command. server trigger irr gkid priority server-id server-ip-address server-port no server trigger irr gkid priority server-id server-ip-address server-port no server trigger all
Syntax Description
all
Deletes all CLI-configured triggers.
gkid
Local gatekeeper identifier.
priority
Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.
server-id
ID number of the external application.
server-ip-address
IP address of the server.
server-port
Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.
Submode Commands
After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line. destination-info {e164 | email-id | h323-id} value
Use to send IRR RAS messages containing a specified destination to the GKTMP server application. Configure one of the following conditions: •
e164—Destination is an E.164 address.
•
email-id—Destination is an e-mail ID.
•
h323-id—Destination is an H.323 ID. – value—Value against which to compare the destination address in
the RAS messages. For E.164 addresses, the following wildcards can be used: – A trailing series of periods, each of which represents a single
character. – A trailing asterisk, which represents one or more characters.
info-only
Cisco IOS Voice Command Reference
VR-1714
Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol (GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.
Cisco IOS Voice Commands: S server trigger irr
redirect-reason reason-number
Use to send IRR RAS messages containing a specific redirect reason to the GKTMP server application. •
reason-number—Range is from 0 to 65535. Currently used values are the following: – 0—Unknown reason. – 1—Call forwarding busy or called DTE busy. – 2—Call forwarded; no reply. – 4—Call deflection. – 9—Called DTE out of order. – 10—Call forwarding by the called DTE. – 15—Call forwarding unconditionally.
shutdown
Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.
Command Default
No trigger servers are set.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(11)T
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810. The irr trigger was added.
Usage Guidelines
Use this command and its optional submode commands to configure the information request response (IRR) static server trigger. The gatekeeper checks incoming gateway IRR messages for the configured trigger information. If an incoming IRR message contains the specified trigger information, the gatekeeper sends the IRR message to the GKTMP server application. In addition, the IRR message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application. If no submode commands are configured for the IRR messages, the gatekeeper sends all IRR messages to the GKTMP server application. If the gatekeeper receives an IRR trigger registration message that contains several trigger conditions, the conditions are treated as “OR” conditions. In other words, if an incoming IRR RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server. If the gatekeeper receives two IRR trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two IRR trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming IRR messages against the conditions on the higher priority registration
Cisco IOS Voice Command Reference
VR-1715
Cisco IOS Voice Commands: S server trigger irr
before using the lower priority registration. If the gatekeeper receives more than one IRR trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations. The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.
Examples
The following example configures a trigger registration on gatekeeper “sj.xyz.com” to send all IRR messages to GKTMP server “Server-123”: Router(config-gk)# server trigger irr sj.xyz.com 1 Server-123 1.14.93.130 1751 Router(config-gk_irrtrigger)# exit
The following example configures an IRR trigger registration on gatekeeper “alpha”, which send to GKTMP server “Server-west” any IRR message containing an H.323 ID “3660-gw1”, e-mail ID “joe.xyz.com, or a redirect reason 1. All other IRR messages are not sent to the GKTMP server application. Router(config-gk)# server trigger irr alpha 1 Server-west 10.10.10.10 1751 Router(config-gk-irrtrigger)# destination-info h323-id 3660-gw1 Router(config-gk-irrtrigger)# destination-info email-id joe.xyz.com Router(config-gk-irrtrigger)# redirect-reason 1 Router(config-gk-irrtrigger# exit
If the IRR registration message defined above for gatekeeper “alpha” is configured and the gatekeeper receives the following trigger registration: Router(config-gk)# server trigger irr alpha 2 Server-west 10.10.10.10 1751 Router(config-gk_irrtrigger)# destination-info e164 1800.... Router(config-gk_irrtrigger)# exit
Then gatekeeper “alpha” checks all incoming IRR messages for the destination H.323 ID, e-mail ID, or redirect reason before checking for the E.164 address 1800 (for example, 18005551212). If any one of those conditions is met, the gatekeeper sends the IRR message to the GKTMP server “Server-west”. If the second gatekeeper “alpha” IRR trigger registration had been defined with a priority 1 instead of priority 2, then the second server trigger definition would have overridden the first one. In other words, the gatekeeper “alpha” would send to GKTMP server “Server-west” only those IRR messages that contain a destination E.164 address starting with 1800. All other IRR messages would not be sent to the GKTMP server.
Related Commands
Command
Description
server registration-port
Configures the server listening port on the gatekeeper.
show gatekeeper servers
Displays the triggers configured on the gatekeeper.
Cisco IOS Voice Command Reference
VR-1716
Cisco IOS Voice Commands: S server trigger lcf
server trigger lcf To configure the location confirm (LCF) trigger statically on the gatekeeper, use the server trigger lcf command in gatekeeper configuration mode. Submode commands are available after entering the server trigger lcf command. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the all form of the command. server trigger lcf gkid priority server-id server-ip-address server-port no server trigger lcf gkid priority server-id server-ip-address server-port no server trigger all
Syntax Description
all
Deletes all CLI-configured triggers.
gkid
Local gatekeeper identifier.
priority
Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.
server-id
ID number of the external application.
server-ip-address
IP address of the server.
server-port
Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.
Submode Commands
After the command is entered, the software enters a submode that permits you to configure additional filters on the RAS message. These filters are optional, and you may configure any of them, one per command line. destination-info {e164 | email-id | h323-id} value
Use to send LCF RAS messages containing a specified destination to the GKTMP server application. Configure one of the following conditions: •
e164—Destination is an E.164 address.
•
email-id—Destination is an e-mail ID.
•
h323-id—Destination is an H.323 ID. – value—Value against which to compare the destination address in
the RAS messages. For E.164 addresses, the following wildcards can be used:A trailing series of periods, each of which represents a single character. – A trailing asterisk, which represents one or more characters.
info-only
Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol (GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.
Cisco IOS Voice Command Reference
VR-1717
Cisco IOS Voice Commands: S server trigger lcf
remote-ext-address e164 value
Use to send LCF RAS messages that contain a specified remote extension address to the GKTMP server application. •
e164—Remote extension address is an E.164 address.
•
value—Value against which to compare the destination address in the RAS messages. The following wildcards can be used: – A trailing series of periods, each of which represents a single
character. – A trailing asterisk, which represents one or more characters.
shutdown
Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.
Command Default
No trigger servers are set.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(11)T
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810. The irr trigger was added.
Usage Guidelines
Use this command and its optional submode commands to configure the location confirm (LCF) static server trigger. The gatekeeper checks incoming gateway LCF messages for the configured trigger information. If an incoming LCF message contains the specified trigger information, the gatekeeper sends the LCF message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the LCF message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application. If no submode commands are configured for the LCF messages, the gatekeeper sends all LCF messages to the GKTMP server application. If the gatekeeper receives an LCF trigger registration message that contains several trigger conditions, the conditions are treated as “OR” conditions. In other words, if an incoming LCF RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server. If the gatekeeper receives two LCF trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two LCF trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming LCF messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one LCF trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations. The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.
Cisco IOS Voice Command Reference
VR-1718
Cisco IOS Voice Commands: S server trigger lcf
Examples
The following example configures a trigger registration on gatekeeper “sj.xyz.com” to send all LCF messages to GKTMP server “Server-123”: Router(config-gk)# server trigger lcf sj.xyz.com 1 Server-123 1.14.93.130 1751 Router(config-gk_lcftrigger)# exit
The following example configures an LCF trigger registration on gatekeeper “alpha”, which send to GKTMP server “Server-west” any LCF message containing an H.323 ID “3660-gw1”, e-mail ID joe.xyz.com, or a remote extension address starting with 1408. All other LCF messages are not sent to the GKTMP server application. Router(config-gk)# server trigger lcf alpha 1 Server-west 10.10.10.10 1751 Router(config-gk-lcftrigger)# destination-info h323-id 3660-gw1 Router(config-gk-lcftrigger)# destination-info email-id joe.xyz.com Router(config-gk-lcftrigger)# remote-ext-address e164 1408.... Router(config-gk-lcftrigger# exit
If the LCF registration message defined above for gatekeeper “alpha” is configured and the gatekeeper receives the following trigger registration: Router(config-gk)# server trigger lcf alpha 2 Server-west 10.10.10.10 1751 Router(config-gk_lcftrigger)# remote-ext-address e164 1800.... Router(config-gk_lcftrigger)# exit
then gatekeeper “alpha” checks all incoming LCF messages for the destination H.323 ID, e-mail ID, or remote extension address 1408 before checking for the remote extension address 1800 (for example, 18005551212). If any one of those conditions is met, the gatekeeper sends the LCF message to the GKTMP server “Server-west”. If the second gatekeeper “alpha” LCF trigger registration had been defined with a priority 1 instead of priority 2, then the second trigger registration would have overridden the first one. In other words, the gatekeeper “alpha” would send to GKTMP server “Server-west” only those LCF messages that contain a remote extension address E.164 address starting with 1800. All other LCF messages would not be sent to the GKTMP server.
Related Commands
Command
Description
server registration-port
Configures the server listening port on the gatekeeper.
show gatekeeper servers
Displays the triggers configured on the gatekeeper.
Cisco IOS Voice Command Reference
VR-1719
Cisco IOS Voice Commands: S server trigger lrj
server trigger lrj To configure the location reject (LRJ) trigger statically on the gatekeeper, use the server trigger lrj command in gatekeeper configuration mode. Submode commands are available after entering the server trigger lrj command. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the all form of the command. server trigger lrj gkid priority server-id server-ip-address server-port no server trigger lrj gkid priority server-id server-ip-address server-port no server trigger all
Syntax Description
all
Deletes all CLI-configured triggers.
gkid
Local gatekeeper identifier.
priority
Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.
server-id
ID number of the external application.
server-ip-address
IP address of the server.
server-port
Port on which the gatekeeper listens for messages from the external server connection.
Submode Commands
After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line. destination-info {e164 | email-id | h323-id} value
Use to send LRJ RAS messages containing a specified destination to the GKTMP server application. Configure one of the following conditions: •
e164—Destination is an E.164 address.
•
email-id—Destination is an e-mail ID.
•
h323-id—Destination is an H.323 ID.
•
value—Value against which to compare the destination address in the RAS messages. For E.164 addresses, the following wildcards can be used: – A trailing series of periods, each of which represents a single
character. – A trailing asterisk, which represents one or more characters.
info-only
Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol (GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.
shutdown
Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.
Cisco IOS Voice Command Reference
VR-1720
Cisco IOS Voice Commands: S server trigger lrj
Command Default
No trigger servers are set.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(11)T
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810.
Usage Guidelines
Use this command and its optional submode commands to configure the location reject (LRJ) static server trigger. The gatekeeper checks incoming gateway LRJ messages for the configured trigger information. If an incoming LRJ message contains the specified trigger information, the gatekeeper sends the LRJ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the LRJ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application. If no submode commands are configured for the LRJ messages, the gatekeeper sends all LRJ messages to the GKTMP server application. If the gatekeeper receives an LRJ trigger registration message that contains several trigger conditions, the conditions are treated as “OR” conditions. In other words, if an incoming LRJ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server. If the gatekeeper receives two LRJ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two LRJ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming LRJ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one LRJ trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations. The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.
Examples
The following example configures a trigger registration on gatekeeper “sj.xyz.com” to send all LRJ messages to GKTMP server “Server-123”: Router(config-gk)# server trigger lrj sj.xyz.com 1 Server-123 1.14.93.130 1751 Router(config-gk_lrjtrigger)# exit
The following example configures an LRJ trigger registration on gatekeeper “alpha”, which send to GKTMP server “Server-west” any LRJ message containing an H.323 ID “3660-gw1” or e-mail ID joe.xyz.com. All other LRJ messages are not sent to the GKTMP server application. Router(config-gk)# server trigger lrj alpha 1 Server-west 10.10.10.10 1751 Router(config-gk-lrjtrigger)# destination-info h323-id 3660-gw1 Router(config-gk-lrjtrigger)# destination-info email-id joe.xyz.com Router(config-gk-lrjtrigger# exit
Cisco IOS Voice Command Reference
VR-1721
Cisco IOS Voice Commands: S server trigger lrj
If the LRJ registration message defined above for gatekeeper “alpha” is configured and the gatekeeper receives the following trigger registration: Router(config-gk)# server trigger lrj alpha 2 Server-west 10.10.10.10 1751 Router(config-gk_lrjtrigger)# destination-info e164 1800.... Router(config-gk_lrjtrigger)# exit
then gatekeeper “alpha” checks all incoming LRJ messages for the destination H.323 ID or email ID before checking for the E.164 address 1800 (for example, 18005551212). If any one of those conditions is met, the gatekeeper sends the LRJ message to the GKTMP server “Server-west”. If the second gatekeeper “alpha” LRJ trigger registration had been defined with a priority 1 instead of priority 2, then the second trigger registration would have overridden the first one. In other words, the gatekeeper “alpha” would send to GKTMP server “Server-west” only those LRJ messages that contain a destination E.164 address starting with 1800. All other LRJ messages would not be sent to the GKTMP server.
Related Commands
Command
Description
server registration-port
Configures the server listening port on the gatekeeper.
show gatekeeper servers
Displays the triggers configured on the gatekeeper.
Cisco IOS Voice Command Reference
VR-1722
Cisco IOS Voice Commands: S server trigger lrq
server trigger lrq To configure the location request (LRQ) trigger statically on the gatekeeper, use the server trigger lrq command in gatekeeper configuration mode. Submode commands are available after entering the server trigger lrq command. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the all form of the command. server trigger lrq gkid priority server-id server-ip-address server-port no server trigger lrq gkid priority server-id server-ip-address server-port no server trigger all
Syntax Description
all
Deletes all CLI-configured triggers.
gkid
Local gatekeeper identifier.
priority
Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.
server-id
ID number of the external application.
server-ip-address
IP address of the server.
server-port
Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.
Submode Commands
After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line. destination-info {e164 | email-id | h323-id} value
Use to send LRQ RAS messages containing a specified destination to the GKTMP server application. Configure one of the following conditions: •
e164—Destination is an E.164 address.
•
email-id—Destination is an e-mail ID.
•
h323-id—Destination is an H.323 ID.
•
value—Value against which to compare the destination address in the RAS messages. For E.164 addresses, the following wildcards can be used: – A trailing series of periods, each of which represents a single
character. – A trailing asterisk, which represents one or more characters.
info-only
Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol (GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.
Cisco IOS Voice Command Reference
VR-1723
Cisco IOS Voice Commands: S server trigger lrq
redirect-reason reason-number
Use to send LRQ RAS messages containing a specific redirect reason to the GKTMP server application. •
reason-number—Range is from 0 to 65535. Currently used values are the following: – 0—Unknown reason. – 1—Call forwarding busy or called DTE busy. – 2—Call forwarded; no reply. – 4—Call deflection. – 9—Called DTE out of order. – 10—Call forwarding by the called DTE. – 15—Call forwarding unconditionally.
shutdown
Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.
Command Default
No trigger servers are set.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(11)T
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810.
Usage Guidelines
Use this command and its optional submode commands to configure the location request (LRQ) static server trigger. The gatekeeper checks incoming gateway LRQ messages for the configured trigger information. If an incoming LRQ message contains the specified trigger information, the gatekeeper sends the LRQ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the LRQ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application. If no submode commands are configured for the LRQ messages, the gatekeeper sends all LRQ messages to the GKTMP server application. If the gatekeeper receives an LRQ trigger registration message that contains several trigger conditions, the conditions are treated as “OR” conditions. In other words, if an incoming LRQ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server. If the gatekeeper receives two LRQ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two LRQ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming LRQ messages against the conditions on the higher priority registration
Cisco IOS Voice Command Reference
VR-1724
Cisco IOS Voice Commands: S server trigger lrq
before using the lower priority registration. If the gatekeeper receives more than one LRQ trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations. The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.
Examples
The following example configures a trigger registration on gatekeeper “sj.xyz.com” to send all LRQ messages to GKTMP server “Server-123”: Router(config-gk)# server trigger lrq sj.xyz.com 1 Server-123 1.14.93.130 1751 Router(config-gk_lrqtrigger)# exit
The following example configures an LRQ trigger registration on gatekeeper “alpha”, which sends to GKTMP server “Server-west” any LRQ message containing an H.323 ID “3660-gw1”, e-mail ID joe.xyz.com, or a redirect reason 1. Other LRQ messages are not sent to the GKTMP server application. Router(config-gk)# server trigger lrq alpha 1 Server-west 10.10.10.10 1751 Router(config-gk-lrqtrigger)# destination-info h323-id 3660-gw1 Router(config-gk-lrqtrigger)# destination-info email-id joe.xyz.com Router(config-gk-lrqtrigger)# redirect-reason 1 Router(config-gk-lrqtrigger# exit
If the LRQ registration message defined above for gatekeeper “alpha” is configured and the gatekeeper receives the following trigger registration: Router(config-gk)# server trigger lrq alpha 2 Server-west 10.10.10.10 1751 Router(config-gk_lrqtrigger)# destination-info e164 1800.... Router(config-gk_lrqtrigger)# exit
then gatekeeper “alpha” checks all incoming LRQ messages for the destination H.323 ID, email ID, or redirect reason before checking for the E.164 address 1800 (for example, 18005551212). If any one of those conditions is met, the gatekeeper sends the LRQ message to the GKTMP server “Server-west”. If the second gatekeeper “alpha” LRQ trigger registration had been defined with a priority 1 instead of priority 2, then the second server trigger definition would have overridden the first one. In other words, the gatekeeper “alpha” would send to GKTMP server “Server-west” only those LRQ messages that contain a destination E.164 address starting with 1800. All other LRQ messages would not be sent to the GKTMP server.
Related Commands
Command
Description
server registration-port
Configures the server listening port on the gatekeeper.
show gatekeeper servers
Displays the triggers configured on the gatekeeper.
Cisco IOS Voice Command Reference
VR-1725
Cisco IOS Voice Commands: S server trigger rai
server trigger rai To configure the resources available indicator (RAI) trigger statically on the gatekeeper, use the server trigger rai command in gatekeeper configuration mode. Submode commands are available after entering the server trigger rai command. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the all form of the command. server trigger rai gkid priority server-id server-ip-address server-port no server trigger rai gkid priority server-id server-ip-address server-port no server trigger all
Syntax Description
all
Deletes all CLI-configured triggers.
gkid
Local gatekeeper identifier.
priority
Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.
server-id
ID number of the external application.
server-ip-address
IP address of the server.
server-port
Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.
Submode Commands
After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line. endpoint-type value
Use to send RAI RAS messages that contain a particular endpoint type to the GKTMP server application. •
value—Value against which to compare the endpoint type in the RAS messages. Valid endpoint types are the following: – gatekeeper—Endpoint is an H.323 gatekeeper. – h320-gateway—Endpoint is an H.320 gateway. – mcu—Endpoint is a multipoint control unit (MCU). – other-gateway—Endpoint is another type of gateway not specified
on this list. – proxy—Endpoint is an H.323 proxy. – terminal—Endpoint is an H.323 terminal. – voice-gateway—Endpoint is a voice gateway.
info-only
Cisco IOS Voice Command Reference
VR-1726
Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol (GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.
Cisco IOS Voice Commands: S server trigger rai
shutdown
Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.
supported-prefix value Use to send RAI RAS messages that contain a specific supported prefix to the GKTMP server application. •
value—Value against which to compare the supported prefix in the RAS messages. The possible values are any E.164 pattern used as a gateway technology prefix. The value string can contain any of the following: 0123456789#*.
Command Default
No trigger servers are set.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(11)T
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810. The irr trigger was added.
Usage Guidelines
Use this command and its optional submode commands to configure the resources available indicator (RAI) static server trigger. The gatekeeper checks incoming gateway RAI messages for the configured trigger information. If an incoming RAI message contains the specified trigger information, the gatekeeper sends the RAI message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the RAI message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application. If no submode commands are configured for the RAI messages, the gatekeeper sends all RAI messages to the GKTMP server application. If the gatekeeper receives an RAI trigger registration message that contains several trigger conditions, the conditions are treated as “OR” conditions. In other words, if an incoming RAI RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server. If the gatekeeper receives two RAI trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two RAI trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming RAI messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one RAI trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations. The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.
Cisco IOS Voice Command Reference
VR-1727
Cisco IOS Voice Commands: S server trigger rai
Examples
The following example configures a trigger registration on gatekeeper “sj.xyz.com” to send all RAI messages to GKTMP server “Server-123”: Router(config-gk)# server trigger rai sj.xyz.com 1 Server-123 1.14.93.130 1751 Router(config-gk_raitrigger)# exit
The following example configures an RAI trigger registration on gatekeeper “alpha”, which sends to the GKTMP server “Server-west” any RAI message that contain an MCU endpoint, an H.323 proxy endpoint, or a supported prefix 1#. All other RAI messages are not sent to the GKTMP server. Router(config-gk)# server trigger rai alpha 1 Server-west 10.10.10.10 1751 Router(config-gk-raitrigger)# endpoint-type mcu Router(config-gk-raitrigger)# endpoint-type proxy Router(config-gk-raitrigger)# supported-prefix 1# Router(config-gk-raitrigger# exit
If the RAI registration message defined above for gatekeeper “alpha” is configured and the gatekeeper receives the following trigger registration: Router(config-gk)# server trigger rai alpha 2 Server-west 10.10.10.10 1751 Router(config-gk_raitrigger)# supported-prefix 1234* Router(config-gk_raitrigger)# exit
Then gatekeeper “alpha” checks all incoming RAI messages for the MCU or H.323 proxy endpoint or the supported prefix 1# before checking for the supported prefix 1234*. If any one of those conditions is met, the gatekeeper sends the RAI message to the GKTMP server “Server-west”. If the second gatekeeper “alpha” RAI trigger registration had been defined with a priority 1 instead of priority 2, then the second trigger registration would have overridden the first one. In other words, the gatekeeper “alpha” would send to GKTMP server “Server-west” only those RAI messages that contain a supported prefix of 1234*. All other RAI messages would not be sent to the GKTMP server.
Related Commands
Command
Description
server registration-port
Configures the server listening port on the gatekeeper.
show gatekeeper servers
Displays the triggers configured on the gatekeeper.
Cisco IOS Voice Command Reference
VR-1728
Cisco IOS Voice Commands: S server trigger rrq
server trigger rrq To configure the registration request (RRQ) trigger statically on the gatekeeper, use the server trigger rrq command in gatekeeper configuration mode. Submode commands are available after entering the server trigger rrq command. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the all form of the command. server trigger rrq gkid priority server-id server-ip-address server-port no server trigger rrq gkid priority server-id server-ip-address server-port no server trigger all
Syntax Description
all
Deletes all CLI-configured triggers.
gkid
Local gatekeeper identifier.
priority
Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.
server-id
ID number of the external application.
server-ip-address
IP address of the server.
server-port
Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.
Submode Commands
After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line. endpoint-type value
Use to send RRQ RAS messages containing a particular endpoint type to the GKTMP server application. •
value—Value against which to compare the endpoint-type in the RAS messages. Valid endpoint types are the following: – gatekeeper—Endpoint is an H.323 gatekeeper. – h320-gateway—Endpoint is an H.320 gateway. – mcu—Endpoint is a multipoint control unit (MCU). – other-gateway—Endpoint is another type of gateway not specified
on this list. – proxy—Endpoint is an H.323 proxy. – terminal—Endpoint is an H.323 terminal. – voice-gateway—Endpoint is a voice gateway.
info-only
Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol (GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.
Cisco IOS Voice Command Reference
VR-1729
Cisco IOS Voice Commands: S server trigger rrq
shutdown
Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.
supported-prefix value Use to send RRQ RAS messages containing a specific supported prefix to the GKTMP server application. •
value—Value against which to compare the supported prefix in the RAS messages. The possible values are any E.164 pattern used as a gateway technology prefix. The value string can contain any of the following: 0123456789#*.
Command Default
No trigger servers are set.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(11)T
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810.
Usage Guidelines
Use this command and its optional submode commands to configure the registration request (RRQ) static server trigger. The gatekeeper checks incoming gateway RRQ messages for the configured trigger information. If an incoming RRQ message contains the specified trigger information, the gatekeeper sends the RRQ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the RRQ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application. If no submode commands are configured for the RRQ messages, the gatekeeper sends all RRQ messages to the GKTMP server application. If the gatekeeper receives an RRQ trigger registration message that contains several trigger conditions, the conditions are treated as “OR” conditions. In other words, if an incoming RRQ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server. If the gatekeeper receives two RRQ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two RRQ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming RRQ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one RRQ trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations. The no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.
Cisco IOS Voice Command Reference
VR-1730
Cisco IOS Voice Commands: S server trigger rrq
Examples
The following example configures a trigger registration on gatekeeper “sj.xyz.com” to send all RRQ messages to GKTMP server “Server-123”: Router(config-gk)# server trigger rrq sj.xyz.com 1 Server-123 1.14.93.130 1751 Router(config-gk_rrqtrigger)# exit
The following example configures an RRQ trigger registration on gatekeeper “alpha”, which sends to the GKTMP server “Server-west” any RRQ message containing an MCU endpoint, an H.323 proxy endpoint, or a supported prefix 1#. Other RRQ messages are not sent to the GKTMP server. Router(config-gk)# server trigger rrq alpha 1 Server-west 10.10.10.10 1751 Router(config-gk-rrqtrigger)# endpoint-type mcu Router(config-gk-rrqtrigger)# endpoint-type proxy Router(config-gk-rrqtrigger)# supported-prefix 1# Router(config-gk-rrqtrigger# exit
If the RRQ registration message defined above for gatekeeper “alpha” is configured and the gatekeeper receives the following trigger registration: Router(config-gk)# server trigger rrq alpha 2 Server-west 10.10.10.10 1751 Router(config-gk_rrqtrigger)# supported-prefix 1234* Router(config-gk_rrqtrigger)# exit
then gatekeeper “alpha” checks all incoming RRQ messages for the MCU or H.323 proxy endpoint or the supported prefix 1# before checking for the supported prefix 1234*. If any one of those conditions is met, the gatekeeper sends the RRQ message to the GKTMP server “Server-west”. If the second gatekeeper “alpha” RRQ trigger registration had been defined with a priority 1 instead of priority 2, then the second trigger registration would have overridden the first one. In other words, the gatekeeper “alpha” would send to GKTMP server “Server-west” only those RRQ messages that contain a supported prefix of 1234*. All other RRQ messages would not be sent to the GKTMP server.
Related Commands
Command
Description
server registration-port
Configures the server listening port on the gatekeeper.
show gatekeeper servers
Displays the triggers configured on the gatekeeper.
Cisco IOS Voice Command Reference
VR-1731
Cisco IOS Voice Commands: S server trigger urq
server trigger urq To configure the unregistration request (URQ) trigger statically on the gatekeeper, use the server trigger urq command in gatekeeper configuration mode. Submode commands are available after entering the server trigger urq command. To delete a single static trigger on the gatekeeper, use the no form of this command. To delete all static triggers on the gatekeeper, use the all form of the command. server trigger urq gkid priority server-id server-ip-address server-port Submode Commands: info-only shutdown endpoint-type value supported-prefix value no server trigger urq gkid priority server-id server-ip-address server-port no server trigger all
Syntax Description
all
Deletes all CLI-configured triggers.
gkid
Local gatekeeper identifier.
priority
Priority for each trigger. Range is from 1 to 20; 1 is the highest priority.
server-id
ID number of the external application.
server-ip-address
IP address of the server.
server-port
Port on which the Cisco IOS gatekeeper listens for messages from the external server connection.
Submode Commands
After the command is entered, the software enters a submode that permits you to configure additional filters on the reliability, availability, and serviceability (RAS) message. These filters are optional, and you may configure any of them, one per command line. endpoint-type value
Use to send URQ RAS messages containing a particular endpoint type to the GKTMP server application. •
value—Value against which to compare the endpoint-type in the RAS messages. Valid endpoint types are the following: – gatekeeper—Endpoint is an H.323 gatekeeper. – h320-gateway—Endpoint is an H.320 gateway. – mcu—Endpoint is a multipoint control unit (MCU). – other-gateway—Endpoint is another type of gateway not specified
on this list. – proxy—Endpoint is an H.323 proxy. – terminal—Endpoint is an H.323 terminal. – voice-gateway—Endpoint is a voice gateway.
Cisco IOS Voice Command Reference
VR-1732
Cisco IOS Voice Commands: S server trigger urq
info-only
Use to indicate to the gatekeeper that messages that meet the specified trigger parameters should be sent to the Gatekeeper Transaction Message Protocol (GKTMP) server application as notifications only and that the gatekeeper should not wait for a response from the GKTMP server application.
shutdown
Use to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward to the GKTMP server application.
supported-prefix value Use to send URQ RAS messages containing a specific supported prefix to the GKTMP server application. •
value—Value against which to compare the supported prefix in the RAS messages. The possible values are any E.164 pattern used as a gateway technology prefix. The value string can contain any of the following: 0123456789#*.
Command Default
No trigger servers are set.
Command Modes
Gatekeeper configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(11)T
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, Cisco 7200 series, and Cisco MC3810.
Usage Guidelines
Use this command and its optional submode commands to configure the unregistration request (URQ) static server trigger. The gatekeeper checks incoming gateway URQ messages for the configured trigger information. If an incoming URQ message contains the specified trigger information, the gatekeeper sends the URQ message to the GKTMP server application. In addition, the gatekeeper processes the message according to its programmed instructions. If the URQ message does not contain the specified information, the gatekeeper processes the message but does not send it to the GKTMP server application. If no submode commands are configured for the URQ messages, the gatekeeper sends all URQ messages to the GKTMP server application. If the gatekeeper receives a URQ trigger registration message that contains several trigger conditions, the conditions are treated as “OR” conditions. In other words, if an incoming URQ RAS message meets any one of the conditions, the gatekeeper sends the RAS message to the GKTMP server. If the gatekeeper receives two URQ trigger registration messages with the same priority for the same GKTMP server, the gatekeeper retains the second registration and discards the first one. If the gatekeeper receives two URQ trigger registration messages with different priorities for the same GKTMP server, the gatekeeper checks incoming URQ messages against the conditions on the higher priority registration before using the lower priority registration. If the gatekeeper receives more than one URQ trigger registration message with the same priority but for different GKTMP servers, the gatekeepers retains all of the registrations.
Cisco IOS Voice Command Reference
VR-1733
Cisco IOS Voice Commands: S server trigger urq
The the no form of the command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.
Examples
The following example configures a trigger registration on gatekeeper “sj.xyz.com” to send all URQ messages to GKTMP server “Server-123”: Router(config-gk)# server trigger urq sj.xyz.com 1 Server-123 1.14.93.130 1751 Router(config-gk_urqtrigger)# exit
The following example configures a URQ trigger registration on gatekeeper “alpha”, which sends to the GKTMP server “Server-west” any URQ message containing an MCU endpoint, an H.323 proxy endpoint, or a supported prefix 1#. Other URQ messages are not sent to the GKTMP server. Router(config-gk)# server trigger urq alpha 1 Server-west 10.10.10.10 1751 Router(config-gk-urqtrigger)# endpoint-type mcu Router(config-gk-urqtrigger)# endpoint-type proxy Router(config-gk-urqtrigger)# supported-prefix 1# Router(config-gk-urqtrigger# exit
If the URQ registration message defined above for gatekeeper “alpha” is configured and the gatekeeper receives the following trigger registration: Router(config-gk)# server trigger urq alpha 2 Server-west 10.10.10.10 1751 Router(config-gk_urqtrigger)# supported-prefix 1234* Router(config-gk_urqtrigger)# exit
then gatekeeper “alpha” checks all incoming URQ messages for the MCU or H.323 proxy endpoint or the supported prefix 1# before checking for the supported prefix 1234*. If any one of those conditions is met, the gatekeeper sends the URQ message to the GKTMP server “Server-west”. If the second gatekeeper “alpha” URQ trigger registration had been defined with a priority 1 instead of priority 2, then the second trigger registration would have overridden the first one. In other words, the gatekeeper “alpha” would send to GKTMP server “Server-west” only those URQ messages that contain a supported prefix of 1234*. All other URQ messages would not be sent to the GKTMP server.
Related Commands
Command
Description
server registration-port
Configures the server listening port on the gatekeeper.
show gatekeeper servers
Displays the triggers configured on the gatekeeper.
Cisco IOS Voice Command Reference
VR-1734
Cisco IOS Voice Commands: S service
service To load and configure a specific, standalone application on a dial peer, use the service command in application configuration mode. To remove the application from the dial peer, use the no form of this command. service [alternate | default] service-name location no service [alternate | default] service-name location
Syntax Description
alternate
(Optional) Alternate service to use if the service that is configured on the dial peer fails.
default
(Optional) Specifies that the default service (“DEFAULT”) on the dial peer is used if the alternate service fails.
service-name
Name that identifies the voice application. This is a user-defined name and does not have to match the script name.
location
Directory and filename of the Tcl script or VoiceXML document in URL format. For example, the following are valid locations: •
built-in applications (builtin:filename)
•
flash memory (flash:filename)
•
HTTP server (http://../filename)
•
HTTPS (HTTP over Secure Socket Layer (SSL)) server (https://../filename)
•
TFTP server (tftp://../filename)
Command Default
The default service (“DEFAULT”) is used if no other services are configured.
Command Modes
Application configuration (config-app)
Command History
Release
Modification
12.3(14)T
This command was introduced.
12.4(15)T
The location argument was modified to accept an HTTPS server URL. The description of the location argument was modified to describe how to specify the location for built-in applications.
Usage Guidelines
Use this command to load a service on the gateway. A service is a standalone application, such as a VoiceXML document or a Tcl script.
Cisco IOS Voice Command Reference
VR-1735
Cisco IOS Voice Commands: S service
Examples
The following example shows a debitcard application configured on the dial peer. Router(config)# application Router(config-app)# service debitcard tftp://server-1//tftpboot/scripts/app_debitcard.2.0.2.8.tcl
The following example shows the VoiceXML application myapp located on an HTTPS server configured on the dial peer. Router(config)# application Router(config-app)# service myapp https://myserver/myfile.vxml
The following example shows the auto-attendant (AA) service, called aa, which is a Tcl script embedded in the Cisco IOS software. Router(config)# application Router(config-app)# service queue builtin:app-b-acd
Related Commands
Command
Description
application (application configuration)
Configures an application on a dial peer.
call application alternate
Specifies an alternate application to use if the application that is configured in the dial peer fails.
call application voice
Defines the name of a voice application and specifies the location of the Tcl or VoiceXML document to load for this application.
Cisco IOS Voice Command Reference
VR-1736
Cisco IOS Voice Commands: S service dsapp
service dsapp To configure supplementary IP Centrex-like services for FXS phones on voice gateways to interwork with SIP-based softswitches, use the service dsapp command in the gateway-application configuration mode. Hookflash triggers a supplementary feature based on the current state of the call. To reset to the defaults, use the no form of this command. service dsapp [paramspace dialpeer dial-peer tag] [paramspace disc-toggle-time seconds] [paramspace callWaiting TRUE | FALSE] [paramspace callConference TRUE | FALSE] [paramspace blind-xfer-wait-time seconds] [paramspace callTransfer TRUE | FALSE] no service dsapp
Syntax Description
paramspace
Defines a package or service on the gateway, the parameters in that package or service become available for configuration when you use this argument.
dialpeer dial-peer tag
(Optional) Specifies the fixed dialpeer used to setup the call to the SIP server (trunk) side.
disc-toggle-time seconds
(Optional) Specifies the seconds to wait before switching to a call on hold if the active call disconnects. You can specify a range between 10 and 30 seconds.
callWaiting TRUE | FALSE
Toggles support for call waiting.
callConference TRUE | FALSE
Toggles support for call conferencing used to establish two calls with a single connection such that all three parties can talk together.
blind-xfer-wait-time seconds Specifies the seconds to wait before triggering a blind call transfer. You can specify a range between 0 and 10 seconds. If you specify 0 seconds, no blind transfer call occurs. callTransfer TRUE | FALSE
Command Defaults
Toggles support for call transfers.
If no supplementary features are defined, the defaults are as follows: •
dialpeer: -1
•
disc-toggle-time: 10 seconds
•
callWaiting: TRUE (enabled)
•
callConference: TRUE (enabled)
•
blind-xfer-wait-time: 0 seconds
•
callTransfer: TRUE (enabled)
Command Modes
Gateway-application configuration
Command History
Release
Modification
12.4(11)T
This command was introduced.
Cisco IOS Voice Command Reference
VR-1737
Cisco IOS Voice Commands: S service dsapp
Usage Guidelines
Use the service dsapp command to configure supplementary Centrex-like features on FXS phones to interwork with SIP-based softswitches. Hookflash triggers supplementary features based on the current state of the call: •
Call Hold
•
Call Waiting
•
Call Transfer
•
3-Way Conference
Call Hold
Allows a call to be placed in a non-active state (with no media exchange). Table 38 summarizes the hookflash feature support for Call Hold. Table 38
Call Hold Hookflash Services
State
Action
Result
Active call
Hookflash
Held call for remote party. Second dial tone for FXS phone.
Call on hold
Hookflash
Active call.
FXS line connects to call.
Call on hold and active call
Hookflash
Active and held calls are swapped.
FXS line connects to previous held call.
On hook
Active call is dropped.
Reminder ring on FXS line.
Call on hold goes on hook
Call on hold is dropped.
None.
Active call goes on hook Active call is dropped
Response to FXS Line
Silence.
Call Waiting
Allows a second call to be received while the phone is active with a call. Table 39 summarizes the hookflash feature support for Call Waiting. Table 39
Call Waiting Hookflash Services
State
Action
Result
Response to FXS Line
Active call and waiting call
Hookflash.
Swap active call and waiting call.
FXS line connects to waiting call.
Active call goes on hook. Active call is disconnected.
Silence.
Waiting call goes on hook.
Stay connected to active call.
None.
On hook.
Active call is dropped.
Reminder ring on FXS line.
Call Transfer
With call transfer, you can do the following: •
Put an active call on hold while establishing a second call.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S service dsapp
•
Set up a call between two users
•
Transfer calls by using these options – -Blind transfer – Semi-attended transfer – Attended transfer
Table 40 summarizes the hookflash feature support for Call Transfer. Table 40
Call Transfer Hookflash Services
State
Action
Result
Response to FXS Line
Active call
Hookflash.
Call is placed on hold.
Second dial tone.
Call on hold and On hook. outgoing dialed or alerting or active call
Call on hold and active call.
Call on hold and outgoing active call
Active call goes on hook. Held call remains; active call dropped.
Silence.
Call on hold and outgoing active call
Call on hold goes on hook.
Active call remains; call on hold dropped.
None.
Call on hold and outgoing alerting call
Hookflash.
Active call dropped.
FXS line connects to previous held call.
3-Way Conference
Establishes two calls with a single connection, so that three parties can talk together. Table 41 summarizes the hookflash feature support for 3-way conferencing. Table 41
3-Way Conference Hookflash Services
State
Action
Result
Response to FXS Line
Active call
Hookflash
Call on hold.
Second dial tone.
Join call on hold and active call.
Media mixing of both calls.
Call on hold and active call
Examples
Enabling the DSApp Service
You can configure DSApp services either to a specific dial-peer, or globally to all dial peers. The following example shows the configuration to enable DSApp on a specific dial peer: Gateway# configure terminal Enter configuration commands, one per line. End with CNTL/Z. Gateway(conf)# application Gateway(conf-app)# dial-peer voice 1000 pots Gateway(config-app)# service dsapp
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S service dsapp
The following example shows the configuration to enable DSApp globally on all dial peers: Gateway# configure terminal Enter configuration commands, one per line. End with CNTL/Z. Gateway(conf)# application Gateway(config-app)# global Gateway(config-app-global)# service default dsapp
Configuring Call Hold
The following example shows the configuration to enable the Call Hold feature: Gateway# configure terminal Enter configuration commands, one per line. End with CNTL/Z. Gateway(conf)# application Gateway(config-app)# service dsapp Gateway(config-app-param)# param callHold TRUE
Configuring Call Waiting
The following example shows the configuration to enable the Call Waiting feature: Gateway# configure terminal Enter configuration commands, one per line. End with CNTL/Z. Gateway(conf)# application Gateway(config-app)# service dsapp Gateway(config-app-param)# param callWaiting TRUE
Configuring Call Transfer
The following example shows the configuration to enable the Call Transfer feature: Gateway# configure terminal Enter configuration commands, one per line. End with CNTL/Z. Gateway(conf)# application Gateway(config-app)# service dsapp Gateway(config-app-param)# param callTransfer TRUE
Configuring 3-Way Conferencing
The following example shows the configuration to enable the 3-Way Conferencing feature: Gateway# configure terminal Enter configuration commands, one per line. End with CNTL/Z. Gateway(conf)# application Gateway(config-app)# service dsapp Gateway(config-app-param)# param callConference TRUE
Configuring Disconnect Toggle Time
In this example, a disconnect toggle time is configured that specifies the amount of time in seconds the system should wait before committing the call transfer after the originating call is placed on hook. Gateway# configure terminal Enter configuration commands, one per line. End with CNTL/Z. Gateway(conf)# application Gateway(config-app)# service dsapp Gateway(config-app-param)# param disc-toggle-time 10
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S service dsapp
Configuring Blind Transfer Wait Time
In this example, a blind transfer call wait time is configured that specifies the amount of time in seconds the system should wait before committing the call transfer, after the originating call is placed on hook. Gateway# configure terminal Enter configuration commands, one per line. End with CNTL/Z. Gateway(conf)# application Gateway(config-app)# service dsapp Gateway(config-app-param)# param blind-xfer-wait-time 10
Configuring a Fixed Dial Peer Used for Outgoing Calls on SIP Trunk Side
In this example, a fixed dial peer is configured to set up a call to the SIP server (trunk) side. Gateway# configure terminal Enter configuration commands, one per line. End with CNTL/Z. Gateway(conf)# application Gateway(config-app)# service dsapp Gateway(config-app-param)# param dialpeer 5000
Related Commands
Command
Description
offer call-hold
Specifies the method of call hold on the gateway.
Cisco IOS Voice Command Reference
VR-1741
Cisco IOS Voice Commands: S service-flow primary upstream
service-flow primary upstream To assign a quality of service (QoS) policy to the data traveling between the cable modem and the multiple service operator (MSO) cable modem termination system (CMTS), use the service-flow primary upstream command in interface configuration mode. To disable the QoS policy, use the no form of this command. service-flow primary upstream no service-flow primary upstream
Syntax Description
This command has no arguments or keywords.
Command Default
This command is disabled by default.
Command Modes
Interface configuration
Command History
Release
Modification
12.4(11)T
This command was introduced.
Usage Guidelines
This command is supported in the upstream direction only. Service flows are unidirectional.
Examples
The following example assigns a QoS policy to the data traveling between the cable modem and the MSO CMTS: Router# configure terminal Router(config)# interface Cable-Modem 0/2/0 Router(config-if)# service-flow primary upstream
Cisco IOS Voice Command Reference
VR-1742
Cisco IOS Voice Commands: S service-relationship
service-relationship To enter Annex G neighbor configuration mode and enable service relationships for the particular neighbor, use the service-relationship command in Annex G neighbor configuration mode. To exit this mode, use the no form of this command. service-relationship no service-relationship
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled
Command Modes
Annex G neighbor configuration
Command History
Release
Modification
12.2 (11)T
This command was introduced.
Usage Guidelines
Note
Examples
Service relationships are defined to be unidirectional. If a service relationship is established between border element A and border element B, A is entitled to send requests to B and to expect responses. For B to send requests to A and to expect responses, a second service relationship must be established. Repeat this command for each border-element neighbor that you configure.
The no shutdown command must be used to enable each service relationship.
The following example enables a service relationship on a border element: Router(config-annexg-neigh)# service-relationship
Related Commands
Command
Description
access-policy
Requires that a neighbor be explicitly configured.
inbound ttl
Sets the inbound time-to-live value.
outbound retry-interval
Defines the retry period for attempting to establish the outbound relationship between border elements.
retry interval
Defines the time between delivery attempts.
retry window
Defines the total time for which a border element attempts delivery.
shutdown
Enables or disables the border element.
Cisco IOS Voice Command Reference
VR-1743
Cisco IOS Voice Commands: S service-type call-check
service-type call-check To identify preauthentication requests to the authentication, authorization, and accounting (AAA) server, use the service-type call-check command in AAA preauthentication configuration mode. To return this setting to the default, use the no form of this command. service-type call-check no service-type call-check
Syntax Description
This command has no arguments or keywords.
Command Default
The service type is not set to call-check.
Command Modes
AAA preauthentication configuration
Command History
Release
Modification
12.2(11)T
This command was introduced.
Usage Guidelines
Setting the service-type attribute to call-check causes preauthentication access requests to include this value, which allows AAA servers to distinguish preauthentication requests from other types of Access-Requests. This command has no effect on packets that are not of the preauthentication type.
Examples
The following example sets the RADIUS service-type attribute to call-check: Router(config)# aaa preauth Router(config-preauth)# service-type call-check
Related Commands
Command
Description
aaa preauth
Enters AAA preauthentication configuration mode.
Cisco IOS Voice Command Reference
VR-1744
Cisco IOS Voice Commands: S session
session To associate a transport session with a specified session group, use the session command in backhaul session manager configuration mode. To delete the session, use the no form of this command. session group group-name remote-ip remote-port local-ip local-port priority no session group group-name remote-ip remote-port local-ip local-port priority
Syntax Description
group-name
Session-group name.
remote-ip
Remote IP address.
remote-port
Remote port number. Range is from 1024 to 9999.
local-ip
Local IP address.
local-port
Local port number. Range is from 1024 to 9999.
priority
Priority of the session-group. Range is from 0 to 9999; 0 is the highest priority.
Command Default
No default behavior or values
Command Modes
Backhaul session manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(4)T
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.2(2)XB
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was implemented on the Cisco IAD2420 series. Support for the Cisco AS5350 and Cisco AS5400 and Cisco AS5850 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Usage Guidelines
It is assumed that the server is located on a remote machine.
Examples
The following example associates a transport session with the session group “group5” and specifies the parameters: Router(config-bsm)# session group group5 172.13.2.72 5555 172.18.72.198 5555 1
Cisco IOS Voice Command Reference
VR-1754
Cisco IOS Voice Commands: S session group
session group To associate a transport session with a specified session group, use the session group command in backhaul session-manager configuration mode. To delete the session, use the no form of this command. session group group-name remote-ip remote-port local-ip local-port priority no session group group-name remote-ip remote-port local-ip local-port priority
Syntax Description
group-name
Session-group name.
remote-ip
Remote IP address.
remote-port
Remote port number. Range is from 1024 to 9999.
local-ip
Local IP address.
local-port
Local port number. Range is from 1024 to 9999.
priority
Priority of the session group. Range is from 0 to 9999; 0 has the highest priority.
Command Default
No default behavior or values.
Command Modes
Backhaul session-manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(2)T
This command was implemented on the Cisco 7200 series.
12.2(4)T
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was implemented on the Cisco IAD2420 series.
Usage Guidelines
The Cisco VSC3000 server is assumed to be located on a remote machine.
Examples
The following example associates a transport session with the session group named “group5” and specifies the keywords described above: session group group5 172.16.2.72 5555 192.168.72.198 5555 1
Cisco IOS Voice Command Reference
VR-1755
Cisco IOS Voice Commands: S session protocol (dial peer)
session protocol (dial peer) To specify a session protocol for calls between local and remote routers using the packet network, use the session protocol command in dial peer configuration mode. To reset to the default, use the no form of this command. session protocol {aal2-trunk | cisco | sipv2 | smtp} no session protocol
Dial peer uses the proprietary Cisco VoIP session protocol.
sipv2
Dial peer uses the Internet Engineering Task Force (IETF) Session Initiation Protocol (SIP). Use this keyword with the SIP option.
smtp
Dial peer uses Simple Mail Transfer Protocol (SMTP) session protocol.
Command Default
No default behaviors or values
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced for VoIP peers on the Cisco 3600 series.
12.0(3)XG
This command was modified to support VoFR) dial peers.
12.0(4)XJ
This command was modified for store-and-forward fax on the Cisco AS5300.
12.1(1)XA
This command was implemented for VoATM dial peers on the Cisco MC3810. The aal2-trunk keyword was added.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T. The sipv2 keyword was added.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.2(2)T
This command was implemented on the Cisco 7200 series.
12.2(4)T
This command was implemented on the Cisco 1750.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the Cisco 7200 series. Supported for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release. The aal2-trunk and smtp keywords are not supported on the Cisco 7200 series in this release.
12.2(11)T
This command is supported on the Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Cisco IOS Voice Command Reference
VR-1756
Cisco IOS Voice Commands: S session protocol (dial peer)
Usage Guidelines
The cisco keyword is applicable only to VoIP on the Cisco 1750, Cisco 1751, Cisco 3600 series, and Cisco 7200 series routers. The aal2-trunk keyword is applicable only to VoATM on the Cisco 7200 series router. This command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following example shows that AAL2 trunking has been configured as the session protocol: dial-peer voice 10 voatm session protocol aal2-trunk
The following example shows that Cisco session protocol has been configured as the session protocol: dial-peer voice 20 voip session protocol cisco
The following example shows that a VoIP dial peer for SIP has been configured as the session protocol for VoIP call signaling: dial-peer voice 102 voip session protocol sipv2
Related Commands
Command
Description
dial-peer voice
Enters dial peer configuration mode and specifies the method of voice-related encapsulation.
session target (VoIP)
Configures a network-specific address for a dial peer.
Cisco IOS Voice Command Reference
VR-1757
Cisco IOS Voice Commands: S session protocol (Voice over Frame Relay)
session protocol (Voice over Frame Relay) To establish a Voice over Frame Relay protocol for calls between the local and remote routers via the packet network, use the session protocol command in dial peer configuration mode. To reset to the default, use the no form of this command. session protocol {cisco-switched | frf11-trunk} no session protocol
Syntax Description
cisco-switched
Proprietary Cisco VoFR session protocol. (This is the only valid session protocol for the Cisco 7200 series.)
frf11-trunk
FRF.11 session protocol.
Command Default
cisco-switched
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced for VoIP.
12.0(3)XG
This command was modified to support VoFR on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, and Cisco MC3810.
12.0(4)T
The cisco-switched and frf11-trunk keywords were added for VoFR dial peers.
Usage Guidelines
Note
Examples
For Cisco-to-Cisco dial peer connections, Cisco recommends that you use the default session protocol because of the advantages it offers over a pure FRF.11 implementation. When connecting to FRF.11-compliant equipment from other vendors, use the FRF.11session protocol.
When using the FRF.11 session protocol, you must also use the called-number command.
The following example configures the FRF.11 session protocol for VoFR dial peer 200: dial-peer voice 200 vofr session protocol frf11-trunk called-number 5552150
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S session protocol (Voice over Frame Relay)
Related Commands
Command
Description
called-number (dial peer)
Enables an incoming VoFR call leg to get bridged to the correct POTS call leg when using a static FRF.11 trunk connection.
codec (dial peer)
Specifies the voice coder rate of speech for a Voice over Frame Relay dial peer.
cptone
Specifies a regional analog voice interface-related tone, ring, and cadence setting.
destination-pattern
Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.
dtmf-relay (Voice over Frame Relay)
Enables the generation of FRF.11 Annex A frames for a dial peer.
preference
Indicates the preferred order of a dial peer within a rotary hunt group.
session target
Specifies a network-specific address for a specified dial peer or destination gatekeeper.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S session protocol aal2
session protocol aal2 To enter voice-service-session configuration mode and specify ATM adaptation layer 2 (AAL2) trunking, use the session protocol aal2 command in voice-service configuration mode. session protocol aal2
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values.
Command Modes
Voice-service configuration
Command History
Release
Modification
12.1(1)XA
This command was introduced on the Cisco MC3810.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.2(2)T
This command was implemented on the Cisco 7200 series.
Usage Guidelines
This command applies to VoATM on theCisco 7200 series router. In the voice-service-session configuration mode for AAL2, you can configure only AAL2 features, such as call admission control and subcell multiplexing.
Examples
The following example accesses voice-service-session configuration mode, beginning in global configuration mode: voice service voatm session protocol aal2
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S session protocol multicast
session protocol multicast To set the session protocol as multicast, use the session protocol multicast command in dial peer configuration mode. To reset to the default protocol, use the no version of this command. session protocol multicast no session protocol multicast
Syntax Description
This command has no arguments or keywords.
Command Default
Default session protocol: Cisco.
Command Modes
Dial peer configuration
Command History
Release
Modification
12.1(2)XH
This command was introduced for the Cisco Hoot and Holler over IP application on the Cisco 2600 series and Cisco 3600 series.
12.1(3)T
This command was integrated into Cisco IOS Release 12.1(3)T.
12.2(8)T
This command was implemented on the Cisco 1750 and Cisco 1751.
Usage Guidelines
Use this command for voice conferencing in a hoot and holler networking implementation. This command allows more than two ports to join the same session simultaneously.
Examples
The following example shows the use of the session protocol multicast dial peer configuration command in context with its accompanying commands: dial-peer voice 111 voip destination-pattern 111 session protocol multicast session target ipv4:237.111.0.111:22222 ip precedence 5 codec g711ulaw
Related Commands
Command
Description
session target ipv4
Assigns the session target for voice-multicasting dial peers.
Cisco IOS Voice Command Reference
VR-1761
Cisco IOS Voice Commands: S session refresh
session refresh To enable SIP session refresh globally, use the session refresh command in SIP configuration mode. To disable the session refresh, use the no form of this command. session refresh no session refresh
Syntax Description
This command has no arguments or keywords.
Command Default
No session refresh
Command Modes
SIP configuration (conf-serv-sip)
Command History
Release
Modification
15.1(2)T
This command was introduced.
Usage Guidelines
Use the SIP session refresh command to send the session refresh request.
Examples
The following example sets the session refresh under SIP configuration mode: Router(conf-serv-sip)# Session refresh
Related Commands
Command
Description
voice-class sip session refresh
Enables session refresh at dial-peer level.
Cisco IOS Voice Command Reference
VR-1762
Cisco IOS Voice Commands: S session start
session start To start a new instance (session) of a Tcl IVR 2.0 application, use the session start command in application configuration mode. To stop the session and remove the configuration, use the no form of this command. session start instance-name application-name no session start instance-name
Syntax Description
instance-name
Alphanumeric label that uniquely identifies this application instance.
application-name
Name of the Tcl application. This is the name of the application that was assigned with the service command.
Command Default
No default behavior or values
Command Modes
Application configuration
Command History
Release
Modification
12.3(14)T
This command was introduced to replace the call application session start (global configuration) command.
Usage Guidelines
Examples
•
This command starts a new session, or instance, of a Tcl IVR 2.0 application. It cannot start a session for a VoiceXML application because Cisco IOS software cannot start a VoiceXML application without an active call leg.
•
You can start an application instance only after the Tcl application is loaded onto the gateway with the service command.
•
If this command is used, the session restarts if the gateway reboots.
•
If the application session stops running, it does not restart unless the gateway reboots. A Tcl script might intentionally stop running by executing a “call close” command for example, or it might fail because of a script error.
•
You can start multiple instances of the same application by using different instance names.
The following example starts a session named my_instance for the application named demo: application session start my_instance demo The following example starts another session for the application named demo: application session start my_instance2 demo
Starts a new instance (session) of a Tcl IVR 2.0 application.
service
Loads a specific, standalone application on a dial peer.
show call application services Displays a one-line summary of all registered services. registry show call application sessions Displays summary or detailed information about voice application sessions.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S session target (MMoIP dial peer)
session target (MMoIP dial peer) To designate an e-mail address to receive T.37 store-and-forward fax calls from a Multimedia Mail over IP (MMoIP) dial peer, use the session target command in dial peer configuration mode. To remove the target address, use the no form of this command. session target mailto:{name | $d$ | $m$ | $e$}[@domain-name] no session target
Syntax Description
mailto:
Matching calls are passed to the network using Simple Mail Transfer Protocol (SMTP) or Extended Simple Mail Transfer Protocol (ESMTP).
name
String that can be an e-mail address, name, or mailing list alias.
$d$
Macro that is replaced by the destination pattern of the gateway access number, which is the called number or dialed number identification service (DNIS) number.
$m$
Macro that is replaced by the redirecting dialed number (RDNIS) if present; otherwise, it is replaced by the gateway access number (DNIS). This macro requires use of the fax detection interactive voice response (IVR) application. Note
$e$
Other strings can be passed to mailto in place of $m$ if you modify the fax detection application Tool Command Language (TCL) script or VoiceXML document. For more information, refer to the readme file that came with the TCL script or the Cisco VoiceXML Programmer’s Guide.
Macro that is replaced by the DNIS, the RDNIS, or a string that represents a valid e-mail address, as specified by the cisco-mailtoaddress variable in the transfer tag of the VoiceXML fax detection document. By default, if the cisco-mailtoaddress variable is not specified in the fax detection document, the DNIS is mapped to $e$. If $e$ is not specified for the session target mailto command in the MMoIP dial peer, but the cisco-mailtoaddress variable is specified in the transfer tag of the fax detection document, then whatever is specified in the MMoIP dial peer takes precedence; the cisco-mailtoaddress variable is ignored. Note
@domain-name
If a domain name is configured with this command, the VoiceXML document should pass only the username portion of the e-mail address and not the domain. If the domain name is passed from cisco-mailtoaddress, the session target mailto command should specify only $e$.
(Optional) String that contains the domain name to be associated with the target address, preceded by the at sign (@); for example, @mycompany.com.
Command Default
No default behavior or values
Command Modes
Dial peer configuration
Cisco IOS Voice Command Reference
VR-1765
Cisco IOS Voice Commands: S session target (MMoIP dial peer)
Command History
Usage Guidelines
Release
Modification
11.3(1)T
This command was introduced.
12.0(4)T
This command was modified to support store-and-forward fax.
12.1(5)XM1
The $m$ keyword was introduced for the fax detection feature on the Cisco AS5300.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB
The $e$ keyword was introduced for VoiceXML fax detection on the Cisco AS5300.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the following platforms: Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.2(11)T
This command was implemented on the following platforms: Cisco AS5300, Cisco AS5350, and Cisco AS5400.
Use this command to deliver e-mail to one recipient by specifying one e-mail name, or to deliver e-mail to multiple recipients by specifying an e-mail alias as the name argument and having that alias expanded by the mailer. Use the $m$ macro to include the redirecting dialed number (RDNIS) as part of the e-mail name when using the fax detection IVR application. If $m$ is specified and RDNIS is not present in the call information, the access number of the gateway (the dialed number, or DNIS) is used instead. For example, if the calling party originally dialed 6015551111 to send a fax, and the call was redirected (forwarded on busy or no answer) to 6015552222 (the gateway), the RDNIS is 6015551111, and the DNIS is 6015552222. Use the $e$ macro to map the cisco-mailtoaddress variable in the VoiceXML fax detection document to the username portion of the e-mail address when sending a fax. If the VoiceXML document does not specify the cisco-mailtoaddress variable in the transfer tag, the application maps the DNIS to the e-mail address username.
Examples
The following example delivers fax-mail to multiple recipients: dial-peer voice 10 mmoip session target mailto:[email protected]
Assuming that mailer.example.com is running the sendmail application, you can put the following information into its /etc/aliases file: marketing-information: [email protected], [email protected]
The following example uses the fax detection IVR application. Here, the session target (MMoIP dial peer) command forwards fax calls to an e-mail account that uses the Redirected Dialed Number Identification Service (RDNIS) as part of its address. In this example, the calling party originally dialed 6015551111 to send a fax, and the call was forwarded (on busy or no answer) to 6015552222, which is the incoming number for the gateway being configured. The RDNIS is 6015551111, and the dialed number (DNIS) is 6015552222. When faxes are forwarded from the gateway, the session target in the example is expanded to [email protected].
Cisco IOS Voice Command Reference
VR-1766
Cisco IOS Voice Commands: S session target (MMoIP dial peer)
The following examples configure a session target for a VoiceXML fax detection application. In this example, the VoiceXML document passes just the username portion of the e-mail address, for example, “johnd”: dial-peer voice 4 mmoip session target mailto:[email protected]
In this example, the VoiceXML document passes the complete e-mail address including domain name, for example, “[email protected]”: dial-peer voice 5 mmoip session target mailto:$e$
Related Commands
Command
Description
destination-pattern
Specifies either the partial or full E.164 telephone number (depending on your dial plan) used to match the dial peer.
dial-peer voice
Enters dial peer configuration mode and defines a dial peer.
Cisco IOS Voice Command Reference
VR-1767
Cisco IOS Voice Commands: S session target (POTS dial peer)
session target (POTS dial peer) To designate loopback calls from a POTS dial peer, use the session target command in dial peer configuration mode. To reset to the default, use the no form of this command. session target loopback:compressed | loopback:uncompressed no session target
Syntax Description
loopback:compressed
All voice data is looped back in compressed mode to the source.
loopback:uncompressed
All voice data is looped back in uncompressed mode to the source.
Command Default
No loopback calls are designated.
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 2600 series and Cisco 3600 series.
12.0(3)T
This command was implemented on the Cisco AS5300.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400 and Cisco AS5850 is not included in this release.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T and is supported on the Cisco AS5200, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Usage Guidelines
Use this command to test the voice transmission path of a call. The loopback point depends on the call origin and the loopback type selected.
Examples
The following example loops back the traffic from the dial peer in compressed mode: dial-peer voice 10 pots session target loopback:compressed
Related Commands
Command
Description
dial-peer voice
Enters dial peer configuration mode and specifies the method of voice-related encapsulation.
Cisco IOS Voice Command Reference
VR-1768
Cisco IOS Voice Commands: S session target (VoATM dial peer)
session target (VoATM dial peer) To specify a network-specific address for a specified VoATM dial peer, use the session target command in dial peer configuration mode. To reset to the default, use the no form of this command. Cisco 3600 Series Routers
session target interface pvc {name | vpi/vci | vci} no session target Cisco 7200 Series Routers
Interface type and interface number on the router.
slot/port
Slot and port numbers for the dial-peer address.
pvc
Specific ATM permanent virtual circuit (PVC) for this dial peer.
name
PVC name.
word
(Optional) Name that identifies the PVC. The argument can identify the PVC if a word identifier was assigned when the PVC was created.
vpi/vci
ATM network virtual path identifier (VPI) and virtual channel identifier (VCI) of this PVC. Values are as follows: •
Cisco 3600 series with Multiport T1/E1 ATM network module with inverse multiplexing over ATM (IMA): vpi range is from 0 to 5; vci range is from 1 to 255.
•
OC3 ATM network module: vpi range is from 0 to 15; vci range is from 1 to 1023.
vci
ATM network virtual channel identifier (VCI) of this PVC.
cid
ATM network channel identifier (CID) of this PVC. Range is from 8 to 255.
Command Default
Command is enabled with no IP address or domain name defined.
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced.
Cisco IOS Voice Command Reference
VR-1769
Cisco IOS Voice Commands: S session target (VoATM dial peer)
Usage Guidelines
Release
Modification
11.3(1)MA
This command was modified to support VoATM, VoHDLC, and POTS dial peers. The command was implemented on the Cisco MC3810.
12.0(3)XG
This command was modified to support VoFR dial peers. The command was implemented on the Cisco 2600 series and Cisco 3600 series.
12.0(4)T
This command was integrated into Cisco IOS Release 12.0(4)T.
12.0(7)XK
This command was modified to support VoATM and VoIP dial peers. The command was implemented on the Cisco 3600 series and the Cisco MC3810. Support for VoHDLC was removed.
12.1(1)XA
This command was modified to provide enhanced support for VoATM dial peers.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.2(2)T
This command was implemented on the Cisco 7200 series.
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol that you select. The syntax of this command complies with the simple syntax of mailto: as described in RFC 1738. Use the session target loopback command to test the voice transmission path of a call. The loopback point depends on the call origin and the loopback type selected. This command applies to on-ramp store-and-forward fax functions. You must enter the session protocol aal2-trunk dial peer configuration command before you can specify a CID for a dial peer for VoATM on the Cisco 7200 series router.
Note
Examples
This command does not apply to POTS dial peers.
The following example configures a session target for VoATM. The session target is sent to ATM interface 0 for a PVC with a VCI of 20. dial-peer voice 12 voatm destination-pattern 13102221111 session target atm0 pvc 20
The following example delivers fax-mail to multiple recipients: dial-peer voice 10 mmoip session target [email protected]
Assuming that mailer.example.com is running sendmail, you can put the following information into its /etc/aliases file: marketing-information: [email protected],
Cisco IOS Voice Commands: S session target (VoATM dial peer)
The following example configures a session target for VoATM. The session target is sent to ATM interface 0, and is for a PVC with a VPI/VCI of 1/100. dial-peer voice 12 voatm destination-pattern 13102221111 session target atm1/0 pvc 1/100
Related Commands
Command
Description
called-number
Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.
codec (dial peer)
Specifies the voice coder rate of speech for a dial peer.
cptone
Specifies a regional tone, ring, and cadence setting for an analog voice port.
destination-pattern
Specifies either the prefix or full E.164 telephone number (depending on the dial plan) to be used for a dial peer.
dtmf-relay
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
preference
Indicates the preferred selection order of a dial peer within a hunt group.
session protocol
Establishes a VoFR protocol for calls between local and remote routers via the packet network.
session target
Configures a network-specific address for a dial peer.
session target loopback
Tests the voice transmission path of a call.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
Cisco IOS Voice Command Reference
VR-1771
Cisco IOS Voice Commands: S session target (VoFR dial peer)
session target (VoFR dial peer) To specify a network-specific address for a specified VoFR dial peer, use the session target command in dial peer configuration mode. To reset to the default, use the no form of this command. Cisco 2600 Series and Cisco 3600 Series Routers
session target interface dlci [cid] no session target Cisco 7200 Series Routers
session target interface dlci no session target
Syntax Description
interface
Serial interface and interface number (slot number and port number) associated with this dial peer. For the range of valid interface numbers for the selected interface type, enter a ? character after the interface type.
dlci
Data link connection identifier for this dial peer. Range is from 16 to 1007.
cid
(Optional) DLCI subchannel to be used for data on FRF.11 calls. A CID must be specified only when the session protocol is frf11-trunk. When the session protocol is cisco-switched, the CID is dynamically allocated. Range is from 4 to 255. Note
By default, CID 4 is used for data; CID 5 is used for call-control. We recommend that you select CID values between 6 and 63 for voice traffic. If the CID is greater than 63, the FRF.11 header contains an extra byte of data.
Command Default
The default for this command is enabled with no IP address or domain name defined.
Command Modes
Dial peer configuration
Command History
Release
Modification
11.3(1)T
This command was introduced.
11.3(1)MA
This command was implemented for VoFR, VoHDLC, and POTS dial peers on the Cisco MC3810.
12.0(3)XG
This command was implemented for VoFR dial peers on the Cisco 2600 series and Cisco 3600 series. The cid option was added.
12.0(4)T
This command was integrated into Cisco IOS Release12.0(4)T and implemented for VoFR and POTS dial peers on the Cisco 7200 series.
Cisco IOS Voice Command Reference
VR-1772
Cisco IOS Voice Commands: S session target (VoFR dial peer)
Usage Guidelines
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select. The syntax of this command complies with the simple syntax of mailto: as described in RFC 1738. The session target loopback command is used for testing the voice transmission path of a call. The loopback point depends on the call origin and the loopback type selected. For VoFR dial peers, the cid option is not allowed when the cisco-switched option for the session protocol command is used.
Examples
The following example configures serial interface 1/0, DLCI 100 as the session target for Voice over Frame Relay dial peer 200 (an FRF.11 dial peer) using the FRF.11 session protocol: dial-peer voice 200 vofr destination-pattern 13102221111 called-number 5552150 session protocol frf11-trunk session target serial 1/0 100 20
The following example delivers fax-mail to multiple recipients: dial-peer voice 10 mmoip session target [email protected]
Assuming that mailer.example.com is running sendmail, you can put the following information into its /etc/aliases file: marketing-information: [email protected], [email protected]
Related Commands
Command
Description
called-number
Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.
codec (dial peer)
Specifies the voice coder rate of speech for a dial peer.
cptone
Specifies a regional tone, ring, and cadence setting for an analog voice port.
destination-pattern
Specifies either the prefix or the full E.164 telephone number (depending on the dial plan) to be used for a dial peer.
dtmf-relay
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
preference
Indicates the preferred selection order of a dial peer within a hunt group.
session protocol
Establishes a VoFR protocol for calls between the local and the remote routers via the packet network.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
Cisco IOS Voice Command Reference
VR-1773
Cisco IOS Voice Commands: S session target (VoIP dial peer)
session target (VoIP dial peer) To designate a network-specific address to receive calls from a VoIP or VoIPv6 dial peer, use the session target command in dial peer configuration mode. To reset to the default, use the no form of this command. Cisco 1751, Cisco 3725, Cisco 3745, and Cisco AS5300
Configures the router to obtain the session target via DHCP. Note
The dhcp option can be made available only if the Session Initiation Protocol (SIP) is used as the session protocol. To enable SIP, use the session protocol (dial peer) command.
ipv4:destination-address
Configures the IP address of the dial peer to receive calls. The colon is required.
ipv6:[destination-address]
Configures the IPv6 address of the dial peer to receive calls. Square brackets must be entered around the IPv6 address. The colon is required.
dns:[$s$] hostname
Configures the host device housing the domain name system (DNS) server that resolves the name of the dial peer to receive calls. The colon is required. Use one of the following macros with this keyword when defining the session target for VoIP peers:
Cisco IOS Voice Command Reference
VR-1774
•
$s$.—(Optional) Source destination pattern is used as part of the domain name.
•
$d$.—(Optional) Destination number is used as part of the domain name.
•
$e$.—(Optional) Digits in the called number are reversed and periods are added between the digits of the called number. The resulting string is used as part of the domain name.
•
$u$.—(Optional) Unmatched portion of the destination pattern (such as a defined extension number) is used as part of the domain name.
•
hostname—String that contains the complete hostname to be associated with the target address; for example, serverA.example1.com.
Cisco IOS Voice Commands: S session target (VoIP dial peer)
enum:table-num
Configures ENUM search table number. Range is from 1 to 15. The colon is required.
loopback:rtp
Configures all voice data to loop back to the source. The colon is required.
ras
Configures the registration, admission, and status (RAS) signaling function protocol. A gatekeeper is consulted to translate the E.164 address into an IP address.
sip-server
Configures the global SIP server is the destination for calls from the dial peer.
:port
(Optional) Port number for the dial-peer address. The colon is required.
settlement provider-number
Configures the settlement server as the target to resolve the terminating gateway address. •
registrar
The provider-number argument specifies the provider IP address.
Specifies to route the call to the registrar end point. •
The registrar keyword is available only for SIP dial peers.
Command Default
No IP address or domain name is defined.
Command Modes
Dial peer configuration (config-dial-peer)
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 2600 series and Cisco 3600 series.
12.0(3)T
This command was modified. This command was implemented on the Cisco AS5300. The ras keyword was added.
12.0(4)XJ
This command was implemented for store-and-forward fax on the Cisco AS5300.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T. The settlement and sip-server keywords were added.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 was not included in this release.
12.2(11)T
This command was implemented on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850. The enum keyword was added.
12.4(22)T
This command was modified. Support for IPv6 was added.
12.4(22)YB
This command was modified. The dhcp keyword was added.
15.0(1)M
This command was integrated into Cisco IOS Release 15.0(1)M.
15.1(3)T
This command was modified. The registrar keyword was added.
Cisco IOS Voice Command Reference
VR-1775
Cisco IOS Voice Commands: S session target (VoIP dial peer)
Usage Guidelines
Use the session target command to specify a network-specific destination for a dial peer to receive calls from the current dial peer. You can select an option to define a network-specific address or domain name as a target, or you can select one of several methods to automatically determine the destination for calls from the current dial peer. Use the session target dns command with or without the specified macros. Using the optional macros can reduce the number of VoIP dial-peer session targets that you must configure if you have groups of numbers associated with a particular router. The session target enum command instructs the dial peer to use a table of translation rules to convert the dialed number identification service (DNIS) number into a number in E.164 format. This translated number is sent to a DNS server that contains a collection of URLs. These URLs identify each user as a destination for a call and may represent various access services, such as SIP, H.323, telephone, fax, e-mail, instant messaging, and personal web pages. Before assigning the session target to the dial peer, configure an ENUM match table with the translation rules using the voice enum-match-table command in global configuration mode. The table is identified in the session target enum command with the table-num argument. Use the session target loopback command to test the voice transmission path of a call. The loopback point depends on the call origin. Use the session target dhcp command to specify that the session target host is obtained via DHCP. The dhcp option can be made available only if the SIP is being used as the session protocol. To enable SIP, use the session protocol (dial peer) command. In Cisco IOS Release 12.1(1)T the session target command configuration cannot combine the target of RAS with the settle-call command. For the session target settlement provider-number command, when the VoIP dial peers are configured for a settlement server, the provider-number argument in the session target and settle-call commands should be identical. Use the session target sip-server command to name the global SIP server interface as the destination for calls from the dial peer. You must first define the SIP server interface by using the sip-server command in SIP user-agent (UA) configuration mode. Then you can enter the session target sip-server option for each dial peer instead of having to enter the entire IP address for the SIP server interface under each dial peer. After the SIP endpoints are registered with the SIP registrar in the hosted unified communications (UC), you can use the session target registrar command to route the call automatically to the registrar end point. You must configure the session target command on a dial pointing towards the end point.
Examples
The following example shows how to create a session target using DNS for a host named “voicerouter” in the domain example.com: dial-peer voice 10 voip session target dns:voicerouter.example.com
The following example shows how to create a session target using DNS with the optional $u$. macro. In this example, the destination pattern ends with four periods (.) to allow for any four-digit extension that has the leading number 1310555. The optional $u$. macro directs the gateway to use the unmatched portion of the dialed number—in this case, the four-digit extension—to identify a dial peer. The domain is “example.com.” dial-peer voice 10 voip destination-pattern 1310555.... session target dns:$u$.example.com
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S session target (VoIP dial peer)
The following example shows how to create a session target using DNS, with the optional $d$. macro. In this example, the destination pattern has been configured to 13105551111. The optional macro $d$. directs the gateway to use the destination pattern to identify a dial peer in the “example.com” domain. dial-peer voice 10 voip destination-pattern 13105551111 session target dns:$d$.example.com
The following example shows how to create a session target using DNS, with the optional $e$. macro. In this example, the destination pattern has been configured to 12345. The optional macro $e$. directs the gateway to do the following: reverse the digits in the destination pattern, add periods between the digits, and use this reverse-exploded destination pattern to identify the dial peer in the “example.com” domain. dial-peer voice 10 voip destination-pattern 12345 session target dns:$e$.example.com
The following example shows how to create a session target using an ENUM match table. It indicates that calls made using dial peer 101 should use the preferential order of rules in enum match table 3: dial-peer voice 101 voip session target enum:3
The following example shows how to create a session target using DHCP: dial-peer voice 1 voip session protocol sipv2 voice-class sip outbound-proxy dhcp session target dhcp
The following example shows how to create a session target using RAS: dial-peer voice 11 voip destination-pattern 13105551111 session target ras
The following example shows how to create a session target using settlement: dial-peer voice 24 voip session target settlement:0
The following example shows how to create a session target using IPv6 for a host at 2001:10:10:10:10:10:10:230a:5090: dial-peer voice 4 voip destination-pattern 5000110011 session protocol sipv2 session target ipv6:[2001:0DB8:10:10:10:10:10:230a]:5090 codec g711ulaw
The following example shows how to configure Cisco Unified Border Element (UBE) to route a call to the registering end point: dial-peer voice 4 voip session target registrar
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Cisco IOS Voice Commands: S session target (VoIP dial peer)
Related Commands
Command
Description
destination-pattern
Specifies either the prefix or the full E.164 telephone number (depending on the dial plan) to be used for a dial peer.
dial-peer voice
Enters dial peer configuration mode and specifies the method of voice-related encapsulation.
session protocol (dial peer)
Specifies a session protocol for calls between local and remote routers using the packet network dial peer configuration mode.
settle-call
Specifies that settlement is to be used for the specified dial peer, regardless of the session target type.
sip-server
Defines a network address for the SIP server interface.
voice enum-match-table
Initiates the ENUM match table definition.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S session transport
session transport To configure a VoIP dial peer to use TCP or User Datagram Protocol (UDP) as the underlying transport layer protocol for Session Initiation Protocol (SIP) messages, use the session transport command in dial peer configuration mode. To reset to the default (udp keyword), use the no form of this command. session transport {system | tcp tls | udp} no session transport {system | tcp tls | udp}
Syntax Description
Command Default
Note
system
The SIP dial peer defers to the voice service VoIP session transport.
tcp tls
The SIP dial peer uses Transport Layer Security (TLS) over the TCP transport layer protocol.
udp
The SIP dial peer uses the UDP transport layer protocol. This is the default.
UDP
The transport protocol specified with the transport command must match the one specified with this command.
Command Modes
Dial peer configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
12.4(6)T
The tls keyword was added to the command.
Usage Guidelines
Use the show sip-ua status command to ensure that the transport protocol that you set using this command matches the protocol set using the transport command. The transport command is used in dial peer configuration mode to specify the SIP transport method, either UDP, TCP, or TLS over TCP.
Examples
The following example shows a VoIP dial peer configured to use TLS over TCP as the underlying transport layer protocol for SIP messages: dial-peer voice 102 voip session transport tcp tls
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Cisco IOS Voice Commands: S session transport
The following example shows a VoIP dial peer configured to use UDP as the underlying transport layer protocol for SIP messages: dial-peer voice 102 voip session transport udp
Related Commands
Command
Description
show sip-ua status
Displays the status of SIP call service on a SIP gateway.
transport
Configures the SIP user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S session transport (H.323 voice-service)
session transport (H.323 voice-service) To configure the underlying transport layer protocol for H.323 messages to be used across all VoIP dial peers, use the session transport command in H.323 voice service configuration mode. To reset the default value, use the no form of this command. session transport {udp | tcp [calls-per-connection value]} no session transport
Syntax Description
udp
Configures the H.323 dial peer to use the UDP transport layer protocol.
tcp
Configures the H.323 dial peer to use the TCP transport layer protocol. This is the default.
calls-per-connection
Configures the number of calls multiplexed into a single TCP connection.
value
The number of calls. The range is from 1 to 9999. The default is 5.
Command Default
TCP is the default session transport protocol; the default calls-per-connection value is 5.
Command Modes
H.323 voice service configuration
Command History
Release
Modification
12.2(1)T
This command was introduced for session initiation protocol (SIP) dial peers.
12.2(2)XA
This command was modified to include support for H323 dial peers and to include the calls-per-connection keyword.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
Examples
The following example shows a dial peer configured to use the UDP transport layer protocol. Router(conf-voi-serv)# h323 Router(conf-serv-h323)# session transport udp
Related Commands
Command
Description
h323
Enables H.323 voice service configuration commands.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S session transport (SIP)
session transport (SIP) To configure the underlying transport layer protocol for SIP messages to transport layer security over TCP (TLS over TCP) or User Datagram Protocol (UDP), use the session transport command in SIP configuration mode. To reset the value of this command to the default, use the no form of this command. session transport {udp | tcp tls} no session transport {udp | tcp tls}
Syntax Description
udp
Configure SIP messages to use the UDP transport layer protocol. This is the default.
tcp tls
Configure SIP messages to use the TLS over TCP transport layer protocol.
Command Default
The default for the command is UDP.
Command Modes
SIP configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced in SIP configuration mode.
12.2(2)XB2
This command was implemented on the Cisco AS5850 platform.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and support was added for the Cisco 3700 series. Cisco AS5300, Cisco AS5350, Cisco AS5850, and Cisco AS5400 platforms were not supported in this release.
12.2(11)T
Support was added for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms.
12.4(6)T
The tls keyword was added to the command.
Usage Guidelines
Use the show sip-ua status command to verify that the transport protocol set with the session transport command matches the protocol set using the transport command in SIP user agent configuration mode.
Examples
The following example configures the underlying transport layer protocol for SIP messages to UDP: voice service voip sip session transport udp
The following example configures the underlying transport layer protocol for SIP messages to TLS over TCP: voice service voip sip session transport tcp tls
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S session transport (SIP)
Related Commands
Command
Description
show sip-ua status
Displays the status of SIP call service on a SIP gateway.
transport
Configures the SIP gateway for SIP signaling messages on inbound calls through the SIP TCP or UDP socket.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S session-set
session-set To create a Signlaing System 7 (SS7)-link-to-SS7-session-set association or to associate an SS7 link with an SS7 session set on the Cisco 2600-based Signaling Link Terminal (SLT), enter the session-set command in global configuration mode. To remove the link from its current SS7 session set and to add it to SS7 session set 0 (the default), use the no form of this command. session-set session-set-id no session-set
Syntax Description
session-set-id
Command Default
SS7 session set 0
Command Modes
Global configuration
Command History
Release
Modification
12.2(15)T
This command was introduced on the Cisco 2600-based SLT.
Usage Guidelines
SS7 session ID. Valid values are 0 and 1. Default is 0.
On Cisco AS5350 and Cisco AS5400 platforms, the channel-id command is used to create an SS7-link-to-SS7-session-set association on the Cisco SLT. The Cisco 26xx platforms do not support the channel-id command, so channel IDs on the Cisco 26xx-based SLT are implicitly assigned on the basis of the slot location of the WAN interface card (WIC) and the channel group ID used to create the SS7 link. If this command is omitted, the link is implicitly added to the SS7 session set 0, which is the default.
Examples
The following example shows how the session-set command is used to add the associated SS7 link to an SS7 session set: session-set 1
The following example shows how the no session-set command is used to remove the link from its current SS7 session set and add it to SS7 session set 0, which is the default: no session-set
Related Commands
Command
Description
channel-id
Assigns a session channel ID to a Signaling System 7 (SS7) serial link or assign an SS7 link to an SS7 session set on a Cisco AS5350 or Cisco AS5400.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S set
set To create a fault-tolerant or non-fault-tolerant session set with the client or server option, use the set command in backhaul session-manager configuration mode. To delete the set, use the no form of this command. set set-name {client | server} {ft | nft} no set set-name {client | server} {ft | nft}
Syntax Description
set-name
Session-set name.
client
The session set operates as a client. Select this option for signaling backhaul.
server
The session set operates as a server.
ft
Fault-tolerant operation. Select fault-tolerant if this session set can contain more than one session group, with each session group connecting the gateway to a different Cisco VSC3000. Fault-tolerance allows the system to operate properly if a session group in the session set fails.
nft
Non-fault-tolerant operation. Select non-fault-tolerant if this session set contains only one session group (which connects the gateway to a single Cisco VSC3000).
Command Default
No default behavior or values
Command Modes
Backhaul session-manager configuration
Command History
Release
Modification
12.1(1)T
This command was introduced.
12.2(4)T
This command was implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.2(2)XB
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the Cisco IAD2420 series. Support for on the Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5350, Cisco AS5400 and Cisco AS5850 in this release.
Usage Guidelines
Multiple session groups can be associated with a session set. For signaling backhaul, session sets should be configured to operate as clients. A session set cannot be deleted unless all session groups associated with the session set are deleted first.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S set
Examples
The following example sets the client set named “set1” as fault-tolerant: Router(config-bsm)# set set1 client ft
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S set http client cache stale
set http client cache stale To set the status of all entries in the HTTP client cache to stale, use the set http client cache stale command in global configuration mode. set http client cache stale
Syntax Description
This command has no arguments or keywords.
Command Default
Entries in the HTTP client cache are not marked stale manually.
Command Modes
Global configuration (config)
Command History
Release
Modification
12.4(15)T
This command was introduced.
12.4(20)T
This command was integrated into Cisco IOS Release 12.4(20)T.
Usage Guidelines
Use this command to force the HTTP client to check with the server to see if an updated version of the file exists when any cached entries are requested by the VoiceXML application. If the router is in nonstreaming mode, a conditional reload is sent to the HTTP server. If the router is in streaming mode, an unconditional reload is sent for the refresh. Regardless of which mode the router is in, the VoiceXML application is guaranteed to receive the most up-to-date file when you use the set http client cache stale command. The show http client cache command shows a pound sign (#) next to the age of entries that are marked stale manually.
Examples
The following example sets the status of all entries in the HTTP client cache to stale: Router# set http client cache stale
Related Commands
Command
Description
show http client cache Displays information about the entries contained in the HTTP client cache.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S set pstn-cause
set pstn-cause To map an incoming PSTN cause code to a Session Initiation Protocol (SIP) error status code, use the set pstn-cause command in SIP UA configuration mode. To reset to the default, use the no form of this command. set pstn-cause value sip-status value no set pstn-cause
Syntax Description
Command Default
pstn-cause value
PSTN cause code. Range is from 1 to 127
sip-status value
SIP status code that is to correspond with the PSTN cause code. Range is from 400 to 699.
The default mappings defined in the following table are used: Table 42
Default PSTN Cause Codes Mapped to SIP Events
PSTN Cause Code
Description
SIP Event
1
Unallocated number
404 Not found
2
No route to specified transit network
404 Not found
3
No route to destination
404 Not found
17
User busy
486 Busy here
18
No user responding
480 Temporarily unavailable
19
No answer from the user
20
Subscriber absent
21
Call rejected
403 Forbidden
22
Number changed
410 Gone
26
Non-selected user clearing
404 Not found
27
Destination out of order
404 Not found
28
Address incomplete
484 Address incomplete
29
Facility rejected
501 Not implemented
31
Normal, unspecified
404 Not found
34
No circuit available
503 Service unavailable
38
Network out of order
503 Service unavailable
41
Temporary failure
503 Service unavailable
42
Switching equipment congestion
503 Service unavailable
47
Resource unavailable
503 Service unavailable
55
Incoming class barred within the Closed User Group (CUG)
403 Forbidden
57
Bearer capability not authorized
403 Forbidden
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S set pstn-cause
Table 42
Default PSTN Cause Codes Mapped to SIP Events (continued)
PSTN Cause Code
Description
SIP Event
58
Bearer capability not currently available
501 Not implemented
65
Bearer capability not implemented
501 Not implemented
79
Service or option not implemented
501 Not implemented
87
User not member of the Closed User Group (CUG)
503 Service unavailable
88
Incompatible destination
400 Bad request
95
Invalid message
400 Bad request
102
Recover on Expires timeout
408 Request timeout
111
Protocol error
400 Bad request
Any code other than those listed above
500 Internal server error
Command Modes
SIP UA configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB2
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. Support for on the Cisco AS5300 Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
Usage Guidelines
A PSTN cause code can be mapped only to one SIP status code at a time.
Examples
The following example maps a SIP status code to correspond to a PSTN cause code: Router(config)# sip-ua Router(config-sip-ua)# set pstn-cause 111 sip-status 400 Router(config-sip-ua)# exit
Related Commands
Command
Description
set sip-status
Sets an incoming SIP error status code to a PSTN release cause code.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S set sip-status
set sip-status To map an incoming Session Initiation Protocol (SIP) error status code to a PSTN cause code, use the set sip-status command in SIP UA configuration mode. To reset to the default, use the no form of this command. set sip-status value pstn-cause value no set sip-status
Syntax Description
Command Default
sip-status value
SIP status code. Range is from 400 to 699.
pstn-cause value
PSTN cause code that is to correspond with the SIP status code. Range is from 1 to 127.
The default mappings defined in the following table are used: Table 43
Default SIP Events Mapped to PSTN Cause Codes
SIP Event
PSTN Cause Code
Description
400 Bad request
127
Interworking, unspecified
401 Unauthorized
57
Bearer capability not authorized
402 Payment required
21
Call rejected
403 Forbidden
57
Bearer capability not authorized
404 Not found
1
Unallocated number
405 Method not allowed
127
Interworking, unspecified
407 Proxy authentication required
21
Call rejected
408 Request timeout
102
Recover on Expires timeout
409 Conflict
41
Temporary failure
410 Gone
1
Unallocated number
411 Length required
127
Interworking, unspecified
415 Unsupported media type
79
Service or option not available
420 Bad extension
127
Interworking, unspecified
480 Temporarily unavailable
18
No user response
481 Call leg does not exist
127
Interworking, unspecified
484 Address incomplete
28
Address incomplete
485 Address ambiguous
1
Unallocated number
406 Not acceptable
413 Request entity too long 414 Request URI (URL) too long
482 Loop detected 483 Too many hops
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S set sip-status
Table 43
Default SIP Events Mapped to PSTN Cause Codes (continued)
SIP Event
PSTN Cause Code
Description
486 Busy here
17
User busy
487 Request canceled
127
Interworking, unspecified
488 Not acceptable here
127
Interworking, unspecified
500 Internal server error
41
Temporary failure
501 Not implemented
79
Service or option not implemented
502 Bad gateway
38
Network out of order
503 Service unavailable
63
Service or option unavailable
504 Gateway timeout
102
Recover on Expires timeout
505 Version not implemented
127
Interworking, unspecified
580 Precondition failed
47
Resource unavailable, unspecified
600 Busy everywhere
17
User busy
603 Decline
21
Call rejected
604 Does not exist anywhere
1
Unallocated number
606 Not acceptable
58
Bearer capability not currently available
Command Modes
SIP UA configuration
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB2
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
Usage Guidelines
A SIP status code can be mapped to many PSTN cause codes. For example, 503 can be mapped to 34, 38, and 58.
Examples
The following example maps a PSTN cause code to correspond to a SIP status code: Router(config)# sip-ua Router(config-sip-ua)# set sip-status 400 pstn-cause 16
Related Commands
Command
Description
set pstn-cause
Sets an incoming PSTN cause code to a SIP error status code.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S settle-call
settle-call To force a call to be authorized with a settlement server that uses the address resolution method specified in the session target command, use the settle-call command in dial peer configuration mode. To ensure that no authorization is performed by a settlement server, use the no form of this command. settle-call provider-number no settle-call provider-number
Syntax Description
provider-number
Digit defining the ID of a particular settlement server. The only valid entry is 0. Note
If session target type is settlement, the provider-number argument in the session target and settle-call commands should be identical.
Command Default
No default behavior or values.
Command Modes
Dial peer configuration
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
Usage Guidelines
With the session target command, a dial peer can determine the address of the terminating gateway through the ipv4, dns, ras, and settlement keywords. If the session target is not settlement, and the settle-call provider-number argument is set, the gateway resolves address of the terminating gateway using the specified method and then requests the settlement server to authorize that address and create a settlement token for that particular address. If the server cannot authorize the terminating gateway address suggested by the gateway, the call fails. Do not combine the session target types ras and settle-call. Combination of session target types is not supported.
Examples
The following example sets a call to be authorized with a settlement server that uses the address resolution method specified in the session target: dial-peer voice 10 voip destination-pattern 1408....... session target ipv4:172.22.95.14 settle-call 0
Related Commands
Command
Description
session target
Specifies a network-specific address for a specified dial peer.
Cisco IOS Voice Command Reference
VR-1792
Cisco IOS Voice Commands: S settlement
settlement To enter settlement configuration mode and specify the attributes specific to a settlement provider, use the settlement command in global configuration mode. To disable the settlement provider, use the no form of this command. settlement provider-number no settlement provider-number
Syntax Description
provider-number
Command Default
0
Command Modes
Global configuration
Command History
Release
Modification
12.0(4)XH1
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Digit that defines a particular settlement server. The only valid entry is 0.
Usage Guidelines
The variable provider-number defines a particular settlement provider. For Cisco IOS Release 12.1, only one clearinghouse per system is allowed, and the only valid value for provider-number is 0.
Examples
This example enters settlement configuration mode: settlement 0
Related Commands
Command
Description
connection-timeout
Configures the length of time for which a connection is maintained after a communication exchange is completed.
customer-id
Identifies a carrier or ISP with a settlement provider.
device-id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
max-connection
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
response-timeout
Configures the maximum time to wait for a response from a server.
retry-delay
Sets the time between attempts to connect with the settlement provider.
retry-limit
Sets the connection retry limit.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S settlement
Command
Description
session-timeout
Sets the interval for closing the connection when there is no input or output traffic.
show settlement
Displays the configuration for all settlement server transactions.
shutdown
Brings up the settlement provider.
type
Configures an SAA-RTR operation type.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S settlement roam-pattern
settlement roam-pattern To configure a pattern that must be matched to determine if a user is roaming, use the settlement roam-pattern command in global configuration mode. To delete a particular pattern, use the no form of this command. settlement provider-number roam-pattern pattern {roaming | noroaming} no settlement provider-number roam-pattern pattern {roaming | noroaming}
Syntax Description
provider-number
Digit defining the ID of particular settlement server. The only valid entry is 0.
pattern
User account pattern.
roaming
Specifies that a user is roaming.
noroaming
Specifies that a user is not roaming.
Command Default
No default pattern is configured.
Command Modes
Global configuration (config)
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
Usage Guidelines
Multiple roam patterns can be entered on one gateway.
Examples
The following example shows how to configure a pattern that determines if a user is roaming: settlement settlement settlement settlement
Enables the roaming capability for a settlement provider.
settlement
Enters settlement configuration mode.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S sgcp
sgcp To start and allocate resources for the Simple Gateway Control Protocol (SGCP) daemon, use the sgcp command in global configuration mode. To terminate all calls, release all allocated resources, and kill the SGCP daemon, use the no form of this command. sgcp no sgcp
Syntax Description
This command has no arguments or keywords.
Command Default
The SGCP daemon is not enabled.
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the Cisco AS5300 only and was not generally available.
12.0(7)XK
This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was implemented on the Cisco 3600 series and Cisco MC3810.
Usage Guidelines
When the SGCP daemon is not active, all SGCP messages are ignored. When you enter the no sgcp command, the SGCP process is removed.
Note
Examples
After you enter the no sgcp command, you must save the configuration and reboot the router for the disabling of SGCP to take effect.
The following example enables the SGCP daemon: sgcp
The following example disables the SGCP daemon: no sgcp
Related Commands
Command
Description
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S sgcp
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
Specifies the number of times to retry sending “notify” and “delete” messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
Cisco IOS Voice Command Reference
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Cisco IOS Voice Commands: S sgcp call-agent
sgcp call-agent To define the IP address of the default Simple Gateway Control Protocol (SGCP) call agent in the router configuration file, use the sgcp call-agent command in global configuration mode. To remove the IP address of the default SGCP call agent from the router configuration, use the no form of this command. sgcp call-agent ipaddress [:udp port] no sgcp call-agent ipaddress
Syntax Description
ipaddress
IP address or hostname of the call agent.
:udp port
(Optional) UDP port of the call agent.
Command Default
No IP address is configured.
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the Cisco AS5300 only and was not generally available.
12.0(7)XK
This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was implemented on the Cisco 3600 series and Cisco MC3810.
Usage Guidelines
This command defines the IP address of the default SGCP call agent to which the router sends an initial RSIP (Restart In Progress) packet when the router boots up. This is used for initial bootup only before the SGCP call agent contacts the router acting as the gateway. When you enter the no sgcp call-agent command, only the IP address of the default SGCP call agent is removed.
Examples
The following example enables SGCP and specifies the IP address of the call agent: sgcp sgcp call-agent 209.165.200.225
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
Cisco IOS Voice Command Reference
VR-1798
Cisco IOS Voice Commands: S sgcp call-agent
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
Specifies the number of times to retry sending “notify” and “delete” messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
Cisco IOS Voice Command Reference
VR-1799
Cisco IOS Voice Commands: S sgcp graceful-shutdown
sgcp graceful-shutdown To block all new calls and gracefully terminate all existing calls (wait for the caller to end the call), use the sgcp graceful-shutdown command in global configuration mode. To unblock all calls and allow new calls to go through, use the no form of this command. sgcp graceful-shutdown no sgcp graceful-shutdown
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values.
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the Cisco AS5300 and was not generally available.
12.0(7)XK
This command was implemented on the Cisco MC3810 and Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was implemented on the Cisco 3600 series and Cisco MC3810.
Usage Guidelines
Once you issue this command, all requests for new connections (CreateConnection requests) are denied. All existing calls are maintained until users terminate them, or until you enter the no sgcp command. When the last active call is terminated, the SGCP daemon is terminated, and all resources allocated to it are released.
Examples
The following example blocks all new calls and terminates existing calls: sgcp graceful-shutdown
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
sgcp quarantine-buffer disable Disables the SGCP quarantine buffer.
Cisco IOS Voice Command Reference
VR-1800
Cisco IOS Voice Commands: S sgcp graceful-shutdown
Command
Description
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete” messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
Cisco IOS Voice Command Reference
VR-1801
Cisco IOS Voice Commands: S sgcp max-waiting-delay
sgcp max-waiting-delay To set the Simple Gateway Control Protocol (SGCP) maximum waiting delay to prevent restart avalanches, use the sgcp max-waiting-delay command in global configuration mode. To reset to the default, use the no form of this command. sgcp max-waiting-delay delay no sgcp max-waiting-delay delay
Syntax Description
delay
Command Default
3,000 ms
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the Cisco AS5300, and was not generally available.
12.0(7)XK
This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
Examples
Maximum waiting delay (MWD), in milliseconds. Range is from 0 to 600000. Default is 3000.
The following example sets the maximum wait delay value to 40 ms: sgcp max-waiting-delay 40
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
Specifies the number of times to retry sending “notify” and “delete” messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
Cisco IOS Voice Command Reference
VR-1802
Cisco IOS Voice Commands: S sgcp max-waiting-delay
Command
Description
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
Cisco IOS Voice Command Reference
VR-1803
Cisco IOS Voice Commands: S sgcp modem passthru
sgcp modem passthru To enable Simple Gateway Control Protocol (SGCP) modem or fax pass-through, use the sgcp modem passthru command in global configuration mode. To disable SGCP modem or fax pass-through, use the no form of this command. sgcp modem passthru {ca | cisco | nse} no sgcp modem passthru {ca | cisco | nse}
Cisco-proprietary upspeed method based on the protocol.
nse
NSE-based modem upspeed method.
Command Default
SGCP modem or fax pass-through is disabled by default.
Command Modes
Global configuration.
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco MC3810 and the Cisco 3600 series (except the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
Usage Guidelines
You can use this command for fax pass-through because the answer tone can come from either modem or fax transmissions. The upspeed method is the method used to dynamically change the codec type and speed to meet network conditions. If you use the nse option, you must also configure the sgcp tse payload command.
Examples
The following example configures SGCP modem pass-through using the call-agent upspeed method: sgcp modem passthru ca
The following example configures SGCP modem pass-through using the proprietary Cisco upspeed method: sgcp modem passthru cisco
The following example configures SGCP modem pass-through using the NSE-based modem upspeed: sgcp modem passthru nse sgcp tse payload 110
Cisco IOS Voice Command Reference
VR-1804
Cisco IOS Voice Commands: S sgcp modem passthru
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
Specifies the number of times to retry sending “notify” and “delete” messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
Cisco IOS Voice Command Reference
VR-1805
Cisco IOS Voice Commands: S sgcp quarantine-buffer disable
sgcp quarantine-buffer disable To disable the Simple Gateway Control Protocol (SGCP) quarantine buffer, use the sgcp quarantine-buffer disable command in global configuration mode. To reenable the SGCP quarantine buffer, use the no form of this command. sgcp quarantine-buffer disable no sgcp quarantine-buffer disable
Syntax Description
This command has no arguments or keywords.
Command Default
The SGCP quarantine buffer is enabled.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was on the Cisco 3600 series and the Cisco MC3810.
Usage Guidelines
The SGCP quarantine buffer is the mechanism for buffering the SGCP events between two notification-request (RQNT) messages.
Examples
The following example disables the SGCP quarantine buffer: sgcp quarantine-buffer disable
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete” messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
Cisco IOS Voice Command Reference
VR-1806
Cisco IOS Voice Commands: S sgcp quarantine-buffer disable
Command
Description
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
Cisco IOS Voice Command Reference
VR-1807
Cisco IOS Voice Commands: S sgcp request retries
sgcp request retries To specify the number of times to retry sending notify and delete messages to the Simple Gateway Control Protocol (SGCP) call agent, use the sgcp request retries command in global configuration mode. To reset to the default, use the no form of this command. sgcp request retries count no sgcp request retries
Syntax Description
count
Command Default
3 times
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the Cisco AS5300 and was not generally available.
12.0(7)XK
This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
Usage Guidelines
Number of times that a notify and delete message is retransmitted to the SGCP call agent before it is dropped. Range is from 1 to 100. Default is 3.
The actual retry count may be different from the value you enter for this command. The retry count is also limited by the call agent. If there is no response from the call agent after 30 seconds, the gateway does not retry anymore, even though the number set using the sgcp request retries command has not been reached. The router stops sending retries after 30 seconds, regardless of the setting for this command.
Examples
The following example configures the system to send the sgcp command 10 times before dropping the request: sgcp request retries 10
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
Cisco IOS Voice Command Reference
VR-1808
Cisco IOS Voice Commands: S sgcp request retries
Command
Description
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
Cisco IOS Voice Command Reference
VR-1809
Cisco IOS Voice Commands: S sgcp request timeout
sgcp request timeout To specify how long the system should wait for a response to a request, use the sgcp request timeout command in global configuration mode. To reset to the default, use the no form of this command. sgcp request timeout timeout no sgcp request timeout
Syntax Description
timeout
Command Default
500 ms
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the Cisco AS5300 and was not generally available.
12.0(7)XK
This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
Time to wait for a response to a request, in milliseconds. Range is from 1 to 10000. Default is 500.
Usage Guidelines
This command is used for “notify” and “delete” messages, which are sent to the SGCP call agent.
Examples
The following example configures the system to wait 40 ms for a reply to a request: sgcp request timeout 40
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
Specifies the number of times to retry sending “notify” and “delete” messages to the SGCP call agent.
Cisco IOS Voice Commands: S sgcp request timeout
Command
Description
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
Cisco IOS Voice Command Reference
VR-1811
Cisco IOS Voice Commands: S sgcp restart
sgcp restart To trigger the router to send a Restart in Progress (RSIP) message to the Simple Gateway Control Protocol (SGCP) call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller, use the sgcp restart command in global configuration mode. To reset to the default, use the no form of this command. sgcp restart {delay delay | notify} no sgcp restart {delay delay | notify}
Syntax Description
delay delay
Restart delay, in milliseconds. Range is from 0 to 600. Default is 0.
notify
Restarts notification upon the SGCP/digital interface state transition.
Command Default
0 ms
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco MC3810 and the Cisco 3600 series (except the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
Usage Guidelines
Use this command to send RSIP messages from the router to the SGCP call agent. RSIP messages are used to synchronize the router and the call agent. RSIP messages are also sent when the sgcp command is entered to enable the SGCP daemon. You must enter the notify option to enable RSIP messages to be sent.
Examples
The following example configures the system to wait 40 ms before restarting SGCP: sgcp restart delay 40
The following example configures the system to send an RSIP notification to the SGCP call agent when the T1 controller state changes: sgcp restart notify
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
Cisco IOS Voice Command Reference
VR-1812
Cisco IOS Voice Commands: S sgcp restart
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
Specifies the number of times to retry sending “notify” and “delete” messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm method.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
Cisco IOS Voice Command Reference
VR-1813
Cisco IOS Voice Commands: S sgcp retransmit timer
sgcp retransmit timer To configure the Simple Gateway Control Protocol (SGCP) retransmission timer to use a random algorithm, use the sgcp retransmit timer command in global configuration mode. To reset to the default, use the no form of this command. sgcp retransmit timer {random} no sgcp retransmit timer {random}
Syntax Description
random
Command Default
The SGCP retransmission timer does not use a random algorithm.
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco 3600 series and the Cisco MC3810 in a private release that was not generally available.
12.1(2)T
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
Usage Guidelines
SGCP retransmission timer uses a random algorithm.
Use this command to enable the random algorithm component of the retransmission timer. For example, if the retransmission timer is set to 200 ms, the first retransmission timer is 200 ms, but the second retransmission timer picks up a timer value randomly between either 200 or 400. The third retransmission timer picks up a timer value randomly of 200, 400, or 800 as shown below: •
First retransmission timer: 200
•
Second retransmission timer: 200 or 400
•
Third retransmission timer: 200, 400, or 800
•
Fourth retransmission timer: 200, 400, 800, or 1600
•
Fifth retransmission timer: 200, 400, 800, 1600, or 3200 and so on.
After 30 seconds, the retransmission timer no longer retries.
Examples
The following example sets the retransmission timer to use a random algorithm: sgcp retransmit timer random
Cisco IOS Voice Command Reference
VR-1814
Cisco IOS Voice Commands: S sgcp retransmit timer
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
Specifies the number of times to retry sending “notify” and “delete” messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
sgcp timer
Configures how the gateway detects the RTP stream host.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
Cisco IOS Voice Command Reference
VR-1815
Cisco IOS Voice Commands: S sgcp timer
sgcp timer To configure how the gateway detects the Real-Time Transport Protocol (RTP) stream lost, use the sgcp timer command in global configuration mode. To reset to the default, use the no form of this command. sgcp timer {receive-rtcp timer | rtp-nse timer} no sgcp timer {receive-rtcp timer | rtp-nse timer}
Syntax Description
receive-rtcp timer RTP Control Protocol (RTCP) transmission interval, in milliseconds. Range is from 1 to 100. Default is 5. rtp-nse timer
Command Default
RTP named signaling event (NSE) timeout, in milliseconds. Range is from 100 to 3000. Default is 200.
receive-rtcp: 5 ms rtp-nse: 200 ms
Command Modes
Global configuration
Command History
Release
Modification
12.0(5)T
This command was introduced in a private release on the Cisco AS5300 and was not generally available.
12.0(7)XK
This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
Usage Guidelines
The RTP NSE timer is used for proxy ringing (the ringback tone is provided at the originating gateway).
Examples
The following example sets the RTPCP transmission interval to 100 ms: sgcp timer receive-rtcp 100
The following example sets the NSE timeout to 1000 ms: sgcp timer rtp-nse 1000
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
Cisco IOS Voice Command Reference
VR-1816
Cisco IOS Voice Commands: S sgcp timer
Command
Description
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
Specifies the number of times to retry sending “notify” and “delete” messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm method.
sgcp tse payload
Enables Inband TSE for fax/modem operation.
Cisco IOS Voice Command Reference
VR-1817
Cisco IOS Voice Commands: S sgcp tse payload
sgcp tse payload To enable Inband Telephony Signaling Events (TSE) for fax and modem operation, use the sgcp tse payload command in global configuration mode. To reset to the default, use the no form of this command. sgcp tse payload type no sgcp tse payload type
Syntax Description
type
Command Default
0 (disabled)
Command Modes
Global configuration
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco MC3810 and the Cisco 3600 series (except the Cisco 3620) in a private release that was not generally available.
12.1(2)T
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
Usage Guidelines
TSE payload type. Range is from 96 to 119. Default is 0, meaning that the command is disabled.
Because this command is disabled by default, you must specify a TSE payload type. If you set the sgcp modem passthru command to the nse value, then you must configure this command.
Examples
The following example sets Simple Gateway Control Protocol (SGCP) modem pass-through using the NSE-based modem upspeed and the Inband Telephony Signaling Events payload value set to 110: sgcp modem passthru nse sgcp tse payload 110
Related Commands
Command
Description
sgcp
Starts and allocates resources for the SGCP daemon.
sgcp call-agent
Defines the IP address of the default SGCP call agent.
sgcp graceful-shutdown
Gracefully terminates all SGCP activity.
sgcp max-waiting-delay
Sets the SGCP maximum waiting delay to prevent restart avalanches.
sgcp modem passthru
Enables SGCP modem or fax pass-through.
sgcp quarantine-buffer disable Disables the SGCP quarantine buffer.
Cisco IOS Voice Command Reference
VR-1818
Cisco IOS Voice Commands: S sgcp tse payload
Command
Description
sgcp request retries
Specifies the number of times to retry sending “notify” and “delete” messages to the SGCP call agent.
sgcp request timeout
Specifies how long the system should wait for a response to a request.
sgcp restart
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
sgcp retransmit timer
Configures the SGCP retransmission timer to use a random algorithm method.up or down so that the call agent can synchronize
sgcp timer
Configures how the gateway detects the RTP stream host.
Cisco IOS Voice Command Reference
VR-1819
show aal2 profile
show aal2 profile To display the ATM adaptation layer 2 (AAL2) profiles configured on the system, use the show aal2 profile command in privileged EXEC mode. show aal2 profile {all {itut profile-number | atmf profile-number | custom profile-number}}
Syntax Description
all
Displays ITU-T, ATMF, and custom AAL2 profiles configured on the system.
itut
Displays ITU-T profiles configured on the system.
atmf
Displays ATMF profiles configured on the system.
custom
Displays custom profiles configured on the system.
profile-number
AAL2 profile number to display. Choices are as follows: For ITU-T: •
1 = G.711 u-law
•
2 = G.711 u-law with silence insertion descriptor (SID)
•
7 = G.711 u-law and G.729ar8
For ATMF: None. ATMF is not supported. For custom: •
100 = G.711 u-law and G.726r32
•
110 = G.711 u-law, G.726r32, and G.729ar8
Command Modes
Privileged EXEC
Command History
Release
Modification
12.1(1)XA
This command was introduced on the Cisco MC3810.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.2(2)T
This command was implemented on the Cisco 7200 series.
Usage Guidelines
This command applies to AAL2 VoATM applications on the Cisco 7200 series routers.
Examples
The following command displays all of the configured profiles in the system: Router# show aal2 profile all Printing all the Profiles in the system Profile Type: ITUT Profile Number: 1 SID Support: 0 Red enable: 1 Num entries: 1 Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15 Profile Type: ITUT Profile Number: 2 SID Support: 1 Red enable: 1 Num entries: 1
Table 44 describes significant fields shown in this output. Table 44
Related Commands
show aal2 profile all Field Descriptions
Field
Description
Coding type
Voice compression algorithm.
ITUT Profile Number
Predefined combination of one or more codec types configured for a digital signal processor (DSP).
Num entries
Number of profile elements.
Packet length
Sample size.
Profile Type
Category of codec types configured on DSP. Possible types are ITU-T, ATMF, and custom.
Red enable
Redundancy for type 3 packets.
SID Support
Silence insertion descriptor.
UUI max
Maximum sequence number on the voice packets.
UUI min
Minimum sequence number on the voice packets.
Command
Description
codec aal2-profile
Sets the codec profile for a DSP on a per-call basis.
Cisco IOS Voice Command Reference
VR-1809
show atm video-voice address
show atm video-voice address To display the network service access point (NSAP) address for the ATM interface, enter the show atm video-voice address command in privileged EXEC mode. show atm video-voice address
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.0(5)XK
This command was introduced on the Cisco MC3810.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Usage Guidelines
Use this command to review ATM interface NSAP addresses that have been assigned with the atm video aesa command and to ensure that ATM management is confirmed for those addresses.
Examples
The following example displays ATM interface NSAP addresses: Router# show atm video-voice address nsap address 47.0091810000000002F26D4901.00107B4832E1.FE 47.0091810000000002F26D4901.00107B4832E1.C8
type VOICE_AAL5 VIDEO_AAL1
ilmi status Confirmed Confirmed
Table 45 describes the significant fields shown in the output. Table 45
Related Commands
show atm video-voice address Field Descriptions
Field
Description
NSAP address
NSAP address for the ATM interface.
Type
Type of ATM interface.
ILMI status
Integrated Local management Interface (ILMI) protocol status for the ATM interface.
Command
Description
codec aal2-profile
Sets the codec profile for a DSP on a per-call basis.
Cisco IOS Voice Command Reference
VR-1810
show auto-config
show auto-config To display the current status of auto-configuration applications, use the show auto-config command in privileged EXEC mode. show auto-config [application sccp]
Syntax Description
application sccp
Command Modes
Privileged EXEC
Command History
Release
Modification
12.3(8)XY
This command was introduced on the Communication Media Module.
12.3(14)T
This command was integrated into Cisco IOS Release 12.3(14)T.
Examples
Displays the current status of only the Skinny Client Control Protocol (SCCP) application.
The following is sample output from show auto-config command: Router# show auto-config application sccp auto-config application: sccp auto-config admin state: ENABLED & ACTIVE download retries: (3) download timeout: no timeout, continuous retry server(s): 172.19.240.41 172.19.240.40 172.19.240.42 Configuration Download statistics: Download Attempted : 2 Download Successful : 2 Download Failed : 0 Configuration Attempted : 2 Configuration Successful : 2 Configuration Failed(parsing): 0 Configuration Failed(config) : 0 Configuration Error History:
Table 46 describes the significant fields shown in the display. Table 46
show auto-config Field Descriptions
Field
Description
ENABLED
Shows auto-config application: SCCP is enabled.
ACTIVE
Shows the SCCP application has registered to use auto-configuration.
timeout
Shows timeout is set to 0, continuous retry without timeout.
Cisco IOS Voice Command Reference
VR-1811
show auto-config
Related Commands
Command
Description
auto-config
Enables auto-configuration or enters auto-config application configuration mode for the SCCP application.
debug auto-config
Enables debugging for auto-configuration applications.
debug sccp config
Enables SCCP event debugging.
Cisco IOS Voice Command Reference
VR-1812
show backhaul-session-manager group
show backhaul-session-manager group To display the status, statistics, or configuration for a particular session group or all available session groups, use the show backhaul-session-manager group command in privileged EXEC mode. show backhaul-session-manager group {status | stats | cfg} {all | name group-name}
Syntax Description
status
Status for available session groups.
stats
Statistics for available session groups.
cfg
Configuration for available session groups.
all
Specified parameters for all session groups.
name group-name
A particular session group.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco AS5300.
12.2(2)T
This command was implemented on the Cisco 7200 series.
12.2(4)T
This command was implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.2(2)XB
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the Cisco IAD2420 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Examples
The following example displays statistics for all session groups: Router# show backhaul-session-manager group stats all Session-Group grp1 statistics Successful Fail-Overs :0 Un-Successful Fail-Over attempts:0 Active Pkts receive count :0 Standby Pkts receive count :0 Total PDUs dispatch err :0
Cisco IOS Voice Command Reference
VR-1813
show backhaul-session-manager group
The following example displays the current configuration for all session groups: Router# show backhaul-session-manager group cfg all Session-Group Group Name :grp1 Set Name :set1 Sessions :3 Dest:10.5.0.3 8304 Local:10.1.2.15 8304 Dest:10.5.0.3 8300 Local:10.1.2.15 8300 Dest:10.5.0.3 8303 Local:10.1.2.15 8303 RUDP Options timer cumulative ack :100 timer keepalive :1000 timer retransmit :300 timer transfer state :2000 receive max :32 cumulative ack max :3 retrans max :2 out-of-sequence max :3 auto-reset max :5
Priority:0 Priority:2 Priority:2
The following example displays the current state of all session groups. The group named “grp1” belongs to the set named “set1”. Router# show backhaul-session-manager group status all Session-Group Group Name :grp1 Set Name :set1 Status :Group-OutOfService Status (use) :Group-None
Table 47 describes the significant fields shown in the output. Table 47
show backhaul-session-manager group Field Descriptions
Field
Descrption
RUDP Options
Reliable User datagram Protocol (RUDP) options.
Status
One of the following:
Status (use)
Related Commands
Command
•
Group-OutOfService—No session in the group has been established.
•
Group-Inservice—At least one session in the group has been established.
One of the following: •
Group-Standby—The virtual switch controller (VSC) connected to the other end of this group goes into standby mode.
•
Group-Active—The VSC connected to the other end of this group is the active VSC.
•
Group-None—The VSC has not yet declared its intent.
Description
show backhaul-session-manager Displays status, statistics, or configuration of sessions. session show backhaul-session-manager Displays session groups associated with a specific session set or set all session sets.
Cisco IOS Voice Command Reference
VR-1814
show backhaul-session-manager session
show backhaul-session-manager session To display various information about a session or sessions, use the show backhaul-session-manager session command in privileged EXEC mode. show backhaul-session-manager session {all | ip ip-address}
Syntax Description
all
Information is displayed about all available sessions.
ip
Information is displayed about the session associated with this IP address only.
ip-address
IP address of the local or remote session.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco AS5300.
12.2(2)T
This command was implemented on the Cisco 7200 series.
12.2(4)T
This command was implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.2(2)XB
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the Cisco IAD2420 series. Support for the Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command was implemented on the Cisco AS5350, Cisco AS5400, and Cisco AS5850.
Examples
The following command displays information for all available sessions: Router# show backhaul-session-manager session all Session information -Session-id:35 Group:grp1 /*this session belongs to the group named 'grp1' */ Configuration: Local:10.1.2.15 , port:8303 Remote:10.5.0.3 , port:8303 Priority:2 RUDP Option:Client, Conn Id:0x2 State: Status:OPEN_WAIT, Use-status:OOS, /*see explanation below */ Statistics: # of resets:0 # of auto_resets 0 # of unexpected RUDP transitions (total) 0 # of unexpected RUDP transitions (since last reset) 0 Receive pkts - Total:0 , Since Last Reset:0 Recieve failures - Total:0 ,Since Last Reset:0 Transmit pkts - Total:0, Since Last Reset:0
Cisco IOS Voice Command Reference
VR-1815
show backhaul-session-manager session
Transmit Failures (PDU Only) Due to Blocking (Not an Error) - Total:0, Since Last Reset:0 Due to causes other than Blocking - Total:0, Since Last Reset:0 Transmit Failures (NON-PDU Only) Due to Blocking(Not an Error) - Total:0, Since Last Reset:0 Due to causes other than Blocking - Total:0, Since Last Reset:0 RUDP statistics Open failures:0 Not ready failures:0 Conn Not Open failures:0 Send window full failures:0 Resource unavailble failures:0 Enqueue failures:0
Table 48 describes significant fields shown in this output. Table 48
show backhaul-session-manager session Field Descriptions
Field
Description
State
Can be any of the following: •
OPEN—The connection is established.
•
OPEN_WAIT—The connection is awaiting establishment.
•
OPEN_XFER—Session failover is in progress for this session, which is a transient state.
•
CLOSE—The session is down, also a transient state.
The session waits a fixed amount of time and then moves to OPEN_WAIT. Use-status
Related Commands
•
OOS—The session is not being used to transport signaling traffic. Out of service (OOS) does not indicate if the connection is established.
•
IS—The session is being used currently to transport all PRI signaling traffic. In service (IS) indicates that the connection is established.
Command
Description
show backhaul-session-manager group
Displays status, statistics, or configuration of a specific session group or all session groups.
show backhaul-session-manager set
Displays session groups associated with a specific session set or all session sets.
Cisco IOS Voice Command Reference
VR-1816
Indicates whether PRI signaling traffic is currently being transported over this session. Can be either of the following:
show backhaul-session-manager set
show backhaul-session-manager set To display session groups associated with a specified session set or all session sets, use the show backhaul-session-manager set command in privileged EXEC mode. show backhaul-session-manager set {all | name session-set-name}
Syntax Description
all
All available session sets.
name session-set-name A specified session set.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.1(1)T
This command was introduced on the Cisco AS5300.
12.2(2)T
This command was implemented on the Cisco 7200 series.
12.2(4)T
This command was implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.2(2)XB
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the Cisco IAD2420 series. Support for the Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Examples
The following command displays session groups associated with all session sets: Router# show backhaul-session-manager set all
Related Commands
Command
Description
show backhaul-session-manager group
Displays status, statistics, or configuration of a specific session group or all session groups.
show backhaul-session-manager session
Displays status, statistics, or configuration of a session or all sessions.
Cisco IOS Voice Command Reference
VR-1817
show call accounting-template voice
show call accounting-template voice To display accounting template activity, use the show call accounting-template voice command in privileged EXEC mode. show call accounting-template voice [acctTempName | master | qdump | summary]
Syntax Description
acctTempName
(Optional) Name of the accounting template.
master
(Optional) Displays all vendor-specific attributes (VSAs) that are filtered by accounting templates.
qdump
(Optional) Displays template activity in the service and free queues.
summary
(Optional) Lists names of all the accounting templates and the number of attributes in each template currently being used.
Command Modes
Privileged EXEC
Command History
Release
Modification
12.2(11)T
This command was introduced on the Cisco 3660, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
Usage Guidelines
•
The show call accounting-template voice command displays the status and attributes defined in each template after it is configured.
•
The show call accounting-template voice acctTempName command displays the status of a specific template and the attributes (VSAs) that are defined for that template.
•
The show call accounting-template voice master command displays all VSAs that can be filtered by accounting templates.
•
The show call accounting-template voice qdump command displays template activity in the service (svc) and free queues. It displays the template URL, the number of legs on which a template is active, and the state of a template. – After an accounting template is defined, it is put in the svc queue to serve new incoming calls.
When a running accounting template is undefined or reloaded during an active call, the template is moved from the svc queue to the free queue and can be reused after all the active calls stop referencing it. Templates that are reloaded or undefined and that are referenced during an active call are considered to be in a “dirty” state and are called dirty templates. – To ensure that start and stop records correspond on an active call that is referencing a dirty
template, all dirty templates must be kept alive until all active calls referencing that dirty template are released. After all active calls are released, the reloaded templates are applied to the next call. •
The show call accounting-template voice summary command displays the current status of all the accounting templates that are configured. It shows if the template was loaded and if it is running successfully.
Cisco IOS Voice Command Reference
VR-1818
show call accounting-template voice
Examples
The following example displays details about two templates named “cdr1” and “cdr2”. Router# show call accounting-template voice CDR template cdr1 is running url: tftp://sanjoe/santa/abc/Templates/cdr1.cdr The last load was successful. attr: h323-call-origin (56) attr: h323-call-type (57) attr: h323-gw-id (65) attr: subscriber (79) attr: in-portgrp-id (80) attr: out-portgrp-id (81) Totally 6 attrs defined. CDR template cdr2 is running url: tftp://sanjoe/santa/abc/Templates/cdr2.cdr The last load was successful. attr: h323-call-origin (56) attr: h323-call-type (57) attr: h323-connect-time (59) attr: h323-disconnect-time (64) attr: h323-gw-id (65) attr: h323-setup-time (76) attr: h323-voice-quality (78) Totally 7 attrs defined.
The following example displays details about the template named “cdr1” only. Router# show call accounting-template voice cdr1 CDR template cdr1 is running url: tftp://sanjoe/santa/abc/Templates/cdr1.cdr The last load was successful. attr: h323-call-origin (56) attr: h323-call-type (57) attr: h323-gw-id (65) attr: subscriber (79) attr: in-portgrp-id (80) attr: out-portgrp-id (81) Totally 6 attrs defined.
The following example displays all 64 attributes that can be filtered by a template. Router# show call accounting-template voice master h323-call-origin h323-call-type h323-gw-id h323-setup-time h323-connect-time h323-disconnect-time h323-disconnect-cause . . . calling-party-category originating-line-info charge-number transmission-medium-req redirecting-number backward-call-indicators Totally 64 attributes are filterable.
Cisco IOS Voice Command Reference
VR-1819
show call accounting-template voice
The following example displays template activity in the service queue. Initially, no templates are in the dirty state. Router# show call accounting-template voice qdump name url is_dirty no_of_legs ========================================================================= cdr1 tftp://sanjoe/santa/abc 0 cdr2 tftp://sanjoe/santa/abc 0 cdr3 tftp://sanjoe/santa/abc 0
After the templates are reloaded during active calls, the display below shows the templates named “cdr1” and “cdr2” to be in a dirty state. . . . Templates in freeq cdr1 tftp://sanjoe/santa/abc cdr2 tftp://sanjoe/santa/abc
dirty dirty
1 1
The following example displays a summary of all configured accounting templates. The template named “cdr3” is not in running mode, either because it has been rejected or because it does not exist at the given URL. Router# show call accounting-template voice summary name url last_load is_running ========================================================================= cdr1 tftp://sanjoe/santa/abc success is running cdr2 tftp://sanjoe/santa/abc success is running cdr3 tftp://sanjoe/santa/abc fail is not running
Table 49 describes the fields shown in the show call accounting-template voice display. Table 49
Related Commands
show call accounting-template voice Field Descriptions
Field
Description
name
Name of the accounting template.
url
Location of the accounting template.
last_load
Describes if the accounting template was successfully or unsuccessfully loaded from its location.
is_running
Describes if the accounting template was activated after it was successfully loaded from its location.
is_dirty
Shows that the accounting template was reloaded during an active call.
no_of_legs
Number of call legs.
attr
Vendor-specific attributes (VSAs) defined in an accounting template.
Command
Description
gw-accounting aaa
Configures a new accounting template.
Cisco IOS Voice Command Reference
VR-1820
show call active fax
show call active fax To display call information for T.37 store-and-forward fax transmissions in progress, use the show call active fax command in user EXEC or privileged EXEC mode. show call active fax [brief [id identifier] | compact [duration {less seconds | more seconds}] | id identifier]
Syntax Description
brief
(Optional) Displays a truncated version of fax call information.
id identifier
(Optional) Displays only the call with the specified identifier. Range is a hex value from 1 to FFFF.
compact
(Optional) Displays a compact version of the fax call information.
duration
(Optional) Displays active calls that are longer or shorter than a specified seconds value. The arguments and keywords are as follows: •
less—Displays calls shorter than the seconds value.
•
more—Displays calls longer than the seconds value.
•
seconds—Elapsed time, in seconds. Range is from 1 to 2147483647. There is no default value.
Command Modes
User EXEC Privileged EXEC
Command History
Release
Modification
11.3(1)T
This command was introduced on the Cisco 2600 series and Cisco 3600 series.
12.0(3)XG
This command was modified. Support for Voice over Frame Relay (VoFR) was added.
12.0(4)XJ
This command was implemented for store-and-forward fax on the Cisco AS5300.
12.0(4)T
This command was implemented on the Cisco 7200 series.
12.0(7)XK
This command was implemented on the Cisco MC3810.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.1(3)T
This command was modified. This command was implemented for modem pass-through over VoIP on the Cisco AS5300.
12.1(5)XM
This command was implemented on the Cisco AS5800.
12.1(5)XM2
The command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. Support was not included for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850.
12.2(11)T
Support was added for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850.
Cisco IOS Voice Command Reference
VR-1821
show call active fax
Usage Guidelines
Release
Modification
12.3(14)T
This command was modified. T.38 fax relay call statistics were made available to Call Detail Records (CDRs) through vendor-specific attributes (VSAs) and added to the call log.
12.4(2)T
This command was modified. The LocalHostname display field was added to the VoIP call leg record.
12.4(15)T
This command was modified. The Port and BearerChannel display fields were added to the TELE call leg record of the command output.
12.4(16)
This command was modified. The Port and BearerChannel display fields were added to the TELE call leg record of the command output.
12.4(22)T
This command was modified. Command output was updated to show IPv6 information.
Use this command to display the contents of the active call table. This command displays information about call times, dial peers, connections, quality of service, and other status and statistical information for T.37 store-and-forward fax calls currently connected through the router. This command works with both on-ramp and off-ramp store-and-forward fax functions. To display information about fax relay calls in progress, use the show call active voice command.
Examples
The following is sample output from the show call active fax command: Router# show call active fax GENERIC: SetupTime=22021 ms Index=1 PeerAddress=peer one PeerSubAddress= PeerId=0 PeerIfIndex=0 LogicalIfIndex=0 ConnectTime=24284 CallState=4 CallOrigin=2 ChargedUnits=0 InfoType=10 TransmitPackets=0 TransmitBytes=0 ReceivePackets=0 ReceiveBytes=41190 MMOIP: ConnectionId[0x37EC7F41 0xB0110001 0x0 0x35C34] CallID=1 RemoteIPAddress=10.0.0.0 SessionProtocol=SMTP SessionTarget= MessageId= AccountId= ImgEncodingType=MH ImgResolution=fine AcceptedMimeTypes=2 DiscardedMimeTypes=1 Notification=None
Table 50 provides an alphabetical listing of the fields displayed in the output of the show call active fax command and a description of each field. Table 50
show call active fax Field Descriptions
Field
Description
ACOM Level
Current ACOM level for this call. ACOM is the combined loss achieved by the echo canceler, which is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call.
BearerChannel
Identification of the bearer channel carrying the call.
Buffer Drain Events
Total number of jitter buffer drain events.
Buffer Fill Events
Total number of jitter buffer fill events.
CallDuration
Length of the call, in hours, minutes, and seconds, hh:mm:ss.
CallOrigin
Call origin: answer or originate.
CallState
Current state of the call.
ChargedUnits
Total number of charging units that apply to this peer since system startup. The unit of measure for this field is hundredths of second.
CodecBytes
Payload size, in bytes, for the codec used.
Cisco IOS Voice Command Reference
VR-1823
show call active fax
Table 50
show call active fax Field Descriptions (continued)
Field
Description
CoderTypeRate
Negotiated coder rate. This value specifies the send rate of voice or fax compression to its associated call leg for this call.
ConnectionId
Global call identifier for this gateway call.
ConnectTime
Time, in milliseconds, at which the call was connected.
Consecutive-packets-lost Events Total number of consecutive (two or more) packet-loss events. Corrected packet-loss Events
Total number of packet-loss events that were corrected using the RFC 2198 method.
Dial-Peer
Tag of the dial peer sending this call.
EchoCancellerMaxReflector=64 The location of the largest reflector, in milliseconds (ms). The reflector size does not exceed the configured echo path capacity. For example, if 32 ms is configured, the reflector does not report beyond 32 ms. ERLLevel
Current echo return loss (ERL) level for this call.
FaxTxDuration
Duration of fax transmission from this peer to the voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value.
GapFillWithInterpolation
Duration of a voice signal played out with a signal synthesized from parameters, or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.
GapFillWithPrediction
Duration of the voice signal played out with signal synthesized from parameters, or samples of data preceding in time, because voice data was lost or not received in time from the voice gateway for this call. Examples of such pullout are frame-eraser and frame-concealment strategies in G.729 and G.723.1 compression algorithms.
GapFillWithRedundancy
Duration of a voice signal played out with a signal synthesized from available redundancy parameters because voice data was lost or not received in time from the voice gateway for this call.
GapFillWithSilence
Duration of a voice signal replaced with silence because voice data was lost or not received in time for this call.
GENERIC
Generic or common parameters, that is, parameters that are common for VoIP and telephony call legs.
H323 call-legs
Total H.323 call legs for which call records are available.
HiWaterPlayoutDelay
High-water-mark Voice Playout FIFO Delay during this call, in ms.
Index
Dial peer identification number.
InfoActivity
Active information transfer activity state for this call.
InfoType
Information type for this call; for example, voice or fax.
InSignalLevel
Active input signal level from the telephony interface used by this call.
Last Buffer Drain/Fill Event
Elapsed time since the last jitter buffer drain or fill event, in seconds.
LocalHostname
Local hostnames used for locally generated gateway URLs.
Cisco IOS Voice Command Reference
VR-1824
show call active fax
Table 50
show call active fax Field Descriptions (continued)
Field
Description
LogicalIfIndex
Index number of the logical interface for this call.
LoWaterPlayoutDelay
Low-water-mark Voice Playout FIFO Delay during this call, in ms.
LowerIFName
Physical lower interface information. Appears only if the medium is ATM, Frame Relay (FR), or High-Level Data Link Control (HDLC).
Media
Medium over which the call is carried. If the call is carried over the (telephone) access side, the entry is TELE. If the call is carried over the voice network side, the entry is either ATM, FR, or HDLC.
Modem passthrough signaling method in use
Indicates that this is a modem pass-through call and that named signaling events (NSEs)—a Cisco-proprietary version of named telephone events in RFC 2833—are used for signaling codec upspeed. The upspeed method is the method used to dynamically change the codec type and speed to meet network conditions. This means that you might move to a faster codec when you have both voice and data calls and then slow down when there is only voice traffic.
NoiseLevel
Active noise level for this call.
OnTimeRvPlayout
Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.
OutSignalLevel
Active output signal level to the telephony interface used by this call.
PeerAddress
Destination pattern or number associated with this peer.
PeerId
ID value of the peer table entry to which this call was made.
PeerIfIndex
Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.
PeerSubAddress
Subaddress when this call is connected.
Percent Packet Loss
Total percent packet loss.
Port
Identification of the time-division multiplexing (TDM) voice port carrying the call.
ReceiveBytes
Number of bytes received by the peer during this call.
ReceiveDelay
Average Playout FIFO Delay plus the Decoder Delay during this voice call, in ms.
ReceivePackets
Number of packets received by this peer during this call.
ReleaseSource
Number value of the release source.
RemoteIPAddress
Remote system IP address for the VoIP call.
RemoteUDPPort
Remote system User Datagram Protocol (UDP) listener port to which voice packets are sent.
RoundTripDelay
Voice packet round-trip delay between the local and remote systems on the IP backbone for this call.
SelectedQoS
Selected Resourse Reservation Protocol (RSVP) quality of service (QoS) for this call.
Cisco IOS Voice Command Reference
VR-1825
show call active fax
Table 50
show call active fax Field Descriptions (continued)
Field
Description
SessionProtocol
Session protocol used for an Internet call between the local and remote routers through the IP backbone.
SessionTarget
Session target of the peer used for this call.
SetupTime
Value of the system UpTime, in milliseconds, when the call associated with this entry was started.
SignalingType
Signaling type for this call; for example, channel-associated signaling (CAS) or common channel signaling (CCS).
SIP call-legs
Total Session Initiation Protocol (SIP) call legs for which call records are available.
Telephony call-legs
Total telephony call legs for which call records are available.
Time between Buffer Drain/Fills Minimum and maximum durations between jitter buffer drain or fill events, in seconds. TransmitBytes
Number of bytes sent by this peer during this call.
TransmitPackets
Number of packets sent by this peer during this call.
TxDuration
The length of the call. Appears only if the medium is TELE.
VAD
Whether voice activation detection (VAD) was enabled for this call.
VoiceTxDuration
Duration of voice transmission from this peer to the voice gateway for this call, in ms. Derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value.
The following is sample output from the show call active fax brief command: Router# show call active fax brief : hs. + pid: \ tx:/ rx:/ IP : rtt:
