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.

Syntax Description

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

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".

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.

Syntax Description

This command has no arguments or keywords.

Command Default

Disabled

Command Modes

Global configuration (config)

Command History

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.

Related Commands

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

Command Default

calling-number

Command Modes

SIP user agent configuration (config-sip-ua)

Command History

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.

Related Commands

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.

Syntax Description

Command Default

No default behavior or values

Command Modes

Voice source-group configuration (cfg-source-grp)

Command History

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.

Related Commands

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.

Syntax Description

Command Default

Border elements accept any and all requests if service relationships are not configured.

Command Modes

Annex G configuration (config-annexg)

Command History

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.

Related Commands

To specify that the secure (encrypted) mode is to be used for accessing the session border controller (SBC), use the access-secure command in phone proxy configuration mode. To remove the secure mode, use the no form of the command.

This command has no arguments or keywords.

Command Default

The non-secure mode is used for communication with the SBC.

Command Modes

Phone proxy configuration mode (config-phone-proxy)

Usage Guidelines

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.

Syntax Description

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

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.

Related Commands

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.

Syntax Description

Command Default

Accounting is automatically generated by the service provider module.

Command Modes

Voice class AAA configuration

Command History

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.

Related Commands

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.

Syntax Description

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

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.

Related Commands

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

Command Default

RSVP makes no bandwidth reservations.

Command Modes

Dial-peer configuration (config-dial-peer)

Command History

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.

Related Commands

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.

Syntax Description

Command Default

No voice attributes are collected.

Command Modes

Gateway accounting AAA configuration (config-gw-accounting-aaa)
Gateway accounting file configuration (config-gw-accounting-file)

Command History

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.

Related Commands

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.

Syntax Description

Command Default

Callback activation key is #1.

Command Modes

STC application feature callback configuration (config-stcapp-callback)

Command History

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.

Related Commands

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.

Syntax Description

Command Default

E.164 address family

Command Modes

TGREP configuration (config-tgrep)

Command History

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.

Related Commands

To hide signaling and media peer addresses from endpoints other than the gateway, use the address-hiding command in voice service voip configuration mode. To allow the peer address known to all endpoints, use the no form of this command.

Syntax Description

There are no keywords or arguments.

Command Default

Signaling and media addresses are visible to all endpoints.

Command Modes

Voice service voip configuration (config-voi-serv)

Command History

Usage Guidelines

All SIP methods or messages must 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.

This command modifies specific supplementary service headers and changes them appropriately from the in-leg to the out-leg.

Those headers include the following:

• Refer-To

• Referred-by

• 3xx response contact header

• History-Info

• Diversion

In these headers, an inside IP would be silently passed from the in-leg to the out-leg by an IP-to-IP gateway resulting in inside IP being sent to the ITSP/Public Internet. When configured with address-hiding the IP-to-IP gateway specifically looks for and changes those headers appropriately to mask the inside IP with its own.

Note	Distinctive ringing headers include ringing information and server address where the ringtone can be obtained. These headers are forwarded as is to the peer side even if address hiding is enabled.

Related Commands

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.

Syntax Description

Command Default

Static

Command Modes

Annex G configuration (config-annexg)

Command History

Related Commands

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.

Syntax Description

Command Default

No attributes for address families are advertised.

Command Modes

TGREP configuration (config-tgrep)

Command History

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.

Related Commands

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.

Syntax Description

Command Default

No alarm is sent.

Command Modes

Controller configuration (config-controller)

Command History

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.

Related Commands

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.

Syntax Description

Command Default

No static aliases exist.

Command Modes

Gatekeeper configuration (config-gk)

Command History

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.

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 Modes

Voice-service configuration (config-voi-serv)

Command History

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.

Related Commands

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.

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

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.

Related Commands

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.

Syntax Description

Command Default

No alternate dial-out string is configured

Command Modes

Dial-peer configuration (config-dial-peer)

Command History

Usage Guidelines

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.

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 or voice class tenant, or dial peer configuration mode. To disable ANAT on SIP trunks, use the no form of this command.

Syntax Description

This command has no arguments or keywords.

Command Default

ANAT is enabled on SIP trunks.

Command Modes

Voice service voip-sip configuration (conf-serv-sip)

Dial peer configuration (config-dial-peer)

Voice class tenant configuration (config-class)

Command History

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.

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.

Syntax Description

Command Default

No default behavior or values

Command Modes

Voice-port configuration (config-voiceport)

Command History

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.

Related Commands

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

Command Default

The default value is enabled with a null string

Command Modes

Dial peer configuration Router (config-dial-peer)

Command History

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.

Note	Cisco IOS software does not check the validity of the E.164 telephone number; it accepts any series of digits as a valid number.

Related Commands

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

Command Default

No default behavior or values

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

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.

Note

In 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.

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.

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.

Related Commands

To enter application configuration mode to configure applications, use the application command in global configuration mode.

Syntax Description

This command has no keywords or arguments.

Command Default

No default behavior or values

Command Modes

Global configuration (config)

Command History

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).

Related Commands

To export the audio and video call quality statistics to flow record using Flexible NetFlow collector, use the aqm-register-fnf command in global configuration mode. To disable the export of audio and video call quality statistics, use the no form of this command.

Syntax Description

This command has no arguments or keywords.

Command Default

The aqm-register-fnf command is enabled.

Command Modes

Global configuration (config)

Usage Guidelines

Use the aqm-register-fnf command when you want to export metrics related to media (voice) quality; for example, conversational mean opinion score (MOS), packet loss rate, conceal ratio, and so on. The aqm-register-fnf command must be configured before you use the media monitoring command to configure voice quality metrics.

Note	Configuring the no aqm-register-fnf command does not disable the command in the device's running and startup configurations.

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.

Syntax Description

This command has no arguments or keywords.

Command Default

No default behavior or values

Command Modes

Gatekeeper configuration (config-gk)

Command History

Related Commands

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.

Syntax Description

This command has no arguments or keywords

Command Default

The gatekeeper accepts and processes all incoming ARQs.

Command Modes

Gatekeeper configuration (config-gk)

Command History

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.

Related Commands

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.

Syntax Description

Command Default

No application server is defined.

Command Modes

IUA configuration (config-iua)

Command History

Usage Guidelines

A maximum of two local IP addresses can be specified. (Note that SCTP has built-in support for multihomed machines.)

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 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.

Related Commands

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.

Syntax Description

Command Default

No ASP is defined.

Command Modes

IUA configuration (config-iua)

Command History

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.

Note	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. 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.

Related Commands

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 or voice class tenant configuration mode. To disable the support for the asserted ID header, use the no form of this command.

Syntax Description

Command Default

The privacy information is sent using the Remote-Party-ID (RPID) header or the FROM header.

Command Modes

Voice service VoIP-SIP configuration (conf-serv-sip)

Voice class tenant configuration (config-class)

Command History

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.

Related Commands

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

Command Default

No application is associated with the DSP farm profile.

Command Modes

DSP farm profile configuration (config-dspfarm-profile)

Command History

Usage Guidelines

Use the associate application command to associate an application to a predefinded DSP farm profile.

Related Commands

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.

Syntax Description

Command Default

No default behavior or values

Command Modes

SCCP Cisco CallManager configuration (config-sccp-ccm)

Command History

Related Commands

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.

Syntax Description

Command Default

This command is not enabled.

Command Modes

SCCP Cisco CallManager configuration (conig-sccp-ccm)

Command History

Usage Guidelines

The device name must match the name configured in Cisco UnifiedCallManager; otherwise the profile is not registered to Cisco Unified CallManager.

Note	Each profile can be associated to only one Cisco CallManager group.

Related Commands

To associate the preloaded route and outbound proxy details with the registered number, use the associateregistered-number command in voice service VoIP SIP configuration mode or voice class tenant configuration mode. To remove the association, use the no form of this command.

Syntax Description

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)

Voice class tenant configuration (config-class)

Command History

Related Commands

To configure Session Initiation Protocol (SIP) asymmetric payload support, use the asymmetricpayload command in SIP configuration mode or voice class tenant configuration mode. To disable asymmetric payload support, use the no form of this command.

Syntax Description

Command Default

This command is disabled.

Command Modes

Voice service SIP configuration (conf-serv-sip)

Voice class tenant configuration (config-class)

Command History

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.

Related Commands

To enable scrambling on E1 links, use the atmscramble-enable command in interface configuration mode. To disable scrambling, use the noform of this command.

Syntax Description

This command has no arguments or keywords.

Command Default

By default, payload scrambling is set off

Command Modes

Interface configuration

Command History

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.

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

Command Default

default

Command Modes

ATM Interface configuration

Command History

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.

Related Commands

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.

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 (config-gw-accounting-aaa)

Command History

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.

Related Commands

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.

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 (config-gw-accounting-aaa)

Command History

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.

Related Commands

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.

Syntax Description

Command Default

This command is disabled by default, and there is no volume control available.

Command Modes

Dial peer configuration (config-dial-peer)

Command History

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.

Related Commands

To allow only audio and image (for T.38 Fax) media types, and drop all other media types (such as video and application), use the audio forced command in voice service voip sip configuration mode. To disable, use no form of this command.

Command Default

Along with audio and image (for T38 fax) media types, all other media types (such as video and application) are also allowed.

Command Modes

voice service voip sip configuration (conf-serv-sip)

Usage Guidelines

Use audio forced command to globally allow only audio and image media types.

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.

Syntax Description

Command Default

No default behavior or values

Command Modes

Privileged EXEC

Command History

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.

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 or voice class tenant 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.

Syntax Description

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)

Voice class tenant configuration (config-class)

Command History

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.

Related Commands

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.

Syntax Description

Command Default

SIP digest authentication is disabled.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

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.

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.

• 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.

Related Commands

To enable SIP digest authentication, use the authentication command in SIP UA or voice class tenant configuration mode. To disable SIP digest authentication, use the no form of this command.

Syntax Description

Command Default

SIP digest authentication is disabled.

Command Modes

SIP UA configuration (config-sip-ua)

Voice class tenant configuration (config-class)

Command History

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.

Related Commands

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.

Syntax Description

Command Default

When this command is not used to specify a login authentication method, the system uses the aaaauthenticationloginh323 command as the default.

Command Modes

Voice class AAA configuration

Command History

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.

Related Commands

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.

Syntax Description

Command Default

When this command is not used to specifiy a login authorization method, the system uses the aaaauthorizationexech323 command as the default.

Command Modes

Voice class AAA configuration

Command History

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.

Related Commands

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.

Syntax Description

Command Default

Auto-configuration is disabled.

Command Modes

Global configuration (config)

Command History

Related Commands

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.

Syntax Description

This command has no arguments or keywords.

Command Default

Auto-cut-through is enabled.

Command Modes

Voice-port configuration (config-voiceport)

Command History

Usage Guidelines

The auto-cut-through command applies to ear and mouth (E&M) voice ports only.

Related Commands

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.

Syntax Description

Command Default

Accounting is disabled.

Command Modes

Gatekeeper configuration

Command History

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.

Related Commands