Cisco IOS Voice Command Reference

voice-class sip error-code-override

To configure the Session Initiation Protocol (SIP) error code that a dial peer uses for options-keepalive failures or call spike failures, use the voice-class sip error-code-override command in dial peer voice configuration mode. To disable the SIP error code configuration, use the no form of this command.

voice-class sip error-code-override {options-keepalive failure | call spike failure} {sip-status-code-number | system}

no voice-class sip error-code-override {options-keepalive failure | call spike failure}

Syntax Description

Defaults

By default the SIP error code is not configured.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

The voice-class sip error-code-override command in dial peer voice configuration mode configures the error code response for options-keepalive failures and call spike failures at dial peer level. The error-code-override command in voice service SIP configuration mode configures the error code responses options-keepalive failures and call spike failures gloablly.

Table 249 describes the SIP error codes.

Examples

The following example shows how to configure the SIP error code for options-keepalive failures using the voice-class sip error-code-override command:

Router(config)# dial-peer voice 432 voip system

Router(config-dial-peer)# voice-class sip error-code-override options-keepalive failure 
502

The following example shows how to configure the SIP error code for call spike failures using the voice-class sip error-code-override command:

Router(config)# dial-peer voice 432 voip system

Router(config-dial-peer)# voice-class sip error-code-override call spike failure 502

Related Commands

voice-class sip g729 annexb-all

To configure settings on a Cisco IOS Session Initiation Protocol (SIP) gateway that determine if a specific dial peer on the gateway treats the G.729br8 codec as superset of G.729r8 and G.729br8 codecs for interoperation with Cisco Unified Communications Manager, use the voice-class sip g729 annexb-all command in dial peer voice configuration mode. To prevent a dial peer from treating the G.729br8 codec as a superset of the G.729r8 and G.729br8 codecs, use the no form of this command.

voice-class sip g729 annexb-all [system]

no voice-class sip g729 annexb-all

Syntax Description

Command Default

The dial peer defers to global (system) settings for the Cisco IOS gateway.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

There are four variations of the G.729 coder-decoder (codec), which fall into two categories:

High Complexity

•G.729 (g729r8)—a high complexity algorithm codec on which all other G.729 codec variations are based.

•G.729 Annex-B (g729br8 or G.729B)—a variation of the G.729 codec that allows the DSP to detect and measure voice activity and convey suppressed noise levels for re-creation at the other end. Additionally, the Annex-B codec includes Internet Engineering Task Force (IETF) voice activity detection (VAD) and comfort noise generation (CNG) functionality.

Medium Complexity

•G.729 Annex-A (g729ar8 or G.729A)—a variation of the G.729 codec that sacrifices some voice quality to lessen the load on the DSP. All platforms that support G.729 also support G.729A.

•G.729A Annex-B (g729abr8 or G.729AB)—a variation of the G.729 Annex-B codec that, like G.729B, sacrifices voice quality to lessen the load on the DSP. Additionally, the G.729AB codec also includes IETF VAD and CNG functionality.

The VAD and CNG functionality is what causes the instability during communication attempts between two DSPs where one DSP is configured with Annex-B (G.729B or G.729AB) and the other without (G.729 or G.729A). All other combinations interoperate. To configure a dial peer on a Cisco IOS SIP gateway for interoperation with Cisco Unified Communications Manager (formerly known as the Cisco CallManager, or CCM), use the voice-class sip g729 annexb-all command in dial peer voice configuration mode to do one of the following:

•Override global settings for a Cisco IOS gateway and configure the dial peer to accept and connect calls between two DSPs with incompatible G.729 codecs.

•Specify that an individual dial peer use the global (system) settings on the Cisco IOS SIP gateway.

•Use the no form of the command to override global settings for the Cisco IOS gateway and specify that the dial peer does not treat the G.729br8 codec as a superset of G.729r8 and G.729br8 codecs.

Use the g729 annexb-all command in voice service SIP configuration mode to configure the global settings for the Cisco IOS SIP gateway.

Examples

The following example shows how to configure a dial peer on a Cisco IOS SIP gateway to connect calls between two DSPs using incompatible G.729 codecs, overriding global gateway settings for this feature:

Router> enable

Router# configure terminal

Router(config)# dial-peer voice 1

Router(config-dial-peer)# voice-class sip g729 annexb-all

Related Commands

voice-class sip history-info

To enable Session Initiation Protocol (SIP) history-info header support on the Cisco IOS gateway at the dial-peer level, use the voice-class sip history-info command in dial peer configuration mode. To disable SIP history-info header support, use the no form of this command.

voice-class sip history-info [system]

no voice-class sip history-info

Syntax Description

Command Default

History-info header support is disabled.

Command Modes

Dial peer configuration (conf-dial-peer)

Command History

Usage Guidelines

Use this command to enable history-info header support at the dial-peer level. The history-info header (as defined in RFC 4244) records the call or dialog history. The receiving application uses the history-info header information to determine how and why the call has reached it.

Note The Cisco IOS SIP gateway cannot use the information in the history-info header to make routing decisions.

Examples

The following example enables SIP history-info header support at the dial-peer level:

Router(config)# dial-peer voice 2 voip

Router(config-dial-peer)# voice-class sip history-info

The following example enables SIP history-info header support at the dial-peer level using the global configuration settings:

Router(config)# dial-peer voice 2 voip

Router(config-dial-peer)# voice-class sip history-info system

Related Commands

voice-class sip localhost

To configure individual dial peers to override global settings on Cisco IOS voice gateways, Cisco Unified Border Elements (Cisco UBEs), or Cisco Unified Communications Manager Express (Cisco Unified CME) and substitute a Domain Name System (DNS) hostname or domain as the localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers in outgoing messages, use the voice-class sip localhost command in dial peer voice configuration mode. To disable substitution of a localhost name on a specific dial peer, use the no form of this command. To configure a specific dial peer to defer to global settings for localhost name substitution, use the default form of this command.

voice-class sip localhost dns:[hostname.]domain [preferred]

no voice-class sip localhost

default voice-class sip localhost

Syntax Description

Command Default

The dial peer uses the global configuration setting to determine whether a DNS localhost name is substituted in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

Use the voice-class sip localhost command in dial peer voice configuration mode to override the global configuration on Cisco IOS voice gateways, Cisco UBEs, or Cisco Unified CME and configure a DNS localhost name to be used in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages on a specific dial peer. When multiple registrars are configured for an individual dial peer you can then use the voice-class sip localhost preferred command to specify which host is preferred for that dial peer.

To globally configure a localhost name on a Cisco IOS voice gateway, Cisco UBE, or Cisco Unified CME, use the localhost command in voice service SIP configuration mode. Use the no voice-class sip localhost command to remove localhost name configurations for the dial peer and to force the dial peer to use the physical IP address in the host portion of the From, Call-ID, and Remote-Party-ID headers regardless of the global configuration.

Examples

The following example shows how to configure dial peer 1 (overriding any global configuration) to substitute a domain (no hostname specified) as the preferred localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages:

Router> enable

Router# configure terminal

Router(config)# dial-peer voice 1 voip

Router(config-dial-peer)# voice-class sip localhost dns:example.com preferred

The following example shows how to configure dial peer 1 (overriding any global configuration) to substitute a specific hostname on a domain as the preferred localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages:

Router> enable

Router# configure terminal

Router(config)# dial-peer voice 1 voip

Router(config-dial-peer)# voice-class sip localhost dns:MyHost.example.com preferred

The following example shows how to force dial peer 1 (overriding any global configuration) to use the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages:

Router> enable

Router# configure terminal

Router(config)# dial-peer voice 1 voip

Router(config-dial-peer)# no voice-class sip localhost

Related Commands

voice-class sip map resp-code

To configure an individual dial peer on a Cisco Unified Border Element (Cisco UBE) to map specific received Session Initiation Protocol (SIP) provisional response messages to a different SIP provisional response message on the outgoing SIP dial peer, use the voice-class sip map resp-code command in dial peer voice configuration mode. To disable mapping of received SIP provisional response messages on an individual dial peer, use the no form of this command. To configure a specific dial peer to defer to global settings for mapping of incoming SIP provisional response messages, use the default form of this command.

voice-class sip map resp-code 181 to 183

no voice-class sip map resp-code 181 to 183

default voice-class sip map resp-code 181 to 183

Syntax Description

Command Default

Mapping behavior is determined by the global configuration setting, which, if not specifically configured, means that incoming SIP provisional responses are passed, as is to the outbound SIP dial peer.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

Use the voice-class sip map resp-code command in dial peer voice configuration mode to configure an individual dial peer on a Cisco UBE to map incoming SIP 181 provisional response messages to SIP 183 provisional response messages on the outgoing SIP dial peer.

Note If the block command is configured for incoming SIP 181 messages, either globally or at the dial-peer level, the messages may be dropped before they can be passed or mapped to a different message—even when the voice-class sip map resp-code command is enabled. To globally configure whether and when incoming SIP 181 messages are dropped, use the block command in voice service SIP configuration mode (or use the voice-class sip block command in dial peer voice configuration mode to configure drop settings on individual dial peers).

To configure mapping of SIP provisional response messages globally on a Cisco UBE, use the map resp-code command in voice service SIP configuration mode. To disable mapping of SIP 181 message for an individual dial peer on a Cisco UBE, use the no voice-class sip map resp-code command in voice service SIP configuration mode.

As an example, to enable interworking of SIP endpoints that do not support the handling of SIP 181 provisional response messages, you could use the block command to configure a Cisco UBE to drop SIP 181 provisional response messages received on the SIP trunk or you can use the map resp-code command to configure the Cisco UBE to map the incoming messages to and send out, instead, SIP 183 provisional response messages to the SIP line in Cisco Unified Communications Manager Express (Cisco Unified CME).

Note This command is supported only for SIP-to-SIP calls and will have no effect on H.323-to-SIP or time-division multiplexing (TDM)-to-SIP calls.

Examples

The following example shows how to configure dial peer 1 to map incoming SIP 181 provisional response messages to SIP 183 provisional response messages on the outbound dial peer:

Router> enable

Router# configure terminal

Router(config)# dial-peer voice 1 voip

Router(config-dial-peer)# voice-class sip map resp-code 181 to 183

Related Commands

voice-class sip options-keepalive

To monitor connectivity between Cisco Unified Border Element VoIP dial-peers and SIP servers to, use the voice-class sip options-keepalive command in dial peer configuration mode. To disable monitoring connectivity, use the no form of this command.

voice-class sip options-keepalive [up-interval seconds | down-interval seconds] [retry retries]

no voice-class sip options-keepalive

Syntax Description

Command Default

The dial-peer is active (UP).

Command Modes

Dial peer configuration mode (config-dial-peer).

Command History

Usage Guidelines

Use the voice-class sip options-keepalive command to configure a out-of-dialog (OOD) Options Ping mechanism between any number of destinations. When monitored endpoint heartbeat responses fails, the configured dial-peer is busied out. If there is a alternate dial-peer configured for the same destination pattern, the call is failed over to the next preference dial peer or the on call is rejected with an error cause code.

The response to options ping will be considered unsuccessful and dial-peer will be busied out for following scenarios:

All other error codes, including 400 are considered a valid response and the dial peer is not busied out.

Examples

The following example shows a sample configuration of dial peer 100 configured to reset:

dial-peer voice 100 voip

  voice-class sip options-keepalive up-interval 12 down-interval 65 retry 3

Related Commands

voice-class sip outbound-proxy

To configure an outbound proxy, use the voice-class sip outbound-proxy command in dial peer configuration mode. To reset the outbound proxy value to its default, use the no form of this command.

voice-class sip outbound-proxy {dhcp | ipv4:ipv4-address | ipv6:[ipv6-address] | dns:host:domain} [:port-number]

no voice-class sip outbound-proxy

Syntax Description

Command Default

An outbound proxy is not configured.

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Usage Guidelines

The voice-class sip outbound-proxy command, in dial peer configuration mode, takes precedence over the command in SIP global configuration mode.

Brackets must be entered around the IPv6 address.

Examples

The following example shows how to configure the voice-class sip outbound-proxy command on a dial peer to generate an IPv4 address (10.1.1.1) as an outbound proxy:

Router> enable

Router# configure terminal

Router(config)# dial-peer voice 111 voip

Router(config-dial-peer)# voice-class sip outbound-proxy ipv4:10.1.1.1

The following example shows how to configure the voice-class sip outbound-proxy command on a dial peer to generate a domain (sipproxy:cisco.com) as an outbound proxy:

Router> enable

Router# configure terminal

Router(config)# dial-peer voice 111 voip

Router(config-dial-peer)# voice-class sip outbound-proxy dns:sipproxy:cisco.com

The following example shows how to configure the voice-class sip outbound-proxy command on a dial peer to generate an outbound proxy using DHCP:

Router> enable

Router# configure terminal

Router(config)# dial-peer voice 111 voip

Router(config-dial-peer)# voice-class sip outbound-proxy dhcp

Related Commands

voice-class sip preloaded-route

To enable preloaded route support for dial-peer Session Initiation Protocol (SIP) calls, use the voice-class sip preloaded-route command in dial peer voice configuration mode. To reset to the default value, use the no form of this command.

voice-class sip preloaded-route {[sip-server] service-route | system}

no voice-class sip preloaded-route

Syntax Description

Command Default

SIP calls at the dial-peer level use the global configuration level settings.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

The voice-class sip preloaded-route command takes precedence over the preloaded-route command configured in SIP configuration mode. However, if the voice-class sip preloaded-route command is used with the system keyword, the gateway uses the global settings configured by the preloaded-route command.

Examples

The following example shows how to configure the dial peer to include SIP server and Service-Route information in the Route header:

dial-peer voice 102 voip

 voice-class sip preloaded-route sip-server service-route

The following example shows how to configure the dial peer to include only Service-Route information in the Route header:

dial-peer voice 102 voip

 voice-class sip preloaded-route service-route

Related Commands

voice-class sip privacy

To set privacy support at the dial-peer level as defined in RFC 3323, use the voice-class sip privacy command in dial peer configuration mode. To remove privacy support as defined in RFC 3323, use the no form of this command.

voice-class sip privacy {disable | pstn | system | privacy-option [critical]}

no voice-class sip privacy

Syntax Description

Command Default

Privacy support is disabled.

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Usage Guidelines

Use the voice-class sip privacy command to instruct the gateway to add a Proxy-Require header, set to a value supported by RFC 3323, in outgoing SIP request messages at the dial-peer level.

Use the voice-class sip privacy critical command to instruct the gateway to add a Proxy-Require header with the value set to critical. If a user agent sends a request to an intermediary that does not support privacy extensions, the request fails.

The voice-class sip privacy command takes precedence over the privacy command in voice service voip sip configuration mode. However, if the voice-class sip privacy command is used with the system keyword, the gateway uses the settings configured globally by the privacy command.

Examples

The following example shows how to disable the privacy on dial peer 2:

Router> enable 

Router# configure terminal

Router(config)# dial-peer voice 2 voip 

Router(config-dial-peer)# voice-class sip privacy disable

The following example shows how to configure the voice-class sip privacy command so that the information held in the history-info header is hidden outside the trust domain:

Router> enable 

Router# configure terminal

Router(config)# dial-peer voice 2 voip 

Router(config-dial-peer)# voice-class sip privacy history

Related Commands

voice-class sip privacy-policy

To configure the privacy header policy options at the dial-peer level, use the voice-class sip privacy-policy command in dial peer voice configuration mode. To disable privacy-policy options, use the no form of this command.

voice-class sip privacy-policy {passthru | send-always | strip {diversion | history-info}} [system]

no voice-class sip privacy-policy {passthru | send-always | strip {diversion | history-info}}

Syntax Description

Command Default

No privacy-policy settings are configured.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

If a received message contains privacy values, use the voice-class sip privacy-policy passthru command to ensure that the privacy values are passed from one call leg to the next. If a received message does not contain privacy values but the privacy header is required, use the voice-class sip privacy-policy send-always command to set the privacy header to None and forward the message to the next call leg. You can configure the system to support both options at the same time.

The voice-class sip privacy-policy command takes precedence over the privacy-policy command in voice service voip sip configuration mode. However, if the voice-class sip privacy-policy command is used with the system keyword, the gateway uses the settings configured globally by the privacy-policy command.

Examples

The following example shows how to enable the pass-through privacy policy on the dial peer:

Router> enable 

Router# configure terminal

Router(config)# dial-peer voice 2611 voip

Router(config-dial-peer)# voice-class sip privacy-policy passthru

The following example shows how to enable the pass-through, send-always, and strip policies on the dial peer:

Router> enable 

Router# configure terminal

Router(config)# dial-peer voice 2611 voip

Router(config-dial-peer)# voice-class sip privacy-policy passthru

Router(config-dial-peer)# voice-class sip privacy-policy send-always

Router(config-dial-peer)# voice-class sip privacy-policy strip diversion

Router(config-dial-peer)# voice-class sip privacy-policy strip history-info

The following example shows how to enable the send-always privacy policy on the dial peer:

Router> enable 

Router# configure terminal

Router(config)# dial-peer voice 2611 voip

Router(config-dial-peer)# voice-class sip privacy-policy send-always

The following example shows how to enable both the pass-through privacy policy and send-always privacy policies on the dial peer:

Router> enable 

Router# configure terminal

Router(config)# dial-peer voice 2611 voip

Router(config-dial-peer)# voice-class sip privacy-policy passthru

Router(config-dial-peer)# voice-class sip privacy-policy send-always

Related Commands

voice-class sip random-contact

To populate the outgoing INVITE message with random-contact information (instead of clear contact information) at the dial-peer level, use the voice-class sip random-contact command in dial peer voice configuration mode. To disable random contact information, use the no form of this command.

voice-class sip random-contact [system]

no voice-class sip random-contact

Syntax Description

Command Default

Support for random contact at the dial-peer level uses the the global configuration level settings.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

To populate outbound INVITE messages (from the Cisco Unified Border Element) with random-contact information instead of clear-contact information at the dial-peer level, use the voice-class sip random-contact command. This functionality will work only when the Cisco Unified Border Element is configured for SIP registration with random-contact, using the credentials and registrar commands.

The voice-class sip random-contact command takes precedence over the random-contact command in voice service voip sip configuration mode. However, if the voice-class sip random-contact command is used with the system keyword, the gateway uses the settings configured globally by the random-contact command.

Examples

The following example shows how to populate outbound INVITE messages, at the dial-peer level, with random-contact information:

Router> enable 

Router# configure terminal

Router(config)# dial-peer voice 2611 voip

Router(config-dial-peer)# voice-class sip random-contact

Related Commands

voice-class sip random-request-uri validate

To enable the validation of the called-number based on the random value generated during the registration of the number, at dial-peer configuration level, use the voice-class sip random-request-uri validate command in dial peer voice configuration mode. To disable validation, use the no form of this command.

voice-class sip random-request-uri validate [system]

no voice-class sip random-request-uri validate

Syntax Description

Command Default

Validation is disabled.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

The system generates a random string when registering a new number. An INVITE message with the P-Called-Party-ID value can have the Request-URI set to this random number. To enable the system to identify the called number from the random number in the Request-URI, use the voice-class sip random-request-uri validate command on the inbound dial peer.

If the P-Called-Party-ID is not set in the INVITE message, the Request URI for that message must contain the called party information (and cannot contain a random number). Therefore validation is performed only on INVITE messages with a P-Called-Party-ID.

The voice-class sip random-request-uri validate command takes precedence over the random-request-uri validate command in voice service voip sip configuration mode. However, if the voice-class sip random-request-uri validate command is used with the system keyword, the gateway uses the settings configured globally by the random-request-uri validate command.

Examples

The following example shows how to enable call routing based on the P-Called-Party-ID header value at the dial-peer configuration level:

Router> enable 

Router# configure terminal

Router(config)# dial-peer voice 2611 voip 

Router(config-dial-peer)# voice-class sip random-request-uri validate

Related Commands

voice-class sip registration passthrough

To configure Session Initiation Protocol (SIP) registration pass-through options on a dial peer, use the voice-class sip registration passthrough command in dial peer voice configuration mode. To disable the configuration, use the no form of this command.

voice-class sip registration passthrough [[static] [rate-limit [expires value] [fail-count value]] [registrar-index [index]] | system]

no voice-class sip registration passthrough

Syntax Description

Command Default

SIP registration pass-through options that are configured at the global level are configured.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

You can use the voice-class sip registration passthrough command to configure the following SIP pass-through functionalities on a dial peer:

•Back-to-back registration facility to register phones for call routing.

•Options to configure the rate-limiting values, such as the expiry time, fail-count, and a list of registrars to be used for registration.

Examples

The following example shows how to set the registrar index of 1 for the SIP registration pass-through rate limiting:

Router# configure terminal

Router(config)# dial-peer voice 444 voip

Router(config-dial-peer)# voice-class sip registration passthrough static rate-limit 
registrar-index 1

Related Commands

voice-class sip rel1xx

To enable all Session Initiation Protocol (SIP) provisional responses (other than 100 Trying) to be sent reliably to the remote SIP endpoint, use the voice-class sip rel1xx command in dial peer configuration mode. To reset to the default, use the no form of this command.

voice-class sip rel1xx {supported value | require value | system | disable}

no sip rel1xx

Syntax Description

Command Default

system

Command Modes

Dial peer configuration

Command History

Usage Guidelines

There are two ways to configure reliable provisional responses:

•Dial-peer mode. You can configure reliable provisional responses for the specific dial peer only by using the voice-class sip rel1xx command.

•SIP mode. You can configure reliable provisional responses globally by using the rel1xx command.

The use of resource reservation with SIP requires that the reliable provisional feature for SIP be enabled either at the VoIP dial-peer level or globally on the router.

This command applies to the dial peer under which it is used or points to the global configuration for reliable provisional responses. If the command is used with the supported keyword, the SIP gateway uses the Supported header in outgoing SIP INVITE requests. If it is used with the require keyword, the gateway uses the Required header.

This command, in dial peer configuration mode, takes precedence over the rel1xx command in global configuration mode with one exception: If this command is used with the system keyword, the gateway uses what was configured under the rel1xx command in global configuration mode.

Examples

The following example shows how to use this command on either an originating or a terminating SIP gateway:

•On an originating gateway, all outgoing SIP INVITE requests matching this dial peer contain the Supported header where value is 100rel.

•On a terminating gateway, all received SIP  INVITE requests matching this dial peer support reliable provisional responses.

Router(config)# dial-peer voice 102 voip

Router(config-dial-peer)# voice-class sip rel1xx supported 100rel

Related Commands

voice-class sip reset timer expires

To configure an individual dial peer on Cisco Unified Communications Manager Express (Cisco Unified CME), a Cisco IOS voice gateway, or a Cisco Unified Border Element (Cisco UBE) to reset the expires timer upon receipt of a Session Initiation Protocol (SIP) 183 Session In Progress message, use the voice-class sip reset timer expires command in dial peer voice configuration mode. To globally disable resetting of the expires timer upon receipt of SIP 183 messages, use the no form of this command.

voice-class sip reset timer expires 183

no voice-class sip reset timer expires 183

Syntax Description

Command Default

The expires timer is not reset after receipt of SIP 183 Session In Progress messages and a session or call that is not connected within the default expiration time (three minutes) is dropped.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

In some scenarios, early media cut-through calls (such as emergency calls) rely on SIP 183 with session description protocol (SDP) Session In Progress messages to keep the session or call alive until receiving a FINAL SIP 200 OK message, which indicates that the call is connected. In these scenarios, the call can time out and be dropped if it does not get connected within the default expiration time (three minutes).

Note The expires timer default is three minutes. However, you can configure the expiration time to a maximum of 30 minutes using the timers expires command in SIP user agent (UA) configuration mode.

To prevent early media cut-through calls from being dropped on a specific dial peer because they reach the expires timer limit, use the voice-class sip reset timer expires command in dial peer voice configuration mode.

To globally configure all dial peers on Cisco Unified CME, a Cisco IOS voice gateway, or a Cisco UBE so that the expires timer is reset upon receipt of any SIP 183 message, use the reset timer expires command in voice service SIP configuration mode. To disable resetting of the expires timer on receipt of SIP 183 messages for an individual dial peer, use the no voice-class sip reset timer expires command in dial peer voice configuration mode.

Examples

The following example shows how to configure dial peer 1 on Cisco Unified CME, a Cisco IOS voice gateway, or a Cisco UBE to reset the expires timer each time a SIP 183 message is received:

Router> enable

Router# configure terminal

Router(config)# dial-peer voice 1 voip

Router(config-dial-peer)# voice-class sip reset timer expires 183

Related Commands

voice-class sip resource priority mode (dial peer)

To push the user access server (UAS) to operate in a loose or strict mode, use the voice-class sip resource priority mode command in dial peer voice configuration mode. To disable the voice-class sip resource priority mode, use the no form of this command.

voice-class sip resource priority mode [loose | strict]

no voice-class sip resource priority mode [loose | strict]

Syntax Description

Command Default

The default value is loose mode.

Command Modes

Dial peer voice configuration

Command History

Usage Guidelines

When the no version of this command is executed, the call operates in the loose mode.

Examples

The following example shows how to set up the voice-class sip resource priority mode command in loose mode:

Router(config)# dial-peer voice 102 voip

Router(config-dial-peer)# voice-class sip resource priority mode loose

The following example shows how to set up the voice-class sip resource priority mode command in strict mode:

Router(config)# dial-peer voice 102 voip

Router(config-dial-peer)# voice-class sip resource priority mode strict

Related Commands

voice-class sip resource priority namespace (dial peer)

To prioritize mandatory call prioritization handling for initial original INVITE message requests, use the voice-class sip resource priority namespace command in dial peer voice configuration mode. To disable the voice-class sip resource priority namespace command, use the no form of this command.

voice-class sip resource priority namespace [drsn | dsn | q735]

no voice-class sip resource priority namespace [drsn | dsn | q735]

Syntax Description

Command Default

When the no version of this command is executed using namespace, the Cisco IOS gateway transparently passes the multilevel precedence and preemption (MLPP) values that were received on the PSTN side.

Command Modes

Dial peer voice configuration

Command History

Usage Guidelines

When the no version of this command is executed using the namespace, the Cisco IOS gateway transparently passes the multilevel precedence and preemption (MLPP) values that were received on the PSTN side.

Examples

The following example shows how to set up the voice-class sip resource priority namespace command in the U. S. DSN format name space:

Router(config)# dial-peer voice 102 voip

Router(config-dial-peer)# voice-class sip resource priority namespace dsn

The following example shows how to set up the voice-class sip resource priority namespace command in the U. S. DRSN format name space:

Router(config)# dial-peer voice 102 voip

Router(config-dial-peer)# voice-class sip resource priority namespace drsn

The following example shows how to set up the voice-class sip resource priority namespace command in the Public SS7 Network format name space:

Router(config)# dial-peer voice 102 voip

Router(config-dial-peer)# voice-class sip resource priority namespace q735

Related Commands

voice-class sip rsvp-fail-policy

To specify the action that takes place at the dial peer level on a Cisco IOS Session Initiation Protocol (SIP) gateway when Resource Reservation Protocol (RSVP) negotiation fails, use the voice-class sip rsvp-fail-policy command in dial peer configuration mode. To reset failure behavior to the default settings, use the no form of this command.

voice-class sip rsvp-fail-policy {video | voice} post-alert {optional keep-alive | mandatory {keep-alive | disconnect retry retry-attempts}} interval seconds

no voice-class sip rsvp-fail-policy {video | voice} post-alert {optional [keep-alive] | mandatory [keep-alive | disconnect retry retry-attempts]} [interval seconds]

Syntax Description

Command Default

Keepalive messages are sent at 30-second intervals when a post alert voice or video call fails to negotiate RSVP regardless of the RSVP negotiation setting (mandatory or optional).

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Usage Guidelines

Use this command to configure call handling behavior when a call fails RSVP negotiation. You can configure the behavior that takes place for either optional or mandatory RSVP negotiation but the behavior will apply only to calls in a post alert call state. To configure the behavior that takes place when RSVP negotiation fails, use the voice-class sip rsvp-fail-policy command in dial peer configuration mode.

If a call fails RSVP negotiation where negotiation is optional, then RSVP negotiation should be retried using the keepalive function at specified intervals until RSVP negotiation is successful.

If a call fails RSVP negotiation where negotiation is mandatory, then RSVP negotiation should be configured in one of two ways:

•The call that failed RSVP negotiation is disconnected after a specified number of attempts to renegotiate RSVP with each retry taking place at a specified interval. If negotiation succeeds during these retry attempts, counters and timers are reset to zero.

•The call that failed RSVP negotiation is kept alive with keepalive messages sent at specified intervals until negotiation is successful.

Examples

The following example shows how to specify sending of keepalive messages at 60-second intervals for a call that fails RSVP negotiation when negotiation is optional:

Router(config)# dial-peer voice 102 voip

Router(config-dial-peer)# voice-class sip rsvp-fail-policy voice post-alert optional 
keep-alive interval 60

Related Commands

voice-class sip tel-config to-hdr

To configure the To: Header (to hdr) request Uniform Resource Identifier (URI) to telephone (TEL) format for dial-peer VoIP Session Initiation Protocol (SIP) calls, use the voice-class sip tel-config to-hdr command in dial peer voice configuration mode. To reset to the default, use the no form of this command.

voice-class sip tel-config to-hdr {phone-context | system}

no voice-class sip tel-config to-hdr

Syntax Description

Command Default

The To: Header request URIs at the dial-peer level use the global configuration level settings.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

The voice-class sip tel-config to-hdr command takes precedence over the tel-config to-hdr command configured in SIP configuration mode. However, if the voice-class sip tel-config to-hdr command is used with the system keyword, the gateway uses the global settings configured by the tel-config to-hdr command.

Examples

The following example configures the To: header in TEL format for a dial peer VoIP SIP call, and appends the phone-context parameter:

dial-peer voice 102 voip

 voice-class sip tel-config to-hdr phone-context

Related Commands

voice-class sip transport switch

To enable switching between UDP and TCP transport mechanisms for large Session Initiation Protocol (SIP) messages for a specific dial peer, use the voice-class sip transport switch command in dial peer configuration mode. To disable switching between UDP and TCP transport mechanisms for large SIP messages for a specific dial peer, use the no form of this command.

voice-class sip transport switch udp tcp

no voice-class sip transport switch udp tcp

Syntax Description

Command Default

Disabled.

Command Modes

Dial peer configuration

Command History

Usage Guidelines

The voice-class sip transport switch command takes precedence over the global transport switch command.

Examples

The following example shows how to set up the voice-class sip transport switch command:

Router(config)# dial-peer voice 102 voip

Router(config-dial-peer)# voice-class sip transport switch udp tcp 

Related Commands

voice-class sip url

To configure URLs to either the Session Initiation Protocol (SIP), SIP security (SIPS), or telephone (TEL) format for your dial-peer SIP calls, use the voice-class sip url command in dial peer voice configuration mode. To reset to the default value use the no form of this command.15.0(1)M

voice-class sip url {sip | sips | tel [phone-context] | system}

no voice-class sip url

Syntax Description

Command Default

SIP calls at the dial-peer level use the global configuration level settings.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

This command affects only user-agent clients (UACs), because it causes the use of a SIP, SIPS, or TEL URL in the request line of outgoing SIP INVITE requests. SIP URLs indicate the originator, recipient, and destination of the SIP request; TEL URLs indicate voice-call connections.

The voice-class sip url command takes precedence over the url command configured in SIP configuration mode. However, if the voice-class sip url command is used with the system keyword, the gateway uses what was globally configured with the url command.

Examples

The following example shows how to configure the voice-class sip url command to generate URLs in the SIP format:

dial-peer voice 102 voip

 voice-class sip url sip

The following example shows how to configure the voice-class sip url command to generate URLs in the SIPS format:

dial-peer voice 102 voip

 voice-class sip url sips

The following example shows how to configure the voice-class sip url command to generate URLs in the TEL format:

dial-peer voice 102 voip

 voice-class sip url tel

The following example shows how to configure the voice-class sip url command to generate URLs in the TEL format, and append the phone-context parameter:

dial-peer voice 102 voip

 voice-class sip url tel phone-context

Related Commands

voice-class source interface

To allow a loopback interface to be associated with a VoIP or VoIPv6 dial-peer profile, use the voice-class source interface command in dial peer configuration mode. To disable this association, use the no form of this command.

voice-class source interface loopback interface-id [ipv4-address | ipv6-address]

no voice-class source interface loopback interface-id [ipv4-address | ipv6-address]

Syntax Description

Command Default

No loopback interface is associated with a VoIPv6 dial-peer profile.

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Usage Guidelines

When the voice-class source interface command is configured, the source address of Routing Table Protocol (RTP) generated by the gateway is taken from the address configured under the loopback interface. This command is used for policy-based routing (PBR) of voice packets originated by the gateway. The policy route map is configured under the loopback interface, and then the loopback interface is specified under the VoIP or VoIPv6 dial peer.

Examples

The following example associates a loopback interface with a VoIPv6 dial-peer profile:

Router(config)# dial-peer voice 1 voip

Router (config-dial-peer)# voice-class source interface loopback0

Related Commands

voice-class stun-usage

To configure voice class, enter voice class configuration mode called stun-usage and use the voice-class stun-usage command in global, dial-peer, ephone, ephone template, voice register pool, or voice register pool template configuration mode. To disable the voice class, use the no form of this command.

voice-class stun-usage tag

no voice-class stun-usage tag

Syntax Description

Command Default

The voice class is not defined.

Command Modes

Global configuration (config)
Dial peer configuration (config-dial-peer)
Ephone configuration (config-ephone)
Ephone template configuration (config-ephone-template)
Voice register pool configuration (config-register-pool)
Voice register pool template configuration (config-register-pool)

Command History

Usage Guidelines

When the voice-class stun-usage is removed, the same is removed automatically from the dial-peer, ephone, ephone template, voice register pool, or voice register pool template configurations.

Examples

The following example shows how to set the voice class stun-usage tag to 10000:

Router(config)# voice class stun-usage 10000

Router(config-ephone)# voice class stun-usage 10000

Router(config-voice-register-pool)# voice class stun-usage 10000

Related Commands

voice-class stun-usage (dial peer)

To enable firewall traversal for VoIP communications, use the voice-class stun-usage command in dial peer voice configuration mode. To disable firewall traversal, use the no form of this command.

voice-class stun-usage tag

no voice-class stun-usage tag

Syntax Description

Command Default

Firewall traversal is not enabled.

Command Modes

Dial-peer voice configuration (config-dial-peer).

Command History

Usage Guidelines

When the voice-class stun-usage command is removed, the same is removed automatically from dial-peer configurations.

Examples

The following example shows how to set the voice-class stun-usage tag to 10.

Router(config)#dial-peer voice 1 voip

Router(config-dial-peer)#voice-class stun-usage 10

Related Commands

voice-class tone-signal

To assign a previously configured tone-signal voice class to a voice port, use the voice-class tone-signal command in voice-port configuration mode. To delete a tone-signal voice class, use the no form of this command.

voice-class tone-signal tag

no voice-class tone-signal tag

Syntax Description

Command Default

Voice ports have no tone-signal voice class assigned.

Command Modes

Voice-port configuration

Command History

Usage Guidelines

The voice-class tone-signal command is available on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). Note that the hyphenation in this command differs from the hyphenation used in a similar command, voice class tone-signal, which is used in global configuration mode.

Examples

The following example assigns a previously configured voice class to voice port 1/1/0:

voice-port 1/0/0

 voice-class tone-signal mytones

Related Commands

voice confirmation-tone

To disable the two-beep confirmation tone for private line, automatic ringdown (PLAR), or PLAR off-premises extension (OPX) connections, use the voice confirmation-tone command in voice-port configuration mode. To enable the two-beep confirmation tone, use the no form of this command.

voice confirmation-tone

no voice confirmation-tone

Syntax Description

This command has no arguments or keywords.

Command Default

The two-beep confirmation tone is heard on PLAR and PLAR OPX connections.

Command Modes

Voice-port configuration

Command History

Usage Guidelines

Use this command to disable the two-beep confirmation tone that a caller hears when picking up the handset for PLAR and PLAR OPX connections. This command is valid only if the voice-port connection command is set to PLAR or PLAR OPX.

Examples

The following example disables the two-beep confirmation tone on voice port 1/0/0:

voice-port 1/0/0

 connection plar-opx

 voice confirmation-tone

Related Commands

voice dnis-map

To create or modify a Digital Number Identification Service (DNIS) map, use the voice dnis-map command in global configuration mode. To delete a DNIS map, use the no form of this command.

voice dnis-map map-name [url]

no voice dnis-map map-name

Syntax Description

Command Default

No default behavior or values

Command Modes

Global configuration

Command History

Usage Guidelines

A DNIS map is a table of DNIS numbers associated with a single dial peer. For applications such as VoiceXML, using a DNIS map makes it possible to configure a single dial peer for all DNIS numbers used to refer to VoiceXML documents. Keep the following considerations in mind when using voice DNIS maps.

•A separate entry must be made for each DNIS entry in a DNIS map. Wildcards are not supported.

•If a URL is not supplied, the command enters DNIS-map configuration mode, permitting the entry of DNIS numbers by using the dnis command.

•The URL argument points to the location of an external text file containing a list of DNIS entries (for example: tftp://dnismap.txt). This allows the administrator to maintain a single master file of all DNIS map entries, if desired, rather than configuring the DNIS entries on each gateway.

The name of the text file extension is not significant; .doc, .txt, or .cfg are all acceptable because the extension is not checked. The entries in the file should look the same as a DNIS entry configured in Cisco IOS software (for example: dnis 5553305 url tftp://global/tickets/movies.vxml).

•External text files used for DNIS maps must be stored on TFTP servers; they cannot be stored on HTTP servers.

•To associate a DNIS map with a dial peer, use the dnis-map command.

•To view the configuration information for DNIS maps, use the show voice dnis-map command.

Examples

The following example shows how the voice dnis-map command is used to create a DNIS map:

voice dnis-map dmap1

The following example shows the voice dnis-map command used with a URL that specifies the location of a text file containing the DNIS entries:

voice dnis-map dmap2 tftp://keyer/dmap2/dmap2.txt

Following is an example of the contents of a text file comprising a DNIS map:

!Example dnis-map with 8 entries.

!

dnis 5550112 url tftp://global/ticket/vapptest1.vxml

dnis 5550111 url tftp://global/ticket/vapptest2.vxml

dnis 5550134 url tftp://global/ticket/vapptest3.vxml

dnis 5550178

dnis 5550100

dnis 5550101

dnis 5550102

dnis 5550103

Related Commands

voice dnis-map load

To reload a DNIS map that has been modified, use the voice dnis-map load command in privileged EXEC mode. This command does not have a no form.

voice dnis-map load map-name

Syntax Description

Command Default

No default behavior or values

Command Modes

Privileged EXEC

Command History

Usage Guidelines

This command reloads a DNIS map residing on an external server. Use this command when the DNIS map file has changed since the previous load.

To create or modify a DNIS map, use the voice dnis-map command.

Examples

The following example reloads a DNIS map named "mapfile1":

Router# voice dnis-map load mapfile1

Related Commands

voice dsp crash-dump

To enable the crash dump feature and to specify the destination file and the file limit, enter the
voice dsp crash-dump command in global configuration mode. To disable the feature, use the no form of the command.

voice dsp crash-dump [destination url | file-limit limit-number]

no voice dsp crash-dump

Syntax Description

Command Default

Crash dump capability is turned off.

Command Modes

Global configuration

Command History

Usage Guidelines

To configure the router to write a crash dump file, the destination url in the voice dsp crash-dump command must be set to a valid file system, and the crash dump file limit must be set to a non-zero value. The default is that the crash dump capability is turned off, as the url field is empty, and the file limit is zero.

As each crash-dump file is created, the name of the file has a number appended to the end. This number is incremented from 1 to up to the file limit for each subsequent crash dump file written. If the router reloads, the number is reset back to 1, and so file number 1 is written again. After the file number reaches the maximum file limit, no more files are written.

The file count can be manually reset by setting the file limit to zero and then setting it to a non-zero limit. This has the effect of restarting the count of files written, causing the files 1 to the file limit of 99 to be able to be written again, thus overwriting the original files.

Setting the file-number argument to zero (the default) disables the collection of the dump from the DSP. In this case, the memory is not collected from the DSP, and the stack is not displayed on the console. If the keepalive mechanism detects a crashed DSP, the DSP is simply restarted.

Setting the file-number argument to a non-zero number but having a null destination url causes the dump to be collected and the stack to be displayed on the console, but no dump file is written.

If auto-recovery is turned off for the router, no DSP dump functions are enabled, no keepalive checks are done, and no dumps are collected or written.

Note Some types of flash need to be completely erased to free up space from deleted files, and some types of flash cannot have files overwritten with new versions until the entire flash is erased. As a result, you might want to set the file limit so that only one or two dump files are written to flash. This prevents flash from being filled up.

Note It is not recommended to write crash dump files to internal flash or bootflash, because these files are normally used to hold configuration information and Cisco IOS software images. Cisco recommends writing crash dump files to spare flash cards, which can be inserted into slot 0 or slot 1 on many of the routers. These cards usually do not hold critical information and may be erased. Additionally, these cards can be conveniently removed from the router and sent to Cisco, so that the crash dump files can be analyzed.

Examples

The following example enables the crash dump feature and specifies the destination file in slot 0:

Router configure terminal 

Enter configuration commands, one per line.  End with CNTL/Z. 

Router(config)# voice dsp crash-dump destination slot0:banjo-152-s 

Router# end 

1w0d:%SYS-5-CONFIG_I:Configured from console by console 

Check your configuration by entering the show voice dsp crash-dump command in privileged EXEC configuration mode:

Router# show voice dsp crash-dump 

Voice DSP Crash-dump status:

    Destination file url is slot0:banjo-152-s 

File limit is 20 

    Last DSP dump file written was 

          tftp://112.29.248.12/tester/26-152-t2 

    Next DSP dump file written will be slot0:banjo-152-s1 

Related Commands

voice echo-canceller extended

To enable the extended G.168 echo canceller (EC) on the Cisco 1700 series, Cisco ICS7750, or Cisco AS5300, use the voice echo-canceller extended command in global configuration mode. To reset to the default, use the no form of this command.

Cisco 1700 series and Cisco ICS 7750

voice echo-canceller extended

no voice echo-canceller extended

Cisco AS5300

voice echo-canceller extended [codec small codec large codec]

no voice echo-canceller extended

Syntax Description

Command DefaultV

Proprietary Cisco G.165 EC is enabled.

Command Modes

Global configuration

Command History

Usage Guidelines

Cisco 1700 series and Cisco ICS7750

You do not have to shut down all the voice ports on the Cisco 1700 series or Cisco ICS7750 to switch the echo canceller, but you should make sure that when you switch the echo canceller, there are no active calls on the router.

Because echo cancellation is an invasive process that can minimally degrade voice quality, you should disable this command if it is not needed.

Cisco AS5300

This command is available only on the Cisco AS5300 with C542 or C549 digital signal processor module (DSPM) high-complexity firmware.

The voice echo-canceller extended command enables the extended EC on a Cisco AS5300 using C549 DSP firmware with one channel of voice per DSP and unrestricted codecs. Any codec is supported.

The voice echo-canceller extended codec command enables the extended EC on a Cisco AS5300 using C542 or C549 DSP firmware with two channels of voice per DSP and restricted codecs. Only specific codecs can be used with the extended EC.

If fax-relay is not selected as the large codec, the VoIP dial peer requires that you use the
fax rate disabled command in dial peer configuration mode.

After choosing the codecs to be supported by the extended echo canceller, either remove all dial peers with different codecs not supported by your new configuration or modify the dial-peer codec selection by selecting a voice codec or fax-relay. When codecs are restricted, only one selection is allowed. You must have a VoIP dial peer configured with an extended EC-compatible codec to ensure voice quality on the connection.

This command is not accepted if there are active calls. If the EC is already in effect and a codec choice is changed, the system scans the dial peers. Any dial peers that do not conform to the new global command settings are changed, and the user is informed of the changes. Similarly, modem relay is incompatible with the extended EC and must be disabled globally for all dial peers.

Note This command is valid only when the echo-cancel enable command and the echo-cancel coverage command are enabled.

Examples

The following example sets the extended G.168 EC on the Cisco 1700 series or Cisco ICS7750:

Router(config)# voice echo-canceller extended

The following example sets the extended G.168 EC on the Cisco AS5300 with restricted codecs:

Router(config)# voice echo-canceller extended codec small g711 large g726

The following example shows an error message that displays when a restricted codec is not allowed:

Cannot configure now, dial-peer 8800 is configured with codec=g728, fax rate=disable, 
modem=passthrough system.If necessary set this command to 'no', re-configure dial-peer 
codec, fax rate and/or modem. Then re-enter this command.

In the above example, dial peer 8800 is misconfigured with a codec type, g728, that was not selected for the large codec type using the voice echo-canceller extended command.

Related Commands

voice enum-match-table

To create an ENUM match table for voice calls, use the voice enum-match-table in global configuration mode. To delete the ENUM match table, use the no form of this command.

voice enum-match-table table-number

no voice enum-match-table table-number

Syntax Description

Command Default

No default behavior or values

Command Modes

Global configuration

Command History

Usage Guidelines

The ENUM match table is a set of rules for matching incoming calls. When a call comes in, its called number is matched against the match pattern of the rule with the highest preference.

If it matches, the replacement pattern is applied to the number. The resulting number and the domain name of the rule are used to make an ENUM query.

If the called number does not match the match pattern, the next rule in order of preference is selected.

Examples

The following example creates ENUM match table 3 for voice calls:

Router(config)# voice enum-match-table 3

Router(config-enum)# rule 1 5/(.*)/ /\1/e164.cisco.com

Router(config-enum)# rule 2 4/^9011\(.*\)/ /\1/e164.arpa

In this table, rule 1 matches any number. The resulting number is the same as the called number. That number and the domain name "e164.cisco.com" are used to make an ENUM query.

Rule 2 matches any number that starts with 9011. The 9011 is removed from the incoming number. The resulting number and the domain name "e164.arpa" are used for the ENUM query.

Suppose an incoming call has a called number of 4085551212. [Rule 2 is applied] first because it has a higher preference. The first few digits, 4085, do not match the 9011 pattern of rule 2, so [rule 1 is applied] next. The called number matches rule 1, and the resulting number is 4085551212. This number and "e164.cisco.com" form the ENUM query (2.1.2.1.5.5.5.8.0.4.e164.cisco.com).

Related Commands

voice hpi capture

To allocate the Host Port Interface (HPI) capture buffer size (in bytes) and to set up or change the destination URL for captured data, use the voice hpi capture command in global configuration mode. To stop all logging and file operations, to disable data transport from the capture buffer, and to automatically set the buffer size to 328, use the no form of this command.

voice hpi capture [buffer size | destination url]

no voice hpi capture buffer size

Syntax Description

Command Default

328 bytes (no buffer is used if it is not configured explicitly)

Command Modes

Global configuration

Command History

Usage Guidelines

If you want to change the size of an existing non-zero buffer, you must first reset it to 0 and then change it from 0 to the new size.

The destination url option sets up or changes the destination URL for captured data. To disable data transport from the capture buffer, use the no form of the command. If the buffer is allocated, captured data is sent to the current URL (if it was already configured) until the new URL is specified.

If a new URL differs from the current URL and logging is enabled, the current URL is closed and all further data is sent to the new URL. Entering a blank URL or prefixing the command with no disables data transport from the capture buffer, and (if capture is enabled) captured data is stored in the capture buffer until it reaches its capacity.

Once the buffer-queueing program is running, the transport process attempts to connect to a new or existing "capture destination" URL. A version message is written to the URL, and if the message is successfully received, any further messages placed into the message queue are written to that URL. If a new URL is entered using the voice hpi capture destination url command, the open URL is closed, and the system attempts to write to the new URL. If the new URL does not work, the transport process exits. The transport process is restarted when another URL is entered or the system is restarted.

The buffer size option sets the maximum amount of memory (in bytes) that the capture system allocates for its buffers when it is active. The capture buffer is where the captured messages are stored before they are sent to the URL specified by the capture destination. The system is started by choosing the amount of memory (greater than 0 bytes) that the buffer-queueing system can allocate to the free message pool. HPI messages can then be captured until buffer capacity is reached. Entering 0 for the buffer size and prefixing the command with no stops all logging and file operations and automatically sets the buffer size to 0.

The voice hpi capture command can be saved with the router configuration so that the command is active during router startup. This allows you to capture the HPI messages sent during router bootup before the CLI is enabled. After you have configured the buffer size in the running configuration (valid range is from 328 to 9000000), save it to the startup configuration using the write command or to the TFTP server using the copy run tftp command.

Caution Using the message logger feature in a production network environment impacts CPU and memory usage on the gateway.

Examples

The following example changes the size (in bytes) of the HPI capture buffer and initializes the buffer-queueing program:

Router# configure terminal

Enter configuration commands, one per line.  End with CNTL/Z.

Router(config)# voice hpi capture buffer 40000

Router(config)# end

Router#

03:23:31:caplog:caplog_cli_interface:hpi capture buffer size set to 40000 bytes

03:23:31:caplog:caplog_logger_init:TRUE, Started task HPI Logger (PID 64)

03:23:31:caplog:caplog_cache_init:TRUE, malloc_named(39852), 123 elements (each 324 bytes 
big)

03:23:31:caplog:caplog_logger_proc:Attempting to open ftp://172.23.184.233/c:b-38-117

03:23:32:%SYS-5-CONFIG_I:Configured from console by console

Router#

The following example sets the capture destination by entering a destination URL using FTP:

Router# configure terminal

Enter configuration commands, one per line.  End with CNTL/Z.

Router(config)# voice hpi capture destination ftp://172.23.184.233/c:b-38-117a

Router(config)#   

04:05:10:caplog:caplog_cli_interface:hpi capture 
destination:ftp://172.23.184.233/c:b-38-117a

04:05:10:caplog:caplog_logger_init:TRUE, Started task HPI Logger (PID 19)

04:05:10:caplog:caplog_cache_init:Cache must be at least 324 bytes

04:05:10:caplog:caplog_logger_proc:Terminating...

Router(config)# end

Router#

Related Commands

voice hunt

To configure an originating or tandem router so that it continues dial-peer hunting if it receives a specified disconnect cause code from a destination router, use the voice hunt command in global configuration mode. To configure the router so that it stops dial-peer hunting if it receives a specified disconnect cause code (the default condition), use the no form of this command. To restore the default dial-peer hunt setting, use the default form of this command.

voice hunt {disconnect-cause-code | all}

no voice hunt {disconnect-cause-code | all}

default voice hunt

Syntax Description

Command Default

The router stops dial-peer hunting if it receives the user-busy or no-answer disconnect cause code.

Command Modes

Global configuration

Command History

Usage Guidelines

This command is used with routers that act as originating or tandem nodes in a VoIP, VoFR, or Voice over ATM environment.

For an outgoing call from an originating VoIP gateway configured for rotary dial-peer hunting, more than one dial peer may match the same destination number. The matching dial peers may have different routes. After the voice call using the first dial peer gets disconnected, it will return a disconnect cause code. To have the router to pick up the next matching dial peer in the rotary group and set up a call, the router must be configure to continue hunting the various routes. Use this command to configure the router's hunting behavior when specified cause codes are received.

You can use this command to enable and disable dial-peer hunting when nonstandard disconnect cause codes are received. Nonstandard disconnect cause codes are those that are not defined in ITU-T Recommendation Q.931, but are used by service providers. When this command is used to disable dial-peer hunting for a specific disconnect cause code, it appears in the running configuration of the router.

The disconnect cause codes are described in Table 251. The decimal and hexadecimal value of the disconnect cause code follows the description of each possible keyword.

Examples

The following example configures the originating or tandem router to continue dial-peer hunting if it receives a user-busy disconnect cause code from a destination router:

voice hunt user-busy

The following example configures the originating or tandem router to continue dial-peer hunting if it receives an invalid-number disconnect cause code from a destination router:

voice hunt 28

The following example configures the originating or tandem router to continue dial-peer hunting if it receives a facility-not-subscribed disconnect cause code from a destination router:

voice hunt 0x32

Related Commands

voice iec syslog

To enable viewing of Internal Error Codes as they are encountered in real time, use the voice iec syslog command in global configuration mode. To disable IEC syslog messages, use the no form of this command.

voice iec syslog

no voice iec syslog

Syntax Description

This command has no arguments or keywords.

Command Default

IEC syslog messages are disabled.

Command Modes

Global configuration

Command History

Examples

The following example enables IEC syslog messages:

router(config)# voice iec syslog

Related Commands

voice local-bypass

To configure local calls to bypass the digital signal processor (DSP), use the voice local-bypass command in global configuration mode. To direct local calls through the DSP, use the no form of this command.

voice local-bypass

no voice local-bypass

Syntax Description

This command has no arguments or keywords.

Command Default

Local calls bypass the DSP.

Command Modes

Global configuration

Command History

Usage Guidelines

Local calls (calls between voice ports on a router or concentrator) normally bypass the DSP to minimize use of system resources. Use the no form of the voice local-bypass command if you need to direct local calls through the DSP. Input gain and output attenuation can be configured only if calls are directed through the DSP.

Examples

The following example configures a Cisco router to pass local calls through the DSP:

no voice local-bypass

Related Commands

voice mlpp

To enter MLPP configuration mode to enable MLPP service, use the voice service command in global configuration mode. To disable MLPP service, use the no form of this command.

voice mlpp

no voice mlpp

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

Voice-mlpp configuration mode is used for the gateway globally.

Examples

The following example shows how to enter voice-mlpp configuration mode:

Router(config)# voice mlpp

Router(config-voice-mlpp)# access-digit

Related Commands

voicemail (stcapp-fsd)

To designate an SCCP telephony control (STC) application feature speed-dial code to speed dial the voice-mail number, use the voicemail command in STC application feature speed-dial configuration mode. To return the code to its default, use the no form of this command.

voicemail keypad-character

no voicemail

Syntax Description

Command Default

The default voice-mail code is 0 (zero) for one-digit codes; 00 (two zeros) for two-digit codes.

Command Modes

STC application feature speed-dial configuration

Command History

Usage Guidelines

This command is used with the STC application, which enables certain features on analog FXS endpoints that use Skinny Client Control Protocol (SCCP) for call control.

To use the speed-dial to voice-mail feature on a phone, dial the feature speed-dial (FSD) prefix and the code that has been configured with this command (or the default if this command was not used). For example, if the FSD prefix is * (the default), and you want to dial the voice-mail phone number, dial *0.

Note that the number that will be speed-dialed for voice mail must be set on Cisco CallManager or the Cisco CallManager Express system.

This command is reset to its default value if you modify the value of the digit command. For example, if you set the digit command to 2, then change the digit command back to its default of 1, the voice-mail FSD code is reset to 0 (zero).

If you set this code to a value that is already in use for another FSD code, you receive a warning message. If you configure a duplicate code, the system implements the first matching feature in the order of precedence shown in the output of the show stcapp feature codes command.

The show running-config command displays nondefault FSD codes only. The show stcapp feature codes command displays all FSD codes.

Examples

The following example sets an FSD prefix of two pound signs (##) and a voice-mail code of 8. After these values have been configured, a phone user presses ##8 to dial the voice-mail number.

Router(config)# stcapp feature speed-dial

Router(stcapp-fsd)# prefix ##

Router(stcapp-fsd)# voicemail 8

Router(stcapp-fsd)# exit

Related Commands

voiceport

To enable a private line automatic ringdown (PLAR) connection for an analog phone, use the voiceport command in SCCP PLAR configuration mode. To remove PLAR from the voice port, use the no form of this command.

voiceport port-number dial dial-string [digit dtmf-digits [wait-connect wait-msecs] [interval inter-digit-msecs]]

no voiceport port-number

Syntax Description

Command Default

Disabled (PLAR is not set for the voice port).

Command Modes

SCCP PLAR configuration

Command History

Usage Guidelines

This command enables PLAR on analog FXS ports that use Skinny Client Control Protocol (SCCP) for call control. If the digit keyword is not used, DTMF digits are not out-pulsed; the voice port uses a simple PLAR connection and the other keywords are not available.

Voice ports can be configured in any order. For example, you can configure port 2/23 before port 2/0. The show running-config command lists the ports in ascending order.

Before a PLAR port can become operational, the STC application must first be enabled in the corresponding dial-peer using the service stcapp command. If you configure a port for PLAR before enabling the STC application in the dial peer you receive a warning message.

PLAR phones support most of the same features as normal analog phones. The PLAR phone handles incoming calls and supports hookflash for basic supplementary features such as call transfer, call waiting, and conference. The PLAR phone does not support other features such as call forwarding, redial, speed dial, call park, call pick up from a PLAR phone, AMWI, or caller ID.

Examples

The following example enables the PLAR feature on port 2/0, 2/1, and 2/3. When a phone user picks up the handset on the analog phone connected to port 2/0, the system automatically rings extension 3660 and after waiting 500 milliseconds, dials 1234. The DTMF digits are out-pulsed to the destination port at an interval of 200 milliseconds.