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voice-class sip error-code-override through vxml version 2.0
- voice-class sip error-code-override
- voice-class sip g729 annexb-all
- voice-class sip history-info
- voice-class sip localhost
- voice-class sip map resp-code
- voice-class sip midcall-signaling
- voice-class sip options-keepalive
- voice-class sip outbound-proxy
- voice-class sip preloaded-route
- voice-class sip privacy
- voice-class sip privacy-policy
- voice-class sip random-contact
- voice-class sip random-request-uri validate
- voice-class sip referto-passing
- voice-class sip registration passthrough
- voice-class sip rel1xx
- voice-class sip reset timer expires
- voice-class sip resource priority dscp-profile
- voice-class sip resource priority mode (dial-peer)
- voice-class sip resource priority namespace (dial-peer)
- voice-class sip rsvp-fail-policy
- voice-class sip send 180 sdp
- voice-class sip srtp negotiate
- voice-class sip tel-config to-hdr
- voice-class sip transport switch
- voice-class sip url
- voice-class source interface
- voice-class stun-usage
- voice-class tone-signal
- voice confirmation-tone
- voice dnis-map
- voice dnis-map load
- voice dsp crash-dump
- voice echo-canceller extended
- voice enum-match-table
- voice hpi capture
- voice hunt
- voice iec syslog
- voice local-bypass
- voice mlpp
- voicemail (stcapp-fsd)
- voice pcm capture
- voiceport
- voice-port
- voice-port (MGCP profile)
- voice-port busyout
- voice rtp send-recv
- voice-service dsp-reservation
- voice service
- voice source-group
- voice statistics accounting method
- voice statistics display-format separator
- voice statistics field-params
- voice statistics max-storage-duration
- voice statistics push
- voice statistics time-range
- voice statistics type csr
- voice statistics type iec
- voice translation-profile
- voice translation-rule
- voice vad-time
- voice vrf
- voip-incoming translation-profile
- voip-incoming translation-rule
- voip trunk group
- volume
- vxml allow-star-digit
- vxml audioerror
- vxml tree memory
- vxml version 2.0
voice-class sip error-code-override
To configure the Session Initiation Protocol (SIP) error code that a dial peer uses for options-keepalive failures, call spike, or cac-bandwidth 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 | cac-bandwidth failure } { sip-status-code-number | system }
no voice-class sip error-code-override { options-keepalive failure | call spike failure | cac-bandwidth failure }
Syntax Description
options-keepalive failure
Configures the SIP error code for options-keepalive failures.
call spike failure
Configures the SIP error code for call spike failures.
cac-bandwidth failure
Configures the SIP error code for Call Admission Control bandwidth failures.
sip-status-code-number
The SIP status code that is sent for the options keepalive, call spike, or cac-bandwidth failure. The range is from 400 to 699. The default value is 503. The table below in the “Usage Guidelines” section describes these error codes.
system
Specifies the system configuration used for keepalive, call spike, or cac-bandwidth failures.
Usage Guidelines
The voice-class sip error-code-override command in dial peer voice configuration mode configures the error code response for keepalive options, call spike, or cac-bandwidth failures at the dial peer level. The error-code-override command in voice service SIP configuration mode configures the error code responses for options-keepalive, call spike, or cac-bandwidth failures globally.
The table below describes the SIP error codes.
Table 1 SIP Error Codes Error Code Number
Description
400
Bad request
401
Unauthorized
402
Payment required
403
Forbidden
404
Not found
408
Request timed out
416
Unsupported Uniform Resource Identifier (URI)
480
Temporarily unavailable
482
Loop detected
484
Address incomplete
486
Busy here
487
Request terminated
488
Not acceptable here
500–599
SIP 5xx—server/service failure
500
Internal server error
502
Bad gateway
503
Service unavailable
600–699
SIP 6xx—global failure
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 502The 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 502The following example shows how to configure the SIP error code for Call Admission Control bandwidth failures:
Router(config)# dial-peer voice 432 voip system Router(config-dial-peer)# voice-class sip error-code-override cac-bandwidth failure 502voice-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.
Syntax Description
annexb-all
Specifies that the G.729br8 codec is treated as a superset of G.729r8 and G.729br8 codecs to communicate with Cisco Unified Communications Manager.
system
(Optional) Specifies that the dial peer allow communication between incompatible G.729 codecs according to global settings configured for this feature on the Cisco IOS SIP gateway.
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-allvoice-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.
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.
NoteThe 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-infoThe 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 systemvoice-class sip localhost
To configure individual dial peers to override global settings on Cisco IOS voice gateways, Cisco Unified Border Element (Cisco UBE), 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
dns: [hostname.]domain
Alphanumeric value representing the DNS domain (consisting of the domain name with or without a specific hostname) in place of the physical IP address that is used in the host portion of the From, Call-ID, and Remote-Party-ID headers in outgoing messages.
This value can be the hostname and the domain separated by a period (dns: hostname.domain) or just the domain name (dns: domain). In both case, the dns: delimiter must be included as the first four characters.
preferred
(Optional) Designates the specified DNS hostname as preferred.
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 History
Release
Modification
12.4(2)T
This command was introduced.
15.0(1)XA
This command was modified. The preferred keyword was added to specify the preferred localhost if multiple registrars are configured on a SIP trunk.
IOS Release XE 2.5
This command was integrated into Cisco IOS XE Release 2.5.
15.1(1)T
This command was integrated into Cisco IOS Release 15.1(1)T.
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 preferredThe 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 preferredThe 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 localhostRelated Commands
Command
Description
authentication (dial peer)
Enables SIP digest authentication on an individual dial peer.
authentication (SIP UA)
Enables SIP digest authentication.
credentials (SIP UA)
Configures a Cisco UBE to send a SIP registration message when in the UP state.
localhost
Configures global settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages.
registrar
Enables Cisco IOS SIP gateways to register E.164 numbers on behalf of FXS, EFXS, and SCCP phones with an external SIP proxy or SIP registrar.
voice-class sip 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
181
The code representing the specific incoming SIP provisional response messages to be mapped and replaced.
to
The designator for specifying that the specified incoming SIP provisional response message should be mapped to and replaced with a different SIP provisional response message on the outgoing SIP dial peer.
183
The code representing the specific SIP provisional response message on the outgoing dial peer to which incoming SIP message responses should be mapped.
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.
Usage Guidelines
Use the voice-class sip map resp-codecommand 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.
NoteIf 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).
NoteThis 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 183Related Commands
Command
Description
block
Configures global settings for dropping specific SIP provisional response messages on a Cisco IOS voice gateway or Cisco UBE.
map resp-code
Configures global settings on a Cisco UBE for mapping specific incoming SIP provisional response messages to a different SIP response message.
voice-class sip block
Configures an individual dial peer on a Cisco IOS voice gateway or Cisco UBE to drop specified SIP provisional response messages.
voice-class sip midcall-signaling
To configure the method used for signaling messages, use the voice-class sip midcall-signaling command in SIP configuration mode or dial peer configuration mode. To disable the mid-call signaling feature, use the no form of this command.
voice-class sip midcall-signaling { passthru media-change | block | preserve-codec }
no voice-class sip midcall-signaling
Command Default
Mid call-signaling is disabled. Codec negotiation in the middle of a call is enabled.
Command History
Release
Modification
12.4(15)XZ
This command was introduced.
12.4(20)T
This command was integrated into Cisco IOS Release 12.4(20)T.
Cisco IOS XE Release 2.5
This command was integrated into Cisco IOS XE Release 2.5.
15.2(1)T
This command was integrated into Cisco IOS Release 15.2(1)T. The media-change and block keywords were added.
15.3(2)S, 15.3(1)T
This command was modified. The preserve-codec keyword was added.
Usage Guidelines
The voice-class sip midcall-signaling command distinguishes between the way Cisco Unified Communications Express and Cisco Unified Border Element handle signaling messages. Most SIP-to-SIP video and SIP-to-SIP reinvite based supplementary services require the voice-class sip midcall-signaling command to be configured before configuring other supplementary services. Supplementary service features that are functional without configuring voice-class sip midcall-signaling include: session refresh, fax, and refer-based supplementary services. The voice-class sip midcall-signaling command is for SIP-to-SIP calls only. All other calls (H323-to-SIP, and H323-to-H323) do not require the voice-class sip midcall-signalingcommand be configured. The allow-connections sip-to-sip command must be configured before the voice-class sip midcall-signaling command.
Configuring the Session Refresh with Reinvites feature on a dial-peer basis is not supported.
Examples
The following example shows SIP messages configured to passthrough from one IP leg to another IP leg:
Router(config)#voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# voice-class sip midcall-signaling passthruThe following example shows SIP messages configured to media passthru from one IP leg to another IP leg:
Router(config)#voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# voice-class sip midcall-signaling passthru media-changeThe following example shows how to block SIP messages.
Router(config)#voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# voice-class sip midcall-signaling blockThe following example shows how to disable codec negotiation in the middle of a call and retains the codec negotiated at the start of the call.
Router(config)#voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# voice-class sip midcall-signaling preserve-codecvoice-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
up-interval seconds
Number of up-interval seconds allowed to pass before marking the UA as unavailable.The range is 5-1200. The default is 60.
down-interval seconds
Number of down-interval seconds allowed to pass before marking the UA as unavailable.The range is 5-1200. The default is 30.
retry retries
Number of retry attempts before marking the UA as unavailable. The range is 1 to 10. The default is 5 attempts.
Usage Guidelines
Use the voice-class sip options-keepalivecommand 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:
Table 2 Error Codes that busyout the endpoint Error Code
Description
503
service unavailable
505
sip version not supported
no response
i.e. request timeout
All other error codes, including 400 are considered a valid response and the dial peer is not busied out.
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
dhcp
Specifies that the outbound-proxy IP address is retrieved from a DHCP server.
ipv4: ipv4-address
Configures proxy on the server, sending all initiating requests to the specified IPv4 address destination. The colon is required.
ipv6:[ ipv6- address ]
Configures proxy on the server, sending all initiating requests to the specified IPv6 address destination. Brackets must be entered around the IPv6 address. The colon is required.
dns: host:domain
Configures proxy on the server, sending all initiating requests to the specified domain destination. The colons are required.
: port-number
(Optional) Port number for the Session Initiation Protocol (SIP) server. The colon is required.
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-proxycommand 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 ipv4:10.1.1.1The following example shows how to configure the voice-class sip outbound-proxycommand 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.comThe following example shows how to configure the voice-class sip outbound-proxycommand 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 dhcpvoice-class sip preloaded-route
To enable preloaded route support for dial-peer Session Initiation Protocol (SIP) calls, use the voice-class sip preloaded-routecommand 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
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-routeThe 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-routevoice-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
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 criticalcommand 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 disableThe 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 historyRelated Commands
Command
Description
asserted-id
Sets the privacy level and enables either PAI or PPI privacy headers in outgoing SIP requests or response messages.
calling-info pstn-to-sip
Specifies calling information treatment for PSTN-to-SIP calls.
clid (voice-service-voip)
Passes the network-provided ISDN numbers in an ISDN calling party information element screening indicator field, removes the calling party name and number from the calling-line identifier in voice service voip configuration mode, or allows a presentation of the calling number by substituting for the missing Display Name field in the Remote-Party-ID and From headers.
privacy
Sets privacy support at the global level as defined in RFC 3323.
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
passthru
Passes the privacy values from the received message to the next call leg.
send-always
Passes a privacy header with a value of None to the next call leg, if the received message does not contain privacy values but a privacy header is required.
strip
Strip the diversion or history-info headers received from the next call leg.
diversion
Strip the diversion header received from the next call leg.
history-info
Strip the history-info header received from the next call leg.
system
(Optional) Uses the global configuration settings to configure the dial peer.
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 passthruThe 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-infoThe 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-alwaysThe 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-alwaysvoice-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.
Command Default
Support for random contact at the dial-peer level uses the the global configuration level settings.
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-contactRelated Commands
Command
Description
credentials (sip ua)
Sends a SIP registration message from a Cisco Unified Border Element in the UP state.
registrar
Enables SIP gateways to register E.164 numbers on behalf of FXS, EFXS, and SCCP phones with an external SIP proxy or SIP registrar.
random-contact
Populates the outgoing INVITE message with random contact information at the global level.
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.
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 validateRelated Commands
Command
Description
credentials (sip ua)
Sends a SIP registration message from a Cisco Unified Border Element in the UP state.
random-request-uri validate
Validates the called number based on the random value generated during the registration of the number at the global configuration level.
registrar
Enables SIP gateways to register E.164 numbers on behalf of FXS, EFXS, and SCCP phones with an external SIP proxy or SIP registrar.
voice-class sip referto-passing
To disable the modification of the Refer-To header during REFER message pass-through on the Cisco Unified Border Element (UBE) on the specified dial peer, use the voice-class sip referto-passing command in dial peer voice configuration mode. To allow the modification of the Refer-To header during REFER message pass-through on the Cisco UBE, use the no form of this command.
Usage Guidelines
The dial peer configuration setting of the voice-class sip referto-passing command takes precedence over the global configuration setting of the referto-passing command. You can use the system keyword to toggle the precedence.
Examples
The following example shows how to enable REFER message pass-through on the Cisco UBE for dial peer 22:
Router(config)# dial-peer voice 22 voip Router(config-dial-peer)# voice-class sip referto-passingRelated Commands
Command
Description
dial-peer voice
Defines a particular dial peer, specifies the method of encapsulation, and enters dial peer voice configuration mode.
referto-passing
Disables dial peer lookup and modification of the Refer-To header when the Cisco UBE passes across a REFER message during a call transfer
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 passthroughcommand 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
static
(Optional) Configures Cisco Unified Border Element (UBE) to use static registrar details for SIP registration. Cisco UBE works in point-to-point mode when the static keyword is used.
rate-limit
(Optional) Configures SIP registration pass-through rate-limiting options.
expires value
(Optional) Sets the expiry value for rate limiting, in seconds. The range is from 60 to 65535. The default is 3600.
fail-count value
(Optional) Sets the fail-count value for rate limiting. The range is from 2 to 20. The default is 0.
registrar-index
(Optional) Configures the registrar index used for registration pass-through.
index
(Optional) Registration index value. The range is from 1 to 6.
system
(Optional) Uses global registration pass-through configuration to configure the SIP registration pass-through options.
Command Default
SIP registration pass-through options that are configured at the global level are configured.
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.
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.
Syntax Description
supported value
Supports reliable provisional responses. The value argument may have any value, as long as both the user-agent client (UAC) and user-agent server (UAS) configure it the same.
require value
Requires reliable provisional responses. The value argument may have any value, as long as both the UAC and UAS configure it the same.
system
Uses the value configured in voice service mode. This is the default.
disable
Disables the use of reliable provisional responses.
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command was applicable to the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
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 rel1xxcommand.
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 100relvoice-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.
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.
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).
NoteThe 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 183Related Commands
Command
Description
reset timer expires
Globally configures Cisco Unified CME, a Cisco IOS voice gateway, or a Cisco UBE to reset the expires timer upon receipt of a SIP 183 message.
timers expires
Specifies how long a SIP INVITE request remains valid before it times out if no appropriate response is received for keeping the session alive.
voice-class sip resource priority dscp-profile
To apply a differentiated services code point (DSCP) profile to a dial peer, use the voice-class sip resource priority dscp-profile in dial peer voice configuration mode. To disable the configuration, use the no form of this command.
voice-class sip resource priority dscp-profile tag
no voice-class sip resource priority dscp-profile
Usage Guidelines
You can use the voice-class sip resource priority dscp-profile command to apply the DSCP profile that is configured using the dscp media command for a dial peer.
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
loose
(Optional) In the loose mode, unknown values of name space or priority values received in the Resource-Priority header in Session Initiation Protocol (SIP) requests are ignored by the gateway. The request is processed as if the Resource-Priority header was not present.
strict
(Optional) In the strict mode, unknown values of name space or priority values received in the Resource-Priority header in SIP requests are rejected by the gateway using a SIP response code 417 (Unknown Resource-Priority) message response. An Accept-Resource-Priority header enumerating the supported name space and values is included in the 417 message response.
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 looseThe 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 strictvoice-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
drsn
(Optional) U. S. Defense Red Switched Network (DRSN).
dsn
(Optional) U. S. Defense Switched Network (DSN).
q735
(Optional) International Telecommunications Union, Stage 3 description for community of interest supplementary services using Signaling System No. 7: Multilevel precedence and preemption, Recommendation Q.735.3 , March 1993.
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.
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 dsnThe 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 drsnThe 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 q735voice-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
video
Specifies the video RSVP stream type.
voice
Specifies the audio or fax RSVP stream type.
post-alert
Specifies that behavior takes place only when the call state is post alert.
optional
Specifies that behavior takes place when RSVP fails even if RSVP negotiation is optional.
mandatory
Specifies that behavior takes place when RSVP fails only if RSVP negotiation is mandatory.
keep-alive
Specifies the sending of keepalive messages when RSVP fails.
disconnect
Specifies that the call is disconnected if RSVP fails after the specified number of retry settings.
retry
Specifies the number of reconnection attempts before disconnecting the call.
retry-attempts
The number of retry attempts. Valid entries are from 1 to 100.
interval
Specifies the interval between keepalive or retry attempts.
seconds
The retry interval in seconds. Valid entries are from 5 to 3600.
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).
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 60Related Commands
Command
Description
acc-qos
Defines the acceptable QoS for inbound and outbound calls on a VoIP dial peer.
handle-replaces
Configures fallback to legacy handling of SIP INVITE.
ip qos defending-priority
Configures the RSVP defending priority value.
ip qos dscp
Sets the DSCP value for QoS.
ip qos policy-locator
Configures application-specific reservations (application IDs) used for specifying bandwidth reservations.
ip qos preemption-priority
Configures the RSVP preemption priority value.
req-qos
Requests a particular QoS using RSVP to be used in reaching a specified dial peer in VoIP.
show-sip-ua calls
Displays the active UAC and UAS information on SIP calls.
voice-class sip send 180 sdp
To configure a Cisco Unified Border Element (Cisco UBE) to map an incoming 180 Session Description Protocol (SDP) message to a 180 SDP message, use the voice-class sip send 180 sdp command in dial peer voice configuration mode or SIP configuration mode. To disable this functionality, use the no form of this command.
Command Default
This command is disabled. Cisco UBE converts an incoming 180 SDP message to a 183 SDP message.
Usage Guidelines
This command must be enabled at the inbound dial peer. Enable the voice-class sip send 180 sdp command to map a 180 SDP message to a 180 SDP message. When this command is disabled, an incoming 180 SDP (Ringing) message is mapped to a 183 SDP (Session in Progress) message.
voice-class sip srtp negotiate
To enable Secure Real-Time Transport Protocol (SRTP) negotiation so that an individual dial peer on a Cisco IOS Session Initiation Protocol (SIP) gateway can accept and send an RTP Audio/Video Profile (AVP) in response to an RTP Secure AVP offer (also known as an SRTP profile), use the voice-class sip srtp negotiate command in dial peer voice configuration mode. To return to the default (global) SRTP negotiation setting on a dial peer, use the system keyword. To disable SRTP negotiation on a dial peer, use the no form of this command.
Syntax Description
cisco
Enables an individual dial peer on a Cisco IOS SIP gateway to negotiate the sending and accepting of RTP profiles in response to SRTP offers, overriding the global setting for the gateway.
system
Specifies that the individual dial peer use global (system) SRTP negotiation settings for the Cisco IOS SIP gateway. This is the default setting.
Command Default
SRTP negotiation is determined by global settings for the Cisco IOS gateway (voice-class sip srtp negotiate system).
Usage Guidelines
The srtp fallback command enables a SIP gateway (or individual dial peer on a SIP gateway) to allow SRTP fallback using SIP 4xx message responses. With the srtp negotiate command, a SIP gateway can be configured to accept and send an RTP (nonsecure) profile in response to an SRTP profile.
Use the voice-class sip srtp negotiate command in dial peer voice configuration mode to enable SRTP negotiation for an individual dial peer on a Cisco IOS SIP gateway, overriding the global settings on the gateway. Enabling SRTP negotiation allows a dial peer to accept and send nonsecure RTP profiles in response to SRTP offers. To configure global SRTP negotiation settings for a SIP gateway, use the srtp negotiate command in voice service SIP configuration mode.
There are two scenarios for SRTP negotiation when the voice-class sip srtp negotiate command is enabled:
- On a SIP dial peer with the srtp fallback command enabled, the dial peer accepts RTP answers to SRTP offers.
- On a SIP dial peer with the srtp fallback command disabled, the dial peer allows incoming SRTP calls and responds with an RTP answer.
These behaviors are accomplished using the “X-cisco-srtp-fallback” extension in the supported header of initial SIP messages involved in establishment of the session.
Examples
The following example shows SRTP negotiation being enabled on a dial peer, overriding global settings:
Device(config)# dial-peer voice 1 Device(config-dial-peer)# voice-class sip srtp negotiate ciscoRelated Commands
Command
Description
srtp (dial peer)
Specifies that an individual dial peer use SRTP to enable secure calls and, optionally, enables fallback to RTP (overriding global settings).
srtp (voice)
Specifies use of SRTP to enable secure calls and, optionally, enables fallback to RTP globally on a Cisco IOS SIP gateway.
srtp negotiate
Enables SRTP negotiation globally on a Cisco IOS SIP gateway.
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-hdrcommand in dial peer voice configuration mode. To reset to the default, use the no form of this command.
Command Default
The To: Header request URIs at the dial-peer level use the global configuration level settings.
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.
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 switchcommand 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.
Usage Guidelines
The voice-class sip transport switchcommand takes precedence over the global transport switchcommand.
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.
Syntax Description
sip
Generates URLs in the SIP format for calls on a dial-peer basis.
sips
Generates URLs in the SIPS format for calls on a dial-peer basis.
tel
Generates URLs in the TEL format for calls on a dial-peer basis.
phone-context
(Optional) Appends the phone context parameter to the TEL URL on a dial-peer basis.
system
Uses the system value. This is the default.
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 was not included in this release.
12.2(11)T
This command was implemented on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 platforms.
12.4(6)T
The sips keyword was added.
12.4(22)YB
The phone-context keyword was added.
15.0(1)M
This command was integrated into Cisco IOS Release 15.0(1)M.
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 urlcommand to generate URLs in the SIP format:
dial-peer voice 102 voip voice-class sip url sipThe following example shows how to configure the voice-class sip urlcommand to generate URLs in the SIPS format:
dial-peer voice 102 voip voice-class sip url sipsThe following example shows how to configure the voice-class sip urlcommand to generate URLs in the TEL format:
dial-peer voice 102 voip voice-class sip url telThe following example shows how to configure the voice-class sip urlcommand to generate URLs in the TEL format, and append the phone-context parameter:
dial-peer voice 102 voip voice-class sip url tel phone-contextvoice-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 ]
Usage Guidelines
When the voice-class source interfacecommand 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.
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.
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)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.
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.
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.
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.
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.
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.
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 (forexample: 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 dmap1The 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.txtFollowing 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 5550103voice 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.
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.
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-dumpcommand in global configuration mode. To disable the feature, use the no form of the command.
Syntax Description
destination url
Designates a valid file system where crash dump analysis is stored. The url argument must be set to a valid file system.
The destination url can be one of the following
- The file on a TFTP server with the following format: tftp://x.x.x.x/subfolder/filename.
The x.x.x.x value is the IP address of the TFTP server
- The file on the flashcard of the router, with the following format: slot0:filename
Note The digital signal processor (DSP) crash dump feature is disabled when either the crash-dump destination is not specified.
file-limit limit-number
The crash dump file-limit keyword must be set to a non-zero value. The default is that the crash dump capability is turned off, as the url argument is empty, and the file-number argument is zero.
The limit-number argument may range from 0 (no file will be written) to 99.
Note The DSP crash dump feature is disabled when the crash-dump file limit is set to 0.
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.
NoteSome 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.
NoteIt 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 consoleCheck 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-s1voice 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 extendedcommand 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
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.
NoteThis command is valid only when the echo-cancel enable command and the echo-cancel coveragecommand 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 extendedThe 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 g726The 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.
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 noform of this command.
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.arpaIn 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 4085550112. [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 4085550112. This number and "e164.cisco.com" form the ENUM query (2.1.2.1.5.5.5.8.0.4.e164.cisco.com).
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.
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 destinationurl 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#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 huntcommand 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
disconnect-cause-code
A code returned from the destination router to indicate why an attempted end-to-end call was unsuccessful. If the specified disconnect cause code is returned from the last destination endpoint, dial peer hunting is enabled or disabled. The table below in the "Usage Guidelines" section describes the possible values. You can enter the keyword, decimal value, or hexadecimal value.
all
Continue dial-peer hunting for all disconnect cause codes returned from the destination endpoint.
default
Restores the default dial-peer hunt setting, that is, the router stops dial-peer hunting if it receives the user-busy or no-answer disconnect cause code.
Command Default
The router stops dial-peer hunting if it receives the user-busy or no-answer disconnect cause code.
Command History
Release
Modification
12.0(5)T
This command was introduced for VoFR on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810. It was also introduced for VoIP on the Cisco 2600 series and Cisco 3600 series.
12.0(7)T
This command was implemented for VoIP on the Cisco AS5300 and Cisco AS5800.
12.0(7)XK
This command was implemented for VoIP on the Cisco MC3810.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T and implemented for VoIP on the Cisco MC3810.
12.1(3)XI
The invalid-numberand unassigned-number keywords were added, and the command name was changed to voice hunt.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
12.2(4)T
Keywords were added for more disconnect cause codes.
12.3(8)T
The disconnect-cause-code argument was modified to accept nonstandard disconnect cause codes.
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 the table below. The decimal and hexadecimal value of the disconnect cause code follows the description of each possible keyword.
Table 3 Standard Disconnect Cause Codes Keyword
Description
Decimal
Hex
access-info-discard
Access information discarded.
43
0x2b
all
Continue dial-peer hunting for all disconnect cause codes received from a destination router.
b-cap-not-implemented
Bearer capability not implemented.
65
0x41
b-cap-restrict
Restricted digital information bearer capability only.
70
0x46
b-cap-unauthorized
Bearer capability not authorized.
57
0x39
b-cap-unavail
Bearer capability not available.
58
0x3a
call-awarded
Call awarded.
7
0x7
call-cid-in-use
Call exists, call ID in use.
83
0x53
call-clear
Call cleared.
86
0x56
call-reject
Call rejected.
21
0x15
cell-rate-unavail
Cell rate not available.
37
0x25
channel-unacceptable
Channel unacceptable.
6
0x6
chantype-not-implement
Channel type not implemented.
66
0x42
cid-in-use
Call ID in use.
84
0x54
codec-incompatible
Codec incompatible.
171
0xab
cug-incalls-bar
Closed user group (CUG) incoming calls barred.
55
0x37
cug-outcalls-bar
CUG outgoing calls barred.
53
0x35
dest-incompatible
Destination incompatible.
88
0x58
dest-out-of-order
Destination out of order.
27
0x1b
dest-unroutable
No route to destination.
3
0x3
dsp-error
Digital signal processor (DSP) error.
172
0xac
dtl-trans-not-node-id
Designated transit list (DTL) transit not my node ID.
160
0xa0
facility-not-implemented
Facility not implemented.
69
0x45
facility-not-subscribed
Facility not subscribed.
50
0x32
facility-reject
Facility rejected.
29
0x1d
glare
Glare.
15
0xf
glaring-switch-pri
Glaring switch PRI.
180
0xb4
htspm-oos
Holst Telephony Service Provider Module (HTSPM) out of service.
129
0x81
ie-missing
Mandatory information element missing.
96
0x60
ie-not-implemented
Information element not implemented.
99
0x63
info-class-inconsistent
Inconsistency in information and class.
62
0x3e
interworking
Interworking.
127
0x7f
invalid-call-ref
Invalid call reference value.
81
0x51
invalid-ie
Invalid information element contents.
100
0x64
invalid-msg
Invalid message.
95
0x5f
invalid-number
Invalid number.
28
0x1c
invalid-transit-net
Invalid transit network.
91
0x5b
misdialled-trunk-prefix
Misdialed trunk prefix.
5
0x5
msg-incomp-call-state
Message in incomplete call state.
101
0x65
msg-not-implemented
Message type not implemented.
97
0x61
msgtype-incompatible
Message type not compatible.
98
0x62
net-out-of-order
Network out of order.
38
0x26
next-node-unreachable
Next node unreachable.
128
0x80
no-answer
No user answer.
19
0x13
no-call-suspend
No call suspended.
85
0x55
no-channel
Channel does not exist.
82
0x52
no-circuit
No circuit.
34
0x22
no-cug
Nonexistent CUG.
90
0x5a
no-dsp-channel
No DSP channel.
170
0xaa
no-req-circuit
No requested circuit.
44
0x2c
no-resource
No resource.
47
0x2f
no-response
No user response.
18
0x12
no-voice-resources
No voice resources available.
126
0x7e
non-select-user-clear
Nonselected user clearing.
26
0x1a
normal-call-clear
Normal call clearing.
16
0x10
normal-unspecified
Normal, unspecified.
31
0x1f
not-in-cug
User not in CUG.
87
0x57
number-changeed
Number changed.
22
0x16
param-not-implemented
Nonimplemented parameter passed on.
103
0x67
perm-frame-mode-oos
Permanent frame mode out of service.
39
0x27
perm-frame-mode-oper
Permanent frame mode operational.
40
0x28
precedence-call-block
Precedence call blocked.
46
0x2e
preempt
Preemption.
8
0x8
preempt-reserved
Preemption reserved.
9
0x9
protocol-error
Protocol error.
111
0x6f
qos-unavail
QoS unavailable.
49
0x31
rec-timer-exp
Recovery on timer expiry.
102
0x66
redirect-to-new-destination
Redirect to new destination.
23
0x17
req-vpci-vci-unavail
Requested VPCI VCI not available.
35
0x23
send-infotone
Send information tone.
4
0x4
serv-not-implemented
Service not implemented.
79
0x4f
serv/opt-unavail-unspecified
Service or option not available, unspecified.
63
0x3f
stat-enquiry-resp
Response to status enquiry.
30
0x1e
subscriber-absent
Subscriber absent.
20
0x14
switch-congestion
Switch congestion.
42
0x2a
temp-fail
Temporary failure.
41
0x29
transit-net-unroutable
No route to transit network.
2
0x2
unassigned-number
Unassigned number.
1
0x1
unknown-param-msg-discard
Unrecognized parameter message discarded.
110
0x6e
unsupported-aal-parms
ATM adaptation layer (AAL) parameters not supported.
93
0x5d
user-busy
User busy.
17
0x11
vpci-vci-assign-fail
Virtual path connection identifier virtual channel identifier (VPCI VCI) assignment failure.
36
0x24
vpci-vci-unavail
No VPCI VCI available.
45
0x2d
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-busyThe 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 28The 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 0x32voice 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.
Related Commands
Command
Description
clear voice statistics
Clears voice statistics, resetting the statistics collection.
show voice statistics iec
Displays iec statistics
show voice statistics interval-tag
Displays interval options available for IEC statistics
voice statistics type iec
Enables collection of IEC statistics
voice local-bypass
To configure local calls to bypass the digital signal processor (DSP), use the voice local-bypass command inglobal configuration mode. To direct local calls through the DSP, use the no form of this command.
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-bypasscommand 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.
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.
Command History
Cisco IOS Release
Cisco Products
Modification
12.4(22)YB
Cisco Unified CME 7.1
This command was introduced.
12.4(24)T
Cisco Unified CME 7.1
This command was integrated into Cisco IOS Release 12.4(24)T.
Voice-mlpp configuration mode is used for the gateway globally.
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.
Command Default
The default voice-mail code is 0 (zero) for one-digit codes; 00 (two zeros) for two-digit codes.
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)# exitRelated Commands
Command
Description
digit
Designates the number of digits for STC application feature speed-dial codes.
prefix (stcapp-fsd)
Designates a prefix to precede the dialing of an STC application feature speed-dial code.
redial
Designates an STC application feature speed-dial code to dial again the last number that was dialed.
show running-config
Displays current nondefault configuration settings.
show stcapp feature codes
Displays configured and default STC application feature codes.
speed dial
Designates a range of STC application feature speed-dial codes.
stcapp feature speed-dial
Enters STC application feature speed-dial configuration mode to set feature speed-dial codes.
voice pcm capture
To allocate the number of Pulse Code Modulation (PCM) capture buffers, to set up or change the destination URL for captured data, to enable PCM capture on-demand, and to change the PCM capture trigger string by the user, use the voice pcm 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 number of buffers to 0, use the no form of this command.
voice pcm capture { buffer number | destination url | on-demand-trigger | user-trigger-string start-string stop-string stream bitmap duration call-duration }
no voice pcm capture { buffer number | destination url | on-demand-trigger | user-trigger-string }
Syntax Description
buffer number
Allocates the number of PCM capture buffers. The range is from 0 to 200000. The default is 0.
destination url
Specifies the destination URL for storing captured data.
on-demand-trigger
(Optional) Configures PCM capture user trigger on-demand.
user-trigger-string start-string stop-string stream bitmap duration call-duration
(Optional) Configures PCM user trigger string.
- start-string—Start string up to 15 characters.
- stop-string—Stop string up to 15 characters.
- stream—Configures the PCM capture stream bitmap.
- bitmap—PCM stream bitmap in hexadecimal. The range is from 1 to FFFFFFF. The default is 7.
- duration—Configures the duration for PCM capture.
- call-duration—Duration of call. The range is from 0 to 255. The default is 0.
Usage Guidelines
If you want to change the number of an existing nonzero buffer, you must first reset it to 0 and then change it from 0 to the new number.
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 this 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 pcm 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.
Examples
The following example shows how to configure the number of PCM capture buffers:
Router> enable Router# configure terminal Router(config)# voice pcm capture buffer 200The following example shows how to configure the destination URL for storing captured data:
Router> enable Router# configure terminal Router(config)# voice pcm capture destination tftp://10.0.1.10/acphan/The following example shows how to configure user trigger PCM capture:
Router> enable Router# configure terminal Router(config)# voice pcm capture on-demand-triggerThe following example shows how to change the default user trigger PCM capture start and stop string, stream, and call duration:
Router> enable Router# configure terminal Router(config)# voice pcm capture #132 #543 stream ff duration 230voiceport
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
port-number
Analog foreign exchange station (FXS) voice port number. Range: 2/0 to 2/23.
dial dial-string
String of up to 16 characters that can be dialed on a telephone keypad. Valid characters are 0 through 9, A through D, an * (asterisk) and # (pound sign). The voice gateway sends this string to the call-control system when the analog phone goes off hook.
digit dtmf-digits
(Optional) String of up to 16 characters that can be dialed on a telephone keypad. Valid characters are 0 through 9, A through D, an * (asterisk), # (pound sign), and comma (,). The voice gateway sends this string to the call-control system after the wait-msecs expires. Each comma represents a one second wait.
wait-connect wait-msecs
(Optional) Number of milliseconds that the voice gateway waits after voice cut-through before out-pulsing the DTMF digits. Range: 0 to 30000, in multiples of 50. Default: 50. If 0, DTMF digits are sent automatically by voice gateway after call is connected.
interval inter-digit-msecs
(Optional) Number of milliseconds between the DTMF digits. Range: 50 to 500, in multiples of 50. Default: 50.
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.
Router(config)# sccp plar Router(config-sccp-plar)# voiceport 2/0 dial 3660 digit 1234 wait-connect 500 interval 200 Router(config-sccp-plar)# voiceport 2/1 dial 3264 digit 678,,,9*0,,#123 interval 100 Router(config-sccp-plar)# voiceport 2/3 dial 3478 digit 34567 wait-connect 500voice-port
Cisco 1750 and Cisco 1751
voice-port slot-number/ port
Cisco 2600 series, Cisco 3600 Series, and Cisco 7200 Series
voice-port { slot-number/ subunit-number/ port | slot/ port: ds0-group-no }
Cisco 2600 and Cisco 3600 Series with a High-Density Analog Network Module (NM-HDA)
slot-number/ subunit-number/ portvoice-port
Cisco AS5300
voice-port controller-number :D
Cisco 1750 and Cisco 1751
Syntax Description
slot-number
Number of the slot in the router in which the voice interface card (VIC) is installed. Valid entries are from 0 to 2, depending on the slot in which it has been installed.
port
Voice port number. Valid entries are 0 and 1.
slot-number
Number of the slot in the router in which the VIC is installed. Valid entries are from 0 to 3, depending on the slot in which it has been installed.
subunit-number
Subunit on the VIC in which the voice port is located. Valid entries are 0 or 1.
port
Voice port number. Valid entries are 0 and 1.
slot
The router location in which the voice port adapter is installed. Valid entries are from 0 to 3.
port:
Indicates the voice interface card location. Valid entries are 0 and 3.
ds0-group-no
Indicates the defined DS0 group number. Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card.
Command History
Release
Modification
11.3(1)T
This command was introduced.
11.3(3)T
This command was implemented on the Cisco 2600 series.
12.0(3)T
This command was implemented on the Cisco AS5300.
12.0(7)T
This command was implemented on the Cisco AS5800, Cisco 7200 series, and Cisco 1750. Arguments were added for the Cisco 2600 series and Cisco 3600 series.
12.2(8)T
This command was implemented on Cisco 1751 and Cisco 1760. This command was modified to accommodate the additional ports of the NM-HDA on the Cisco 2600 series, Cisco 3640, and Cisco 3660.
12.2(2)XN
Support for enhanced MGCP voice gateway interoperability was added to Cisco CallManager Version 3.1 for the Cisco 2600 series, Cisco 3600 series, and Cisco VG200.
12.2(11)T
This command was integrated into the Cisco IOS Release 12.2(11)T and Cisco CallManager Version 3.2 and implemented on the Cisco IAD2420 series.
12.2(13)T
This command was integrated into Cisco IOS Release 12.2(13)T. This command does not support the extended echo canceller (EC) feature on the Cisco AS5300 or the Cisco AS5800.
Usage Guidelines
Use the voice-portglobal configuration command to switch to voice-port configuration mode from global configuration mode. Use the exit command to exit voice-port configuration mode and return to global configuration mode.
NoteThis command does not support the extended echo canceller (EC) feature on the Cisco AS5300.
voice-port busyout
To place all voice ports associated with a serial or ATM interface into a busyout state, use the voice-port busyoutcommand in interfaceconfiguration mode. To remove the busyout state on the voice ports associated with this interface, use the no form of this command.
Usage Guidelines
This command busies out all voice ports associated with the interface, except any voice ports configured to busy out under specific conditions using the busyout monitor and busyout seize commands.
Examples
The following example places the voice ports associated with serial interface 1 into busyout state:
interface serial 1 voice-port busyoutThe following example places the voice ports associated with ATM interface 0 into busyout state:
interface atm 0 voice-port busyoutvoice rtp send-recv
To establish a two-way voice path when the Real-Time Transport Protocol (RTP) channel is opened, use the voice rtp send-recv command inglobal configuration mode. To reset to the default, use the no form of this command.
Command Default
The voice path is cut-through in only the backward direction when the RTP channel is opened.
Command History
Release
Modification
12.1(5)T
This command was introduced on Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco 7500 series, Cisco AS5300, Cisco AS5800, and Cisco MC3810 platforms.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into the Cisco IOS Release 12.2(11)T.
voice-service dsp-reservation
To specify the percentage of DSP resources that are reserved strictly for VOIP on the voice card, use the voice-service dsp-reservationcommand in voice-card configuration. To reset the percentage of DSP resources, use the no form of this command.
Usage Guidelines
Use this command to reserve a percentage of the voice card for voice services. The remaining DSP resources will be used for video services. A reservation of 100% specified that all DSP resources will be used for voice services.
NoteYou can configure a percentage less than 100% only when there is a video license and the appropriate PVDM# modules are installed.
Tip
DSP can become fragmented when you change the percentage of DSP resources reserved for voice services when there are TDM voice or DSP farm profiles configured. To ensure the best system performance, reload the router when you change the voice-service-dsp-reservation.
voice service
To enter voice-service configuration mode and to specify a voice-encapsulation type, use the voice service command in global configuration mode..
Command History
Release
Modification
12.1(1)XA
This command was introduced on the Cisco MC3810.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.1(3)T
This command was integrated into Cisco IOS Release 12.1(3)T for VoIP on the Cisco 2600 series and the Cisco 3600 series.
12.1(3)XI
This command was implemented on the Cisco AS5300.
12.1(5)T
This command was integrated into Cisco IOS Release 12.1(5)T.
12.1(5)XM
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(2)T
This command was integrated into Cisco IOS Release 12.2(2)T and implemented on the Cisco 7200 series.
12.2(11)T
This command was implemented on the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
voice source-group
To define a source IP group for voice calls, use the voice source-group command in global configuration mode. To delete the source IP group, use the no form of this command.
Usage Guidelines
Use the voice source-group command to assign a name to a set of source IP group characteristics. The terminating gateway uses these characteristics to identify and translate the incoming VoIP call.
Carrier IDs and trunk group labels must not have the same names.
Do not mix carrier IDs and trunk group labels within a source IP group.
A terminating gateway can be configured with carrier ID source IP groups and trunk-group-label source IP groups. Th e name of the source IP group must be unique to the gateway.
Examples
The following example initiates source IP group "utah2" for VoIP calls:
Router(config)# voice source-group utah2Related Commands
Command
Description
access-list
Defines a list of source groups for identifying incoming calls.
carrier-id (voice source group)
Specifies the carrier handling a VoIP call.
description (voice source group)
Assigns a disconnect cause to a source IP group.
h323zone-id (voice source group)
Assigns a zone ID to an incoming H.323 call.
translation-profile (source group)
Assigns a translation profile to a source IP group.
trunk-group-label (voice source group)
Specifies the trunk handling a VoIP call.
voice statistics accounting method
To enable voice accounting statistics to be collected for a specific accounting method list and to specify the pass criteria for call legs, use the voice statistics accounting methodcommand in global configuration mode. To disable the collection of statistics for the accounting method, use the no form of this command.
voice statistics accounting method method-list-name pass { start-interim-stop | start-stop | stop-only }
no voice statistics accounting method method-list-name pass { start-interim-stop | start-stop | stop-only }
Syntax Description
method-list-name
Name of the accounting method list. The method-list-name argument is the same as that configured using the method command in gateway accounting AAA configuration mode.
pass
The pass criteria for call legs (PSTN or IP) and call directions (inbound or outbound) that is used by the method list.
Note The definition of pass implies that all start, stop, or interim messages are acknowledged by the designated servers. The definition of failure implies that any start, stop, or interim message is rejected or is timed out by the designated servers.
start-interim-stop
All start, interim, and stop pass criteria records are counted.
start-stop
All start and stop pass criteria records are counted.
stop-only
Only stop pass criteria records are counted.
Examples
The following example shows that h323 is specified as the method list and that the pass criterion is stop-only:
Router(config)# voice statistics accounting method h323 pass stop-onlyRelated Commands
Command
Description
method
Specifies the AAA method list name to be used.
show voice statistics csr interval accounting
Displays statistical information by configured intervals for accounting statistics.
show voice statistics csr since-reset accounting
Displays all accounting CSRs since the last reset.
voice statistics display-format separator
Specifies the format for CSR display.
voice statistics field-params
Specifies MCD, lost-packet, packet-latency, and packet-jitter parameters.
voice statistics max-storage-duration
Specifies the maximum time for which CSRs are stored in system memory.
voice statistics push
Specifies an FTP or syslog server for downloading CSRs, the maximum file size, and the maximum message size.
voice statistics time-range
Specifies the time range to collect CSRs.
voice statistics type
Enables the collection of accounting and signaling CSRs.
voice statistics display-format separator
To configure the display format of the statistics on the gateway, use the voice statistics display-format separatorcommand in global configuration mode. To return the display format of the statistics to the default value, use the no form of this command.
voice statistics display-format separator { space | tab | new-line | char char }
no voice statistics display-format separator { space | tab | new-line | char char }
Syntax Description
separator
Type of separator used in the displayed format.
space
A space is used for the formatting between each statistic in the displayed output.
tab
A tab is used for the formatting between each statistic in the displayed output.
new-line
A new line is used for the formatting between each statistic in the displayed output.
char char
A character is used for the formatting between each statistic in the displayed output. The char argument is a visible ASCII character used for the formatting between each statistic in the displayed output.
Examples
The following example shows that a space is specified as the display separator:
Router(config)# voice statistics display-format separator spaceRelated Commands
Command
Description
voice statistics accounting method
Enables the accounting method and the pass and fail criteria.
voice statistics field-params
Specifies MCD, lost-packet, packet-latency, and packet-jitter parameters.
voice statistics max-storage-duration
Specifies the maximum time for which CSRs are stored in system memory.
voice statistics push
Specifies an FTP or syslog server for downloading CSRs, the maximum file size, and the maximum message size.
voice statistics time-range
Specifies the time range to collect CSRs.
voice statistics type
Enables the collection of accounting and signaling CSRs.
voice statistics field-params
To configure the parameters of call statistics fields on the gateway, use the voice statistics field-params command in global configuration mode. To return the call statistics parameters to the default values, use the no form of this command.
voice statistics field-params { mcd value | lost-packet value | packet-latency value | packet-jitter value }
no voice statistics field-params { mcd value | lost-packet value | packet-latency value | packet-jitter value }
Syntax Description
mcd
Minimum call duration. The value argument is an integer that represents the number of milliseconds. Valid values are from 0 to 30. The default is 2.
lost-packet
Lost voice packet threshold. The value argument is an integer that represents milliseconds. Valid values are from 0 to 65535. The default is 1000.
packet-latency
Voice packet latency threshold. The value argument is an integer that represents milliseconds. Valid values are from 0 to 500. The default is 250.
packet-jitter
Voice packet jitter threshold. The value argument is an integer that represents milliseconds. Valid values are from 0 to 1000. The default is 250.
Command Default
MCD is 2 milliseconds. Lost packet threshold is 1000 milliseconds. Packet latency threshold is 250 milliseconds. Packet jitter threshold is 250 milliseconds.
Examples
The following example configures a minimum call duration of 5 milliseconds:
Router(config)# voice statistics field-params mcd 5The following example configures a lost packet threshold of 250 milliseconds:
Router(config)# voice statistics field-params lost-packet 250The following example configures a packet-latency threshold of 300 milliseconds:
Router(config)# voice statistics field-params packet-latency 300The following example configures a packet-jitter threshold of 245 milliseconds:
Router(config)# voice statistics field-params packet-jitter 245Related Commands
Command
Description
voice statistics accounting method
Enables the accounting method and the pass and fail criteria.
voice statistics display-format separator
Specifies the format for CSR display.
voice statistics max-storage-duration
Specifies the maximum time for which CSRs are stored in system memory.
voice statistics push
Specifies an FTP or syslog server for downloading CSRs, the maximum file size, and the maximum message size.
voice statistics time-range
Specifies the time range to collect CSRs.
voice statistics type
Enables the collection of accounting and signaling CSRs.
voice statistics max-storage-duration
To configure the maximum amount of time for which collected statistics are stored in the system memory of the gateway, use the voice statistics max-storage-durationcommand in global configuration mode. To remove the configured maximum storage duration, use the no form of this command.
voice statistics max-storage-duration { dayvalue | hour value | minutevalue }
no voice statistics max-storage-duration { dayvalue | hour value | minutevalue }
Syntax Description
day
Number of days for which call statistics data are to be stored. The value argument has a valid range from 0 to 365.
hour
Number of hours for which call statistics data are to be stored. The value argument has a valid range from 0 to 720.
minute
Number of minutes for which call statistics data are to be stored. The value argument has a valid range from 0 to 1440.
Command Default
If no length of time is configured, no memory is allocated for those call statistic records that have stopped after the end of their collection intervals. If no memory is allocated, only active call statistic record buffers are kept in system memory.
Usage Guidelines
The maximum storage duration means the time-to-exist duration of the call statistic records on the gateway.
The values entered using this command also apply to the collection of VoIP internal error codes (IECs).
Examples
The following example shows that the maximum storage duration for the collection of voice call statistics has been set for 60 minutes:
Router(config)# voice statistics max-storage-duration minute 60Related Commands
Command
Description
voice statistics accounting method
Enables the accounting method and the pass and fail criteria.
voice statistics display-format separator
Specifies the format for CSR display.
voice statistics field-params
Specifies MCD, lost-packet, packet-latency, and packet-jitter parameters.
voice statistics push
Specifies an FTP or syslog server for downloading CSRs, the maximum file size, and the maximum message size.
voice statistics time-range
Specifies the time range to collect CSRs.
voice statistics type
Enables the collection of accounting and signaling CSRs.
voice statistics push
To configure the method for pushing signaling statistics, VoIP AAA accounting statistics, or Cisco internal error codes (IECs) to an FTP or syslog server, use the voice statistics push command in global configuration mode. To disable the configured push method, use the no form of this command.
{ voice statistics push ftp url ftp-url [ max-file-size value ] | syslog [ max-msg-size value ] }
{ no voice statistics push ftp url ftp-url [ max-file-size value ] | syslog [ max-msg-size value ] }
Syntax Description
ftp url
URL of the FTP server to which voice statistics are to be pushed. The syntax of the ftp-url argument follows: ftp://user:password@host:port//directory1/directory2
max-file-size
(Optional) Maximum size of a voice statistics file to be pushed to an FTP server, in bytes. The valid range of the value argument is from 1024 to 4294967296. The default value is 400000000 (4 GB).
syslog
Voice statistics are pushed to a syslog server.
max-msg-size
(Optional) Maximum size of a voice statistics file to be pushed to a syslog server, in bytes. The valid range of the value argument is from 1024 to 4294967296. The default value is 400000000 (4 GB).
Usage Guidelines
The gateway configuration should be consistent with the configuration on the FTP or syslog servers. This command may also be used to push Cisco VoIP internal error codes (IECs) to either an FTP server or a syslog server.
Examples
The following is a configuration example showing a specified FTP server and maximum file size:
Router(config)# voice statistics push ftp url ftp://john:doe@abc:23//directory1/directory2 max-file-size 10000Related Commands
Command
Description
voice statistics accounting method
Enables the accounting method and the pass and fail criteria.
voice statistics display-format separator
Specifies the format for CSR display.
voice statistics field-params
Specifies MCD, lost-packet, packet-latency, and packet-jitter parameters.
voice statistics max-storage-duration
Specifies the maximum time for which CSRs are stored in system memory.
voice statistics time-range
Specifies the time range to collect CSRs.
voice statistics type
Enables the collection of accounting and signaling CSRs.
voice statistics time-range
To specify a time range to collect statistics from the gateway on a periodic basis, since the last reset, or for a specific time duration , use the voice statistics time-rangecommand in global configuration mode. To disable the time-range settings, use the no form of this command.
Statistics Collection on a Periodic Basis
voice statistics time-rangeperiodicintervalstarthh:mm { days-of-week { Monday | Tuesday | Wednesday | Thursday | Friday | Saturday | Sunday | daily | weekday | weekend } } [ endhh:mm { days-of-week | Monday | Tuesday | Wednesday | Thursday | Friday | Saturday | Sunday | daily | weekday | weekend } ]
no voice statistics time-rangeperiodicintervalstarthh:mm { days-of-week { Monday | Tuesday | Wednesday | Thursday | Friday | Saturday | Sunday | daily | weekday | weekend } } [ endhh:mm { days-of-week | Monday | Tuesday | Wednesday | Thursday | Friday | Saturday | Sunday | daily | weekday | weekend } ]
Statistics Collection Since the Last Reset or Reboot of the Gateway
voice statistics time-range since-reset
no voice statistics time-range since-reset
Statistics Collection at a Specific Time Duration
voice statistics time-range specific start hh : mm day month year end hh : mm day month year
no voice statistics time-range specific start hh : mm day month year end hh : mm day month year
Syntax Description
Statistics Collection on a Periodic Basis:
periodic
Call statistics are collected for a configured period.
interval
Specifies the periodic interval during which statistics will be collected. Valid entries for this value are 5minutes, 15minutes, 30minutes, 60minutes, or 1day.
start/end
Specifies the start and ending periods of the statistics collection. If no end time is entered, then the statistics collection continues nonstop. By default, there is no end of the collection period.
hh:mm
Specifies the start and ending times for the periodic statistics collection in hours and minutes. The times entered must be in 24-hour format.
days-of-week
Specifies the start and ending days of the week that call statistics are collected. You can configure a specific day of the week, or one of the following:
- daily--Call statistics are collected daily.
- weekdays--Call statistics are collected on weekdays only.
- weekend--Call statistics are collected on weekends only.
The default value is daily.
Statistics Collection Since the Last Reset or Reboot of the Gateway
since-reset
Call statistics are collected only since a reset or reboot of the gateway.
Note Voice statistics collection on the gateway is reset using the clear voice statistics csrcommand.
Statistics Collection at a Specified Time Duration:
specific
Call statistics are collected for a specific time duration.
start/end
Specifies the start and end times of the statistics collection. The required arguments for both the start and end keywords are as follows:
- hh:mm--Hour and minute. The times entered must be in 24-hour format.
- day--Day of the month. Valid values are from 1 to 31.
- month--Month for the statistics collection to start. Enter the month name, for example, January, or February. The default is the current month.
- year--Year. Valid values are from 1993 to 2035. The default is the current year.
No statistics are collected by default.
Usage Guidelines
There should be only one specific or periodic configuration at any one time. If a second specific or periodic configuration is configured, the request is rejected and a warning message displays. If the no form of the command is used during the specific time range, the corresponding collection will stop and FTP or syslog messages will not be sent.
Examples
The following example shows that the time range is periodic and set to collect statistics for a 60-minute period on weekdays only beginning at 12:00 a.m.:
Router(config)# voice statistics time-range periodic 60minutes start 12:00 days-of-week weekdaysThe following example configures the gateway to collect call statistics since the last reset (specified with the clear voice statistics csrcommand) or since the last time the gateway was rebooted:
Router(config)# voice statistics time-range since-resetThe following example configures the gateway to collect statistics from 10:00 a.m. on the first day of January to 12:00 a.m. on the second day of January:
Router(config) # voice statistics time-range specific start 10:00 1 January 2004 end 12:00 2 January 2004Related Commands
Command
Description
clear voice statistics
Clears voice statistics, resetting the statistics collection.
voice statistics accounting method
Enables the accounting method and the pass and fail criteria.
voice statistics display-format separator
Specifies the format for CSR display.
voice statistics field-params
Specifies MCD, lost-packet, packet-latency, and packet-jitter parameters.
voice statistics max-storage-duration
Specifies the maximum time for which CSRs are stored in system memory.
voice statistics push
Specifies an FTP or syslog server for downloading CSRs, the maximum file size, and the maximum message size.
voice statistics type
Enables the collection of accounting and signaling CSRs.
voice statistics type csr
To configure a gateway to collect VoIP AAA accounting statistics or voice signaling statistics, independently or at the same time, use the voice statistics type csrcommand in global configuration mode. To disable the counters, use the no form of this command.
voice statistics type csr [ accounting | signaling ]
no voice statistics type csr [ accounting | signaling ]
Command Default
No accounting or signaling call statistics records (CSRs) are collected on the gateway.
Usage Guidelines
If you do not specify a keyword, both accounting and signaling CSRs are collected. Accounting and signaling CSR collection can be enabled and disabled independently.
Examples
The following example shows that both types of CSRs will be collected:
Router(config)# voice statistics type csrThe following example enables accounting CSRs to be collected:
Router(config)# voice statistics type csr accountingThe following example enables signaling CSRs to be collected:
Router(config)# voice statistics type csr signalingThe following example disables the collection of both signaling and accounting CSRs:
Router(config)# no voice statistics type csrThe following example disables the collection of signaling CSRs only:
Router(config)# no voice statistics type csr signalingRelated Commands
Command
Description
voice statistics accounting method
Enables the accounting method and the pass and fail criteria.
voice statistics display-format separator
Specifies the format for CSR display.
voice statistics field-params
Specifies MCD, lost-packet, packet-latency, and packet-jitter parameters.
voice statistics max-storage-duration
Specifies the maximum time for which CSRs are stored in system memory.
voice statistics push
Specifies an FTP or syslog server for downloading CSRs, the maximum file size, and the maximum message size.
voice statistics time range
Specifies the time range to collect CSRs.
voice statistics type iec
To enable collection of Internal Error Code (IEC) statistics, use the voice statistics type iec command in global configuration mode. To disable IEC statistics collection, use the no form of this command.
Examples
The following example enables IEC statistics collection:
Router(config)# voice statistics type iecRelated Commands
Command
Description
clear voice statistics
Clears voice statistics, resetting the statistics collection.
show voice statistics
Displays voice statistics
show voice statistics interval-tag
Displays interval options available for IEC statistics
voice statistics time-range since-reset
Enables collection of call statistics accumulated since the last resetting of IEC counters
voice translation-profile
To define a translation profile for voice calls, use the voice translation-profile command in global configuration mode. To delete the translation profile, use the no form of this command.
Usage Guidelines
After translation rules are defined, they are grouped into profiles. The profiles collect a set of rules that, taken together, translate the called, calling, and redirected numbers in specific ways. Up to 1000 profiles can be defined. Each profile must have a unique name .
These profiles are referenced by trunk groups, dial peers, source IP groups, voice ports, and interfaces for handling call translations.
Examples
The following example initiates translation profile "westcoast" for voice calls. The profile uses translation rules 1, 2, and 3 for various types of calls.
Router(config)# voice translation-profile westcoast Router(cfg-translation-profile)# translate calling 2 Router(cfg-translation-profile)# translate called 1 Router(cfg-translation-profile)# translate redirect-called 3voice translation-rule
To define a translation rule for voice calls, use the voice translation-rule command in global configuration mode. To delete the translation rule, use the no form of this command.
Usage Guidelines
Use the voice translation-rule command to create the definition of a translation rule. Each definition includes up to 15 rules that include SED-like expressions for processing the call translation. A maximum of 128 translation rules are supported.
These translation rules are grouped into profiles that are referenced by trunk groups, dial peers, source IP groups, voice ports, and interfaces.
voice vad-time
To change the minimum silence detection time for voice activity detection (VAD), use the voice vad-time command in global configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
Thiscommand affects all voice ports on a router or concentrator, but it does not affect calls already in progress.
You can use this command in transparent common-channel signaling (CCS) applications in which you want VAD to activate when the voice channel is idle, but not during active calls. With a longer silence detection delay, VAD reacts to the silence of an idle voice channel, but not to pauses in conversation.
Thiscommand does not affect voice codecs that have ITU-standardized built-in VAD features--for example, G.729B, G.729AB, G.723.1A. The VAD behavior and parameters of these codecs are defined exclusively by the applicable ITU standard.
voice vrf
To configure a voice VRF, use the voice vrfcommand in global configuration mode. To remove the voice VRF configuration, use the no form of this command.
Usage Guidelines
You must create a VRF using the ip vrf vrfname command before you can configure it as a voice VRF.
To ensure there are no active calls on the voice gateway during a VRF change, voices services must be shut down on the voice gateway before you configure or make changes to a voice VRF.
voip-incoming translation-profile
To specify a translation profile for all incoming VoIP calls, use the voip-incoming translation-profile command in global configuration mode. To delete the profile, use the no form of this command.
Usage Guidelines
Use the voip-incoming translation-profile command to globally assign a translation profile for all incoming VoIP calls. The translation profile was previously defined using the voice translation-profile command. The voip-incoming translation-profilecommand does not require additional steps to complete its definition.
If an H.323 call comes in and the call is associated with a source IP group that is defined with a translation profile, the source IP group translation profile overrides the global translation profile.
voip-incoming translation-rule
To set the incoming translation rule for calls that originate from H.323-compatible clients, use the voip-incoming translation-rule command in global configuration mode. To disable the incoming translation rule, use the no form of this command.
voip-incoming translation-rule { calling | called } name-tag
no voip-incoming translation-rule { calling | called } name-tag
Syntax Description
name-tag
Tag number by which the rule set is referenced. This is an arbitrarily chosen number. Range is from 1 to 2147483647. There is no default value.
calling
Automatic number identification (ANI) number or the number of the calling party.
called
Dial Number Information Service (DNIS) number or the number of the called party.
Command History
Release
Modification
12.0(7)XR1
This command was introduced for VoIP on the Cisco AS5300.
12.0(7)XK
This command was implemented for VoIP on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T and implemented for VoIP on the Cisco 1750, Cisco AS5300, Cisco 7200 series, and Cisco 7500 series platforms.
12.1(2)T
This command was implemented for VoIP on Cisco MC3810.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
Usage Guidelines
With this command, all IP-based calls are captured and handled, depending on either the calling number or the called number to the specified tag name.
Examples
The following example identifies the rule set for calls that originate from H.323-compatible clients:
Router(config)# voip-incoming translation-rule called 5Related Commands
Command
Description
numbering-type
Matches one number type for a dial-peer call leg.
rule
Applies a translation rule to a calling party number or a called party number for both incoming and outgoing calls.
show translation-rule
Displays the contents of all the rules that have been configured for a specific translation name.
test translation-rule
Tests the execution of the translation rules on a specific name-tag.
translate
Applies a translation rule to a calling party number or a called party number for incoming calls.
translate-outgoing
Applies a translation rule to a calling party number or a called party number for outgoing calls.
translation-rule
Creates a translation name and enters translation-rule configuration mode.
voip trunk group
To define or modify a VOIP trunk group and to enter trunk group configuration mode, use the voip trunk group command in global configuration mode. To delete the VOIP trunk group, use the no form of this command.
Usage Guidelines
Use the voip trunk group command to define the VOIP trunk and extend serviceability to the trunk. By default, the session protocol of the IP trunk is h323. Up to 1000 trunk groups can be configured on the gateway provided that the gateway has sufficient memory to store the profiles
volume
To set the receiver volume level for a POTS port on a router, use the volume command in dial-peer voice configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
Set the volume command for each POTS port separately. Setting the volume level affects only the port for which it has been set.
NoteOnly the receiver volume is set with this command.
Use the show pots volume command to check the volume status and level.
vxml allow-star-digit
To configure a Voice Extensible Markup Language (VXML) interpreter to allow the star digit for built-in type digits, use the vxml allow-star-digit command in global configuration mode. To disable the configuration, use the no form of this command.
vxml audioerror
To enable throwing an error event when audio playout fails, use the vxml audioerror command in global configuration mode. To return to the default, use the no form of this command.
Command Default
An audio error event, error.badfetch, is not thrown when an audio file cannot be played.
Usage Guidelines
Entering this command causes an audio error event, error.badfetch, to be thrown when an audio file cannot be played, for instance, because the file is in an unsupported format, the src attribute references an invalid URI, or the expr attribute evaluates to an invalid URI.
The vxml audioerror command overrides the vxml version 2.0 command, so that if both commands are entered, the audio error event will be thrown when an audio file cannot be played.
vxml tree memory
To set the maximum memory size for the VoiceXML parser tree, use the vxml tree memory command in global configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
This command limits the memory resources available for parsing VoiceXML documents, preventing large documents from consuming excessive system memory. Increasing the maximum memory size for the VoiceXML tree enables calls to use larger VoiceXML documents. If a VoiceXML document exceeds the limit, the gateway aborts the document execution and the debug vxml error command displays a "vxml malloc fail" error.
NoteIn Cisco IOS Release 12.3(4)T and later releases, less memory is consumed when parsing a VoiceXML document because the document is not stored by the VoiceXML tree.
vxml version 2.0
To enable VoiceXML 2.0 features, use the vxml version 2.0 command in global configuration mode. To return to the default, use the no form of this command.
Command Default
The default VoiceXML behavior is compatible with versions earlier than W3C VoiceXML 2.0 Specification .
Usage Guidelines
This command enables the following VoiceXML features:
- An audio error event, error.badfetch, is not thrown when an audio file cannot be played, for instance, because the file is in an unsupported format, the src attribute references an invalid URI, or the expr attribute evaluates to an invalid URI.
- Support for the beep attribute of the <record> element.
- Blind transfer compliant with W3C VoiceXML 2.0 and not the same as consultation transfer.
- Compatibility with W3C VoiceXML 2.0 Specification.
- A semantic error is generated if an undeclared variable is used. You must declare variables before using them.
