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Contents
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- aal2-profile custom
- aaa nas port voip
- aaa username
- access-list (voice source-group)
- access-policy
- accounting method
- accounting suppress
- accounting template
- acc-qos
- acct-template
- activation-key
- address-family (tgrep)
- address-hiding
- advertise (annex g)
- advertise (tgrep)
- alarm-trigger
- alias static
- allow-connections
- allow subscribe
- alt-dial
- anat
- ani mapping
- answer-address
- application (dial-peer)
- application (global)
- arq reject-resource-low
- arq reject-unknown-prefix
- as
- asp
- asserted-id
- associate application
- associate ccm
- associate profile
- associate registered-number
- asymmetric payload
- atm scramble-enable
- atm video aesa
- attribute acct-session-id overloaded
- attribute h323-remote-id resolved
- audio
- audio-prompt load
- authenticate redirecting-number
- authentication (dial peer)
- authentication (SIP UA)
- authentication method
- authorization method
- auto-config
- auto-cut-through
- accounting (gatekeeper)
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 customcommand 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
Command Default
One of the predefined International Telecommunication Union - Telecommunication Standardization Sector (ITU-T) profiles can be used.
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".
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.
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.
aaa username
To determine the information with which to populate the username attribute for Authentication, Authorization, and Accounting (AAA). billing records, use the aaa usernamecommand in SIP user agent configuration mode. To achieve default capabilities, use the no form of this command.
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 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.
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.
Usage Guidelines
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-numbercommand 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 ipaccess-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 1Related 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.
access-policy
To require that a neighbor be explicitly configured in order for requests to be accepted, use the access-policycommand in Annex G configuration mode. To reset the configuration to accept all requests, use the no form of this command.
Command Default
Border elements accept any and all requests if service relationships are not 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.
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.
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
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.
accounting suppress
To disable accounting that is automatically generated by a service provider module for a specific dial peer, use the accounting suppress command invoice class AAA configuration mode. To allow accounting to be automatically generated, use the no form of this command.
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 configuregw-accounting aaacommand in global configuration mode before configuring dial-peer-based accounting in voice class AAA configuration mode.
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.
Command Default
The dial peer does not choose and send a customized accounting template to the RADIUS server.
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 verbaaa accounting start [-tacctTempName] takes precedence over theaccounting template command in voice class AAA configuration mode.
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.
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.
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.
You must use the iprsvpbandwidth 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-delayas the acceptable QoS for inbound and outbound audio calls on VoIP dial peer 10:
dial-peer voice 10 voip acc-qos guaranteed-delayThe 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 videoacct-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 }
Command Modes
Gateway accounting AAA configuration (config-gw-accounting-aaa)
Gateway accounting file configuration (config-gw-accounting-file)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 theacct - 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.
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.
Usage Guidelines
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 30address-family (tgrep)
To set the global address family to be used on all dial peers, use the address-familycommand 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 }
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.
address-hiding
To hide signaling and media peer addresses from endpoints other than the gateway, use the address-hidding command in voice service voip configuration mode. To allow the peer address known to all endpoints, use the no form of this command.
Usage Guidelines
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.
NoteDistinctive 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.
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.
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.
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 }
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.
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.
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 exitRelated 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.
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.
Usage Guidelines
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.
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.
Syntax Description
Cisco IOS Release 12.3(4)T, Cisco IOS Release 12.3, and Earlier Releases
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.
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 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 are disabled by default, and POTS-to-any and any-to-POTS connections are enabled.
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.
12.4(4)T
This command was modified. 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.
Usage Guidelines
Cisco IOS Release 12.3(4)T, Cisco IOS Release 12.3, and Earlier Releases
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-connectionscommand 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 sipThe following example specifies that connections between H.323 endpoints are allowed:
Router(config-voi-serv)# allow-connections h323 to h323The following example specifies that connections between SIP endpoints are allowed:
Router(config-voi-serv)# allow-connections sip to sipallow 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.
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 subscribeRelated 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.
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.
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.
anat
To enable Alternative Network Address Types (ANAT) on a Session Initiation Protocol (SIP) trunk, use the anat command in voice service SIP configuration mode or dial peer configuration mode. To disable ANAT on SIP trunks, use the no form of this command.
Command Modes
Voice service voip-sip configuration (conf-serv-sip)
Dial peer configuration (config-dial-peer)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.
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.
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-STanswer-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.
Syntax Description
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 characterTto 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.
NoteCisco IOS software does not check the validity of the E.164 telephone number; it accepts any series of digits as a valid number.
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.
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 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.
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.
Usage Guidelines
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.
NoteIn Cisco IOS Release xx.x(x)X and later releases, the application name default refers to the 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.
NoteIn 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.
NoteMGCP scripting is not supported on the Cisco 1750 router or on Cisco 7200 series routers.
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:$d$@mail-server.cisco.comThe following example applies the MGCP application to a dial peer:
dial-peer voice 1 pots application MGCPAPPThe following example applies a predefined application to an incoming POTS dial peer:
dial-peer voice 100 pots application c4The 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:$d$@mail-server.cisco.comThe following example applies the predefined SIP application to a dial peer:
dial-peer voice 10 pots application sessionFor 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:DRelated 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.
application (global)
To enter application configuration mode to configure applications, use the application command in global 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)# applicationLoad the debit card script:
Router(config-app)# service debitcard tftp://server-1/tftpboot/scripts/app_debitcard.2.0.2.8.tclConfigure 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/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-unknown-prefix
To enable the gatekeeper to reject admission requests (ARQs) for zone prefixes that are not configured, use thearqreject-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.
Usage Guidelines
Use the arqreject-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 arqreject-unknown-prefix command. (The term hairpinis 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 local gk408 cisco.com zone remote gk415 cisco.com 172.21.139.91 zone prefix gk408 1408....... zone prefix gk415 1415.......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 zoneprefix 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 arqrequest-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 arqreject-unknown-prefix command is 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 rejected because the destination address does not match any configured prefix.
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 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 xx.x(x)X 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.)
NoteAll 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 showiuaasallcommand 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 2577The following output shows that the application server (as1) is defined for backhaul:
AS as1 10.21.0.2 9900asp
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 as-name { remote-p1 [ [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
(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-path-retransmissions
(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 History
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.
Usage Guidelines
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 showipsctpassociationparameters output. The default ToS value is 0.
NoteAll 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. Theip-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 2577Multiple 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.
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 7The following example specifies the IP address to enable and disable keepalives:
Router(config-iua)# asp asp1 as sctp-keepalive 10.1.2.34The 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 60000The 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 20The 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 10The 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 60000asserted-id
To enable support for the asserted ID header in incoming Session Initiation Protocol (SIP) requests or response messages, and to send the asserted ID privacy information in outgoing SIP requests or response messages, use the asserted-id command in voice service VoIP-SIP configuration mode. To disable the support for the asserted ID header, use the no form of this command.
Command Default
The privacy information is sent using the Remote-Party-ID (RPID) header or the FROM header.
Usage Guidelines
If you choose the pai keyword or the ppi keyword, the gateway builds the PAI header or the PPI header, respectively, into the common SIP stack. The pai keyword or the ppi keyword has the priority over the Remote-Party-ID (RPID) header, and removes the RPID header from the outbound message, even if the router is configured to use the RPID header at the global level.
Examples
The following example shows how to enable support for the PAI privacy header:
Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# asserted-id paiRelated Commands
Command
Description
calling-info pstn-to-sip
Specifies calling information treatment for PSTN-to-SIP calls.
privacy
Sets privacy in support of RFC 3323.
voice-class sip asserted-id
Enables support for the asserted ID header in incoming and outgoing SIP requests or response messages in dial-peer configuration mode.
associate application
To associate an application to the digital signal processor (DSP) farm profile, use the associateapplicationcommand in DSP farm profile configuration mode. To remove the protocol, use the no form of this command.
Syntax Description
cube
Associates the Cisco Unified Border Element application to a defined profile in the DSP farm.
sbc
Associates the SBC application to a defined profile in the DSP farm.
sccp
Associates the skinny client control protocol application to a defined profile in the DSP farm.
profile-description-text
(Optional) User defined name for the associated applicaion.
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.
Cisco IOS XE Release 3.2S
This command was modified. The cubeand sbckeywords and the profile-description-textargument were added.
Usage Guidelines
Use the associate application command to associate an application to a predefinded DSP farm profile.
Examples
The following example associates SCCP to the DSP farm profile:
Router(config-dspfarm-profile)# associate application sccpThe following example associates Cisco Unified Border Element to the DSP farm profile:
Router(config-dspfarm-profile)# associate application cubeRelated Commands
Command
Description
voice-card
Enters voice card configuration mode
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.
associate ccm
To associate a Cisco Unified CallManager with a Cisco CallManager group and establish its priority within the group, use the associateccmcommand 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
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)# sccpccmgroup999
Router(config-sccp-ccm)#associateccm125priority2
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.
associate profile
To associate a digital signal processor (DSP) farm profile with a Cisco CallManager group, use the associateprofilecommand 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
Usage Guidelines
The device name must match the name configured in Cisco UnifiedCallManager; otherwise the profile is not registered to Cisco Unified CallManager.
NoteEach profile can be associated to only one Cisco CallManager group.
Examples
The following example associates DSP farm profile abgz12345 to Cisco CallManager group 999:
Router(config)# sccp ccm group 999 Router(config-sccp-ccm)# associate profile 1 register abgz12345associate registered-number
To associate the preloaded route and outbound proxy details with the registered number, use the associateregistered-numbercommand in voice service VoIP SIP configuration mode. To remove the association, use the no form of this command.
Command Default
The preloaded route and outbound proxy details are not associated with the registered number by default.
asymmetric payload
To configure Session Initiation Protocol (SIP) asymmetric payload support, use the asymmetricpayloadcommand in SIP configuration mode. To disable asymmetric payload support, use the no form of this command.
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.
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.
atm scramble-enable
To enable scrambling on E1 links, use the atmscramble-enable command in interface configuration mode. To disable scrambling, use the noform of this command.
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.
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 atmvideoaesa command in ATM interface configuration mode. To remove any configured address for the interface, use the no form of this command.
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.
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.
attribute acct-session-id overloaded
To overload the acct-session-id attribute with call detail records, use the attributeacct-session-idoverloaded 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.
Usage Guidelines
The attributeacct-session-idoverloadedcommand replaces the gw-accountingh323command.
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.
attribute h323-remote-id resolved
To resolve the h323-remote-id attribute, use the attributeh323-remote-idresolvedcommand in gateway accounting AAA configuration mode. To keep the h323-remote-id attribute unresolved, use the no form of this command.
Usage Guidelines
In Cisco IOS Release 12.2(11)T, the attributeh323-remote-idresolved command replaces the gw-accountingh323resolvecommand, 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.
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 theno form of this command.
audio { incoming | outgoing } level adjustment value
no audio { incoming | outgoing } level adjustment value
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.
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-promptloadcommand in privileged EXEC mode. This command does not have a no form.
Command History
Release
Modification
11.3(6)NA2
This command was introduced.
Note With Cisco IOS Release 11.3(6)NA2, the URL pointer refers to the directory where Flash memory is stored.
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.
Usage Guidelines
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.
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.
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 authenticateredirecting-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.
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.
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 authenticateredirecting-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-classsipauthenticateredirecting-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-classsipauthenticateredirecting-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.
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-numberauthentication (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] ] ] | all }
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.
all
(Optional) Specifies all the authentication entries for the user (dial-peer).
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.
NoteThe 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.
- 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.
- Use the no authentication all command to remove all the authentication entries for the user.
Examples
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.comThe 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 MyPasswordRelated 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.
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 ] ] | all }
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.
all
(Optional) Specifies all the authentication entries for the user (sip-ua).
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 usernameargument.
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.
- 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.
- 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.
- Use the no authentication all command to remove all the authentication entries for the user.
Examples
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.comThe 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.comRelated 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.
authentication method
To set an authentication method at login for calls that come into a dial peer, use the authenticationmethod command in voice class AAA configuration mode. To disable the authentication method set at login, use the no form of this command.
Command Default
When this command is not used to specify a login authentication method, the system uses the aaaauthenticationloginh323 command as the default.
Usage Guidelines
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.
authorization method
To set an authorization method at login for calls that are into a dial peer, use the authorizationmethod command in voice class AAA configuration mode. To disable the authorization method set at login, use the no form of this command.
Command Default
When this command is not used to specifiy a login authorization method, the system uses the aaaauthorizationexech323 command as the default.
Usage Guidelines
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.
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-configcommand in global configuration mode. To disable auto-configuration, use the no form of this command.
Examples
The following example shows theauto-config command used to enter auto-configuration application configuration mode for the SCCP application and thenoshutdown command used to enable the SCCP application for download:
Router(config)# auto-config application sccp Router(auto-config-app)# no shutdownauto-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.
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 noform of this command.
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 h323idkeyword was added.
12.2SX
This command is supported in the Cisco IOS Release 12.2SX train. Support in a specific 12.2SX release of this train depends on your feature set, platform, and platform hardware.
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 accountingThe following example shows how to enable VSA accounting:
aaa accounting connection start-stop group radius gatekeeper accounting exec vsaThe following example configures H.323 accounting using IETF RADIUS attributes:
Router(config-gk)# accounting username h323idThe following example configures H.323 accounting using VSA RADIUS attributes:
Router(config-gk)# accounting vsa
