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Table Of Contents
Command Reference: A through M
after-hour exempt (voice register pool)
application (voice register global)
application (voice register pool)
call-forward b2bua all (voice register dn and voice register pool)
call-forward b2bua busy (voice register dn and voice register pool)
call-forward b2bua mailbox (voice register dn and voice register pool)
call-forward b2bua noan (voice register dn and voice register pool)
dialplan-pattern (voice register pool)
dtmf-relay (voice register pool)
external-ring (voice register global)
incoming called-number (voice register pool)
max-pool (voice register global)
max registrations (voice register pool)
Command Reference: A through M
This chapter contains commands to configure and maintain a typical Cisco SIP Survivable Remote Site Telephony (SRST) environment. The commands are presented in alphabetical order. Some commands required for configuring voice may be found in other Cisco IOS command references. Use the command reference master index or search online to find these commands.
For detailed information on how to configure Cisco SIP SRST applications and features, see the Cisco IOS SIP SRST Version 3.4 System Administrator Guide.
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after-hour exempt (voice register pool)
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after-hour exempt (voice register pool)
To specify that for a particular voice register pool no outgoing calls are blocked even though global system call blocking is enabled, use the after-hours exempt command in voice register pool configuration mode. To return to the default, use the no form of this command.
after-hour exempt
no after-hour exempt
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled (global call blocking remains active, as configured).
Command Modes
Voice register pool configuration
Command History
12.4(4)T
Cisco CME 3.4 and Cisco SIP SRST 3.4
This command was introduced.
Usage Guidelines
This command exempts individual Cisco SIP phones and phone extensions from call blocking.
Call blocking on Cisco IP phones is defined in the following way. First, define one or more patterns of outgoing digits by using the after-hours block pattern command in either telephony-service configuration mode for Cisco CME or in call-manager-fallback configuration mode for Cisco SIP SRST. Next, define one or more time periods during which calls that match those patterns are to be blocked are by using the after-hours date or after-hours day command or both. By default, all Cisco IP phones in a Cisco CME or Cisco SIP SRST system are restricted during the specified time if at least one pattern and at least one time period are defined.
A phone extension is exempt as long as the after-hour exempt command is configured in voice register dn or in voice register pool configuration mode.
Note
Examples
The following example exempts blocking of outgoing calls from SIP phone 23:
Router(config)# voice register pool 23Router(config-register-pool)# after-hour exemptThe following example specifies that outgoing calls from extension 5001 under voice register pool 2 are not blocked:
Router(config)# voice register pool 2Router(config-register-pool)# number 5001Router(config-register-pool)# after-hour exemptRelated Commands
alias (voice register pool)
To allow Cisco SIP IP phones to handle inbound PSTN calls to telephone numbers that are unavailable when the main proxy is not available, use the alias command in voice register pool configuration mode. To disable rerouting of unmatched call destination calls, use the no form of this command.
alias tag pattern to target [preference value]
no alias tag
Syntax Description
Defaults
None
Command Modes
Voice register pool configuration
Command History
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
Usage Guidelines
The alias command services calls placed to telephone numbers that are unavailable because the main proxy is not available. The alias command is activated when a Cisco SIP IP phone that has an extension number matching the target number registers.
When a phone with the target number registers, calls that match the number pattern are rerouted to the target number. The target number must be a local phone number to enable rerouting of a range of number patterns. When a Cisco SIP IP phone registers with a target number, an additional VoIP dial peer is created using the target number IP address as a session target and destination pattern as configured with the alias pattern command. For the alias command to work, the VoIP dial peer must be set with a translation rule to translate the called number to the target number. Translation rules can be configured under voice register pool configuration mode.
If other Cisco SIP IP phones register that have specific phone numbers that fall within the alias range or if another static dial peer exists for this pattern, the call is routed using the appropriate dial peer in preference to being rerouted to this alternate alias number (according to normal dial-peer longest-match, preference, and huntstop rules).
Note
Before the alias command is configured, translation rules must be set using the translate-outgoing (voice register pool) command. Translation rules are a general-purpose number-manipulation mechanism that perform operations such as automatically adding telephone area and prefix codes to dialed numbers.Examples
The following example configures calls to numbers in the 5000 to 5099 range that are not otherwise explicitly resolved to a specific extension number to be routed to the phone with extension 5001. Phone calls intended for phones that are not given fallback service can then be redirected to the specified extension number.
Router(config)# voice register poolRouter(config-register-pool)# alias 1 50.. to 5001Related Commands
application (voice register global)
To select the session-level application for all dial peers associated with SIP phones, use the application command in voice register global configuration mode. To disable use of the application, use the no form of this command.
application application-name
no application
Syntax Description
Defaults
Default application on router
Command Modes
Voice register global configuration
Command History
12.4(4)T
Cisco CME 3.4 and Cisco SIP SRST 3.4
This command was introduced.
Usage Guidelines
During Cisco CME or Cisco SIP SRST registration, a dial peer is created for each SIP phone and that dial peer includes the default session application. The application command allows you to change the default application for all dial peers associated with the Cisco SIP IP phones, if desired. The applied application (or TCL IVR script) must support call redirection. Use the show call application voice summary command to display a list of applications.
The application command in voice register pool configuration mode takes precedence over this command in voice register global configuration mode.
Note
Examples
The following example shows how to set the Tcl IVR application globally for all SIP phones:
Router(config)# voice register globalRouter(config-register-global)# mode cmeRouter(config-register-global)# application sipapp2Related Commands
application (voice register pool)
To select the session-level application for the dial peer associated an individual SIP phone in a Cisco CME environment or for a group of phones in a Cisco SIP SRST environment, use the application command in voice register pool configuration mode. To disable use of the application, use the no form of this command.
application application-name
no application
Syntax Description
Defaults
Default application on router
Command Modes
Voice register pool configuration
Command History
Usage Guidelines
During Cisco CME or Cisco SIP SRST registration, a dial peer is created for each SIP phone and that dial peer includes the default session application. The application command allows you to change the default application for all dial peers associated with the Cisco SIP IP phones, if desired. The applied application (or TCL IVR script) must support call redirection. Use the show call application voice summary command to display a list of applications.
The application command in voice register pool configuration mode takes precedence over this command in voice register global configuration mode.
Note
Examples
The following example shows how to set the IVR application for the SIP phone specified by the voice register pool command:
Router(config)# voice register pool 1Router(config-register-pool) application sipapp2The following partial sample output from the show running-config command shows that voice register pool 1 has been set up to use the SIP.app application:
voice register pool 1id network 172.16.0.0 mask 255.255.0.0application SIP.appvoice-class codec 1Related Commands
b2bua
To configure a dial peer associated with an individual SIP phone in a Cisco CME environment or a group of phones in a Cisco SIP SRST environment to point to Cisco Unity Express, use the b2bua command in dial-peer configuration mode. To disable B2BUA call flow on the dial peer, use the no form of this command.
b2bua
no b2bua
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Dial-peer configuration
Command History
12.4(4)T
Cisco CME 3.4 and Cisco SIP SRST 3.4
This command was introduced.
Usage Guidelines
Use the b2bua command to set the Cisco CME source address as the 302 redirect contact address for all calls forwarded to Cisco Unity Express.
Note
Examples
The following example shows b2bua included in the configuration for voice dial peer 1:
dial-peer voice 1 voipdestination-pattern 4...session target ipv4:10.5.49.80session protocol sipv2dtmf-relay sip-notifyb2buaRelated Commands
call-forward b2bua all (voice register dn and voice register pool)
To enable call forwarding for a SIP back-to-back user agent (B2BUA) so that all incoming calls are forwarded to another extension, use the call-forward b2bua all command in voice register dn or voice register pool configuration mode. To disable call forwarding, use the no form of this command.
call-forward b2bua all directory-number
no call-forward b2bua all
Syntax Description
directory-number
Telephone number to which calls are forwarded. Represents a fully qualified E.164 number. Maximum length of the telephone number is 32.
Defaults
Disabled (no incoming call forwarding to another extension).
Command Modes
Voice register dn configuration
Voice register pool configurationCommand History
12.4(4)T
Cisco CME 3.4 and Cisco SIP SRST 3.4
This command was introduced.
Usage Guidelines
You can apply call forward to an individual extension (voice register dn) or to the phone on which the extension appears (voice register pool). Use this command in voice register pool configuration mode to enable call forwarding for all extensions on a Cisco SIP IP phone. Use this command in voice register dn configuration mode to enable call forwarding for an individual extension.
If information is configured in both voice register dn and voice register pool mode, the information under voice register dn mode takes precedence.
It is recommended that you do not use this command with hunt groups. If the command is used, consider removing the phone from any assigned hunt groups, unless you want to forward calls to all phones in the hunt group.
The call-forward b2bua all command takes precedence over the call-forward b2bua busy and call-forward b2bua noan commands.
Note
Examples
The following example shows how to forward all incoming calls to extension 5001 on directory number 4, to extension 5005.
Router(config)# voice register dn 4Router(config-register-dn)# number 5001Router(config-register-dn)# call-forward b2bua all 5005The following example forwards to extension 5005 all incoming calls to extension 5001 on pool number 4.
Router(config)# voice register pool 4Router(config-register-pool)# number 5001Router(config-register-pool)# call-forward b2bua all 5005Related Commands
call-forward b2bua busy (voice register dn and voice register pool)
To enable call forwarding for a SIP back-to-back user agent (B2BUA) so that incoming calls to a busy extension are forwarded to another extension, use the call-forward b2bua busy command in voice register dn or voice register pool configuration mode. To disable call forwarding, use the no form of this command.
call-forward b2bua busy directory-number
no call-forward b2bua busy
Syntax Description
directory-number
Telephone number to which calls are forwarded. Represents a fully qualified E.164 number. Maximum length of the telephone number is 32.
Defaults
Disabled (no incoming calls to a busy extension are forwarded).
Command Modes
Voice register dn configuration
Voice register pool configurationCommand History
12.4(4)T
Cisco CME 3.4 and Cisco SIP SRST 3.4
This command was introduced.
Usage Guidelines
The call-forward b2bua busy response is triggered when a call is sent to a Cisco SIP IP phone using a VoIP dial peer and a busy response is received back from the phone. This command functions only with phones that are registered to a Cisco SIP SRST or Cisco CME router.
You can apply call forward to an individual extension (voice register dn) or to the SIP phone on which the extension appears (voice register pool). Use this command in voice register pool configuration mode to enable call forwarding for all extensions on a SIP phone. Use this command in voice register dn configuration mode to enable call forwarding for a specific extension. If information is configured in both voice register dn and voice register pool mode, the information under voice register dn takes precedence.
It is recommended that you do not use this command with hunt groups. If the command is used, consider removing the phone from any assigned hunt groups, unless you want to forward calls to all phones in the hunt group.
The call-forward b2bua all command takes precedence over the call-forward b2bua busy and call-forward b2bua noan commands.
Note
Cisco CME
Call forward busy can also get invoked if a number is unreachable but the call forward b2bua unreachable command is not configured.
Examples
The following example forwards calls to extension 5005 when extension 5001 in pool number 4 is busy.
Router(config)# voice register pool 4Router(config-register-pool)# number 5001Router(config-register-pool)# call-forward b2bua busy 5005Related Commands
call-forward b2bua mailbox (voice register dn and voice register pool)
To control the specific voice-mail box selected in a voice-mail system at the end of a call forwarding exchange, use the call-forward b2bua mailbox command in voice register dn or voice register pool configuration mode. To disable call forwarding, use the no form of this command.
call-forward b2bua mailbox directory-number
no call-forward b2bua mailbox
Syntax Description
Defaults
Disabled (no voice-mail box is selected for call forwarding).
Command Modes
Voice register dn configuration
Voice register pool configurationCommand History
12.4(4)T
Cisco CME 3.4 and Cisco SIP SRST 3.4
This command was introduced.
Usage Guidelines
Use this command to denote the voice-mail box to use at the end of a chain of call forwards to busy or no answer destinations. It can be used to forward calls to a voice-mail box that has a different number than the forwarding extension. A sample of this would be in the case of a shared voice-mail box, for instance one between a manager and her assistant. This command functions only with phones that are registered to a Cisco CME or Cisco SIP SRST router.
If information is configured in both voice register dn and voice register pool mode, the information under voice register dn takes precedence.
It is recommended that you do not use the call-forward b2bua mailbox command with hunt groups. If the command is used, consider removing the phone from any assigned hunt groups, unless you want to forward calls to all phones in the hunt group.
This command is used in conjunction with the call-forward b2bua all, call-forward b2bua busy, and call-forward b2bua noan commands.
Note
Examples
The following example forwards calls to extension 5005 if an incoming call is forwarded to extension 5001 on pool number 4 and extension 5001 is busy or does not answer.
Router(config)# voice register pool 4Router(config-register-pool)# number 5001Router(config-register-pool)# call-forward b2bua mailbox 5005Related Commands
call-forward b2bua noan (voice register dn and voice register pool)
To enable call forwarding for a SIP back-to-back user agent (B2BUA) so that incoming calls to an extension that does not answer after a configured amount of time are forwarded to another extension, use the call-forward b2bua noan command in voice register dn or voice register pool configuration mode. To disable call forwarding, use the no form of this command.
call-forward b2bua noan directory-number timeout seconds
no call-forward b2bua noan
Syntax Description
Defaults
Disabled (no incoming calls to an extension that does not answer are forwarded).
Command Modes
Voice register dn configuration
Voice register pool configurationCommand History
12.4(4)T
Cisco CME 3.4 and Cisco SIP SRST 3.4
This command was introduced.
Usage Guidelines
This command functions only with phones that are registered to a Cisco SIP SRST or Cisco CME router. You can apply call forward to an individual extension (voice register dn) or to the SIP phone on which the extension appears (voice register pool). Use this command in voice register pool configuration mode to enable call forwarding for all extensions on a SIP phone. Use this command in voice register dn configuration mode to enable call forwarding for a specific extension.
If information is configured in both voice register dn and voice register pool mode, the information under voice register dn takes precedence.
It is recommended that you do not use this command with hunt groups. If the command is used, consider removing the phone from any assigned hunt groups, unless you want to forward calls to all phones in the hunt group.
The call-forward b2bua all command takes precedence over the call-forward b2bua busy and call-forward b2bua noan commands.
Note
Examples
The following example forwards calls to extension 5005 when extension 5001 on pool number 4 is unanswered. The timeout before the call is forwarded to extension 5005 is 10 seconds.
Router(config)# voice register pool 4Router(config-register-pool)# number 5001Router(config-register-pool)# call-forward b2bua noan 5005 timeout 10Related Commands
codec (voice register pool)
To specify the codec supported by a single Cisco SIP phone or a VoIP dial peer in a Cisco SIP SRST or a Cisco CME environment, use the codec command in voice register pool configuration mode. To disable a specified codec, use the no form of this command.
codec codec-type [bytes]
no codec
Syntax Description
Defaults
g729r8
Command Modes
Voice register pool configuration
Command History
12.4(4)T
Cisco CME 3.4 and Cisco SIP SRST 3.4
This command was introduced.
Usage Guidelines
This command sets the codec configuration for an individual phone and overrides any previously configured codec selection set with the voice-class codec command.
Note
Examples
The following example shows codec complexity set to g729r8 for a Cisco SIP IP phone in pool 1:
Router(config)# voice register pool 1Router(config-register-pool)# codec g729r8Related Commands
cor (voice register pool)
To configure a class of restriction (COR) on the VoIP dial peers associated with directory numbers, use the cor command in voice register pool configuration mode. To disable a COR associated with directory numbers, use the no form of this command.
cor {incoming | outgoing} cor-list-name {cor-list-number starting-number [- ending-number] | default}
no cor {incoming | outgoing} cor-list-name {cor-list-number starting-number [- ending-number] | default}
Syntax Description
Defaults
None
Command Modes
Voice register pool configuration
Command History
Usage Guidelines
The cor command sets the dial-peer COR parameter for dynamically created VoIP dial peers. A list-based mechanism assigns COR parameters to specific set of number ranges. The COR functionality provides the ability to deny certain call attempts on the basis of the incoming and outgoing class of restrictions provisioned on the dial peers. This functionality provides flexibility in network design, allows users to block calls (for example, calls to 900 numbers), and applies different restrictions to call attempts from different originators.
COR specifies which incoming dial peer can use which outgoing dial peer to make a call. Each dial peer can be provisioned with an incoming and an outgoing COR list.
A default COR is assigned to the directory numbers that do not match any COR list number or number range. During Cisco SIP SRST registration, a dial peer is created and that dial peer includes a default COR value. The cor command allows you to change the automatically selected default.
In dial-peer configuration mode, build your COR list and add members. Then in voice register pool configuration mode, use the cor command to apply the name of the dial-peer COR list.
You can have up to four COR lists for the Cisco SIP SRST configuration, comprised of incoming or outgoing dial peers. The first four COR lists are applied to a range of phone numbers. The phone numbers that do not have a COR list configuration are assigned to the default COR list, providing that a default COR list has been defined.
Note
Examples
The following is sample output from the show running-config command:
...voice register pool 1id mac 0030.94C2.A22Apreference 5cor incoming call91 1 91011translate-outgoing called 1proxy 10.2.161.187 preference 1 monitor probe icmp-pingalias 1 94... to 91011 preference 8voice-class codec 1...dial-peer cor customname 95name 94name 91!dial-peer cor list call91member 91!dial-peer voice 91500 potscorlist incoming call91corlist outgoing call91destination-pattern 91500port 1/0/0...Related Commands
debug voice register errors
To display debug information on voice register module errors during registration in a Cisco CME or Cisco SIP SRST environment, use the debug voice register errors command in privileged EXEC mode. To disable debugging, use the no form of the command.
debug voice register errors
no debug voice register errors
Syntax Description
This command has no arguments or keywords
Defaults
Disabled
Command Modes
Privileged EXEC mode
Command History
Usage Guidelines
Registration errors include failure to match pools or any internal errors that happen during registration.
Examples
Cisco SIP SRST
The following is sample output from this command:
Router# debug voice register errors*Apr 22 11:52:54.523 PDT: VOICE_REG_POOL: Contact doesn't match any pools*Apr 22 11:52:54.539 PDT: VOICE_REG_POOL: Register request for (33015) from (10.2.152.39)*Apr 22 11:52:54.539 PDT: VOICE_REG_POOL: Contact doesn't match any pools.*Apr 22 11:52:54.559 PDT: VOICE_REG_POOL: Register request for (33017) from (10.2.152.39)*Apr 22 11:53:04.559 PDT: VOICE_REG_POOL: Maximum registration threshold for pool(3) hit
If there are no voice register pools configured for a particular registration request, the message "Contact doesn't match any pools" is displayed.
If the max registrations command is configured, when registration requests reach the maximum limit, the "Maximum registration threshold for pool(x) hit" message is displayed for the particular pool.
Table 1 describes the significant fields shown in the display.
Related Commands
debug voice register events
Displays debug information on voice register module events during SIP phone registrations in a Cisco CME or Cisco SIP SRST environment.
debug voice register events
To display debug information on voice register module events during SIP phone registrations in a Cisco CME or Cisco SIP SRST environment, use the debug voice register events command in privileged EXEC mode. To disable debugging, use the no form of this command.
debug voice register errors
no debug voice register errors
Syntax Description
This command has no arguments or keywords
Defaults
Disabled
Command Modes
Privileged EXEC mode
Command History
Usage Guidelines
Using the debug voice register events command should suffice to view registration activity.
Registration activity includes matching of pools, registration creation, and automatic creation of dial
peers. For more details and error conditions, you can use the debug voice register errors command.
Cisco SIP SRST
The following is sample output from this command:
Router# debug voice register eventsApr 22 10:50:21.731 PDT: VOICE_REG_POOL: Contact matches pool 1Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: key(91011) contact(192.168.0.2) add to contact tableApr 22 10:50:21.731 PDT: VOICE_REG_POOL: key(91011) exists in contact tableApr 22 10:50:21.731 PDT: VOICE_REG_POOL: contact(192.168.0.2) exists in contact table, ref updatedApr 22 10:50:21.731 PDT: VOICE_REG_POOL: Created dial-peer entry of type 1Apr 22 10:50:21.731 PDT: VOICE_REG_POOL: Registration successful for 91011, registration id is 257The phone number 91011 registered successfully, and type 1 is reported in the debug, which means that there is a preexisting VoIP dial peer.
Apr 22 10:50:38.119 PDT: VOICE_REG_POOL: Register request for (91021) from (192.168.0.3)Apr 22 10:50:38.119 PDT: VOICE_REG_POOL: Contact matches pool 2Apr 22 10:50:38.123 PDT: VOICE_REG_POOL: key(91021) contact(192.168.0.3) add to contact tableApr 22 10:50:38.123 PDT: VOICE_REG_POOL: key(91021) exists in contact tableApr 22 10:50:38.123 PDT: VOICE_REG_POOL: contact(192.168.0.3) exists in contact table, ref updatedApr 22 10:50:38.123 PDT: VOICE_REG_POOL: Created dial-peer entry of type 1Apr 22 10:50:38.123 PDT: VOICE_REG_POOL: Registration successful for 91021, registration id is 258A dynamic VoIP dial peer has been created for entry 91021. The dial peer can be verified using the show voice register dial-peers and show sip-ua status registrar commands.
Apr 22 10:51:08.971 PDT: VOICE_REG_POOL: Register request for (95021) from (10.2.161.50)Apr 22 10:51:08.971 PDT: VOICE_REG_POOL: Contact matches pool 3Apr 22 10:51:08.971 PDT: VOICE_REG_POOL: key(95021) contact(10.2.161.50) add to contact tableApr 22 10:51:08.971 PDT: VOICE_REG_POOL: No entry for (95021) found in contact tableApr 22 10:51:08.975 PDT: VOICE_REG_POOL: key(95021) contact(10.2.161.50) added to contact tableApr 22 10:51:08.979 PDT: VOICE_REG_POOL: Created dial-peer entry of type 0Apr 22 10:51:08.979 PDT: VOICE_REG_POOL: Registration successful for 95021, registration id is 259Apr 22 10:51:09.019 PDT: VOICE_REG_POOL: Register request for (95012) from (10.2.161.50)Apr 22 10:51:09.019 PDT: VOICE_REG_POOL: Contact matches pool 3Apr 22 10:51:09.019 PDT: VOICE_REG_POOL: key(95012) contact(10.2.161.50) add to contact tableApr 22 10:51:09.019 PDT: VOICE_REG_POOL: No entry for (95012) found in contact tableApr 22 10:51:09.023 PDT: VOICE_REG_POOL: key(95012) contact(10.2.161.50) added to contact tableApr 22 10:51:09.027 PDT: VOICE_REG_POOL: Created dial-peer entry of type 0Apr 22 10:51:09.027 PDT: VOICE_REG_POOL: Registration successful for 95012, registration id is 260Apr 22 10:51:09.071 PDT: VOICE_REG_POOL: Register request for (95011) from (10.2.161.50)Apr 22 10:51:09.071 PDT: VOICE_REG_POOL: Contact matches pool 3Apr 22 10:51:09.071 PDT: VOICE_REG_POOL: key(95011) contact(10.2.161.50) add to contact tableApr 22 10:51:09.071 PDT: VOICE_REG_POOL: No entry for (95011) found in contact tableApr 22 10:51:09.075 PDT: VOICE_REG_POOL: key(95011) contact(10.2.161.50) added to contact tableApr 22 10:51:09.079 PDT: VOICE_REG_POOL: Created dial-peer entry of type 0Apr 22 10:51:09.079 PDT: VOICE_REG_POOL: Registration successful for 95011, registration id is 261Apr 22 10:51:09.123 PDT: VOICE_REG_POOL: Register request for (95500) from (10.2.161.50)Apr 22 10:51:09.123 PDT: VOICE_REG_POOL: Contact matches pool 3Apr 22 10:51:09.123 PDT: VOICE_REG_POOL: key(95500) contact(10.2.161.50) add to contact tableApr 22 10:51:09.123 PDT: VOICE_REG_POOL: No entry for (95500) found in contact tableApr 22 10:51:09.127 PDT: VOICE_REG_POOL: key(95500) contact(10.2.161.50) added to contact tableApr 22 10:51:09.131 PDT: VOICE_REG_POOL: Created dial-peer entry of type 0Apr 22 10:51:09.131 PDT: VOICE_REG_POOL: Registration successful for 95500, registration id is 262*Apr 22 11:52:54.523 PDT: VOICE_REG_POOL: Contact doesn't match any pools*Apr 22 11:52:54.539 PDT: VOICE_REG_POOL: Register request for (33015) from (10.2.152.39)*Apr 22 11:52:54.539 PDT: VOICE_REG_POOL: Contact doesn't match any pools*Apr 22 11:52:54.559 PDT: VOICE_REG_POOL: Register request for (33017) from (10.2.152.39)Table 2 describes the significant fields shown in the display.
Related Commands
dialplan-pattern (voice register pool)
Note
To create a global prefix that can be used to expand the abbreviated extension numbers (automatically obtained from the Cisco IP phone) into fully qualified E.164 telephone numbers, use the dialplan-pattern command in voice register pool configuration mode. To disable a global prefix, use the no form of this command.
dialplan-pattern tag pattern extension-length length [extension-pattern extension-pattern]
no dialplan-pattern tag
Syntax Description
Defaults
Global prefixes are disabled.
Command Modes
Voice register pool configuration
Command History
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
12.4(4)T
This command was removed.
Usage Guidelines
The dialplan-pattern command creates a global prefix that can be used to expand the abbreviated extension numbers into fully qualified E.164 telephone numbers. The extension number should be greater than or equal to the extension length. Otherwise, the extension number cannot be converted into a qualified E.164 number.
The extension-length keyword enables the system to convert a full E.164 telephone number back into an extension number. For example, a company uses extension number range 0100 to 0199 across several sites, with only the extensions 0100 to 0199 present on the local router. An incoming call to 0199 arrives as 4085550199 in its full E.164 format. The dialplan-pattern command translates to the extension number and routes the call to Cisco SIP IP phone extension 0199. The dialplan-pattern command creates an additional VoIP dial peer with a destination pattern of 4085550199 when extension 0199 registers to SIP SRST. Then the IP address of extension 0199 is used as the session target. In order for full E.164 telephone calls to be accepted by Cisco SIP IP phone 0199, a translation rule must be applied in a voice register pool to convert the full E.164 telephone called number into the extension number.
The number of extension-pattern characters must match the extension length (for example, if the extension length is three, the extension pattern can be 8.., 1.., 5...0).
Note
Before configuring the dialplan-pattern command, translation rules must be set using the translate-outgoing command. Translation rules are a general-purpose number-manipulation mechanism that perform operations such as automatically adding telephone area and prefix codes to dialed numbers.Examples
The following example shows how to create dial-plan pattern 1 for extension numbers 0100 to 0199 with the telephone prefix starting with 408555. If the following commands are configured, the routers recognize that the number 4085550100 matches dial-plan pattern 1 and use the extension-length keyword to extract the last four digits of the number 4085550100 and present this as the caller ID for the incoming call.
Router(config)# voice register poolRouter(config-register-pool)# dialplan-pattern 1 40855501.. extension-length 4Related Commands
dtmf-relay (voice register pool)
To specify the list of DTMF relay methods that can be used to relay dual-tone multifrequency (DTMF) audio tones between Session Initiation Protocol (SIP) endpoints, use the dtmf-relay command in voice register pool configuration mode. To send the DTMF audio tones as part of an audio stream, use the no form of this command.
dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify]
no dtmf-relay
Syntax Description
Defaults
DTMF tones are disabled and sent in-band. That is, they remain in the audio stream.
Command Modes
Voice register pool configuration
Command History
12.3(4)T
Cisco SIP SRST 3.0
This command was introduced.
12.4(4)T
Cisco CME 3.4 and Cisco SIP SRST 3.4
This command was added to Cisco CME.
Usage Guidelines
During Cisco SIP SRST or Cisco CME registration, a dial peer is created and that dial peer has a default DTMF relay of in-band. The dtmf-relay command allows you to change the default to a desired value.
DTMF audio tones are generated when you press a button on a touchtone phone. The tones are compressed at one end of the call and when the digits are decompressed at the other end, there is a risk that they can become distorted. DTMF relay reliably transports the DTMF audio tones generated after call establishment out-of-band.
The SIP Notify method sends Notify messages bidirectionally between the originating and terminating gateways for a DTMF event during a call. If multiple DTMF relay mechanisms are enabled on a SIP dial peer and are negotiated successfully, the SIP Notify method takes precedence.
SIP Notify messages are advertised in an Invite message to the remote end only if the dtmf-relay command is set.
For SIP calls, the most appropriate methods to transport DTMF tones are RTP-NTE or SIP-NOTIFY.
Note
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Examples
Cisco CME
The following example enables the RTP-NTE and SIP-NOTIFY mechanisms for DTMF relay for SIP phone 4:
Router(config)# voice register pool 4Router(config-register-pool)# dtmf-relay rtp-nte sip-notifyCisco SIP SRST
The following is sample output from the show running-config command that shows that voice register pool 1 has been set up to send DTMF tones:
voice register pool 1application SIP.appincoming called-number 308voice-class codec 1dtmf-relay rtp-nteRelated Commands
external-ring (voice register global)
To specify the type of ring sound used on Cisco SIP or Cisco SCCP IP phones for external calls, use the external-ring command in voice register global configuration mode. To return to the default ring sound, use the no form of this command.
external-ring {bellcore-dr1 | bellcore-dr2 | bellcore-dr3 | bellcore-dr4 | bellcore-dr5}
no external-ring
Syntax Description
Defaults
The default ring sound is an internal ring pattern.
Command Modes
Voice register global configuration
Command History
12.4(4)T
Cisco CME 3.4 and Cisco SIP SRST 3.4
This command was introduced.
Usage Guidelines
When set, this command defines varying ring tones so that you can discriminate between internal and external calls from Cisco SIP or Cisco SCCP IP phones.
Examples
The following example specifies that Bellcore DR1 be used for external ringing on Cisco SIP IP phones:
Router(config)# voice register globalRouter(config-register-global)# external-ring bellcore-dr1Related Commands
id (voice register pool)
To explicitly identify a locally available individual Cisco SIP IP phone, or when running Cisco SIP SRST, set of Cisco SIP IP phones, use the id command in voice register pool configuration mode. To remove local identification, use the no form of this command.
id {network address mask mask | ip address mask mask | mac address}
no id {network address mask mask | ip address mask mask | mac address}
Syntax Description
Command Default
None
Command Modes
Voice register pool configuration
Command History
Usage Guidelines
Configure the id (voice register pool) command before any other voice register pool commands.
The id command allows explicit identification of an individual Cisco SIP IP phone to support a degree of authentication, which is required to accept registrations, based upon the following:
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When the mac address keyword and argument are used, the IP phone must be in the same subnet as that of the router's LAN interface, such that the phone's MAC address is visible in the router's Address Resolution Protocol (ARP) cache. Once a MAC address is configured for a specific voice register pool, remove the existing MAC address before changing to a new MAC address.
Note
Examples
The following is partial sample output from the show running-config command. The id command identifies the MAC address of a particular Cisco IP phone. The output shows that voice register pool 1 has been set up to accept SIP Register messages from a specific IP phone through the use of the id command.
voice register pool 1id mac 0030.94C2.A22Apreference 5cor incoming call91 1 91011translate-outgoing called 1proxy 10.2.161.187 preference 1 monitor probe icmp-pingalias 1 94... to 91011 preference 8voice-class codec 1Related Commands
incoming called-number (voice register pool)
To apply incoming called-number parameters to dynamically created dial peers, use the incoming called-number command in voice register pool configuration mode. To remove incoming called-number parameters from a dial peer, use the no form of this command.
incoming called-number [number]
no incoming called-number
Syntax Description
Defaults
None
Command Modes
Voice register pool configuration
Command History
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
Usage Guidelines
The id (voice register pool) command must be configured before any other voice register pool commands, including the incoming called-number command. The id command identifies a locally available individual Cisco SIP IP phone or a set of Cisco SIP IP phones.
Examples
The following is partial sample output from the show running-config command that applies the prefix 308 to dynamically created dial peers:
voice register pool 1application SIP.appincoming called-number 308voice-class codec 1Related Commands
max-pool (voice register global)
To set the maximum number of SIP voice register pools that are supported in a Cisco SIP SRST or Cisco CME environment, use the max-pool command in voice register global configuration mode. To reset the maximum number to the default, use the no form of this command.
max-pool max-voice-register-pools
no max-pool
Syntax Description
Defaults
Default is 0 pools.
Command Modes
Voice register global configuration
Command History
12.4(4)T
Cisco CME 3.4 and Cisco SIP SRST 3.4
This command was introduced.
Usage Guidelines
This command limits the number of SIP phones supported by a Cisco CME or Cisco SIP SRST environment. The max-pool command is platform specific and defines the limit for the voice register pool command. The max-dn command similarly limits the number of directory numbers (extensions) in a Cisco CME system.
Note
Examples
The following example sets the maximum number of Cisco SIP IP phones in a Cisco SIP SRST environment to 24:
Router(config)# voice register globalRouter(config-register-global)# max-pool 24Related Commands
max registrations (voice register pool)
Note
To set the maximum number of registrations accepted by the voice register pool, use the max registrations command in voice register pool configuration mode. To disable registration setup, use the no form of this command.
max registrations value
no max registrations
Syntax Description
value
Digit, beginning with 0, that represents the maximum number of registrations. The maximum registration value is platform dependent.
Defaults
The maximum number of IP phones that can be configured per platform
Command Modes
Voice register pool configuration
Command History
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
12.4(4)T
This command was removed.
Usage Guidelines
The id (voice register pool) command must be configured before any other voice register pool commands, including the max registrations command. The id command identifies a locally available individual Cisco SIP IP phone or sets of Cisco SIP IP phones.
If two phones attempt to register the same phone number, only the first phone can register the number. You can control which phone is accepted by using multiple voice register pools. In general, the best usage is one pool per phone; with multiple pools, some flexibility is granted.
Examples
The following partial sample output from the show running-config command shows that 5 is the maximum number of SIP telephone registrations accepted.
voice register pool 3id network 10.2.161.0 mask 255.255.255.0number 1 95... preference 1cor outgoing call95 1 95011max registrations 5voice-class codec 1Related Commands
Posted: Tue Dec 13 12:26:35 PST 2005
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