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Table Of Contents

Command Reference: N through Z

notify redirect (dial peer)

notify redirect (voice service)

number (voice register pool)

preference (voice register pool)

proxy (voice register pool)

registrar server (SIP)

show sip-ua status registrar

show voice register all

show voice register dial-peers

show voice register global

show voice register pool

show voice register statistics

translate-outgoing (voice register pool)

vad (voice register pool)

voice-class codec (voice register pool)

voice register global

voice register pool


Command Reference: N through Z


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.

notify redirect (dial peer)

notify redirect (voice service)

number (voice register pool)

preference (voice register pool)

proxy (voice register pool)

registrar server (SIP)

show sip-ua status registrar

show voice register all

show voice register dial-peers

show voice register global

show voice register pool

show voice register statistics

translate-outgoing (voice register pool)

vad (voice register pool)

voice-class codec (voice register pool)

voice register global

voice register pool

notify redirect (dial peer)

To send a redirect facility to the application handling the redirect request on a specific VoIP dial peer using the Cisco IOS voice gateway, use the notify redirect command in the dial-peer configuration mode. To return to the default, use the no form of this command.

notify redirect {ip2ip | ip2pots}

no notify redirect

Syntax Description

ip2ip

Sends redirect facility to the application handling redirect requests for IP-to-IP calls.

ip2pots

Sends redirect facility to the application handling redirect requests for IP-to-POTS calls.


Defaults

Disabled

Command Modes

Dial peer configuration

Command History

Cisco IOS Release
Version
Modification

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was introduced.


Usage Guidelines

The notify redirect (dial peer) command must be configured on the inbound dial peer of the gateway. This command enables, on a per dial peer basis, IP-to-IP or IP-to-POTS notify redirection for the gateway.

When notify redirect is configured in dial-peer configuration mode, the configuration for the specific dial peer is activated only if the dial peer is an inbound dial peer. To enable notify redirect globally, use the notify redirect (voice service) command.


Note Use the notify redirect (dial peer) command to configure Cisco SIP SRST 3.4 only after using the allow-connections command to enable B2BUA call flow on the SRST gateway.


Examples

The following is partial sample output from the show running-config command showing that notify redirect has been set up for IP-to-POTS calls on VoIP dial peer 8000:

dial-peer voice 8000 voip destination-pattern 80.. notify redirect ip2pots session protocol sipv2 session target ipv4:1.5.33.200 dtmf-relay rtp-nte codec g711ulaw !

Related Commands

Command
Description

dial peer

Enters dial peer configuration mode for defining a particular dial peer and specifying the method of voice encapsulation.

notify redirect (voice service)

Enables global IP-to-IP or IP-to-POTS notify redirection for all VoIP dial peers.


notify redirect (voice service)

To send a redirect facility to the application handling redirect requests for all VoIp dial peers on the Cisco IOS voice gateway, use the notify redirect command in the voice-service configuration mode. To return to the default, use the no form of this command.

notify redirect {ip2ip | ip2pots}

no notify redirect

Syntax Description

ip2ip

Sends redirect facility to the application handling redirect requests for IP-to-IP calls.

ip2pots

Sends redirect facility to the application handling redirect requests for IP-to-POTS calls.


Defaults

Disabled

Command Modes

Voice-service configuration

Command History

Cisco IOS Release
Version
Modification

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was introduced.


Usage Guidelines

Use this command to enable notify redirection globally on a gateway. Use the notify redirect (dial-peer) command to configure IP-to-IP or IP-to-POTS notify redirection on a specific inbound dial peer.


Note Use the notify redirect (voice service) command to configure Cisco SIP SRST 3.4 only after using the allow-connections command to enable B2BUA call flow on the SRST gateway.


Examples

The following is partial sample output from the show running-config command showing that notify redirect has been set up globally for IP-to-POTS calling:

voice service voip notify redirect ip2pots allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to sip no supplementary-service h450.2 no supplementary-service h450.3 sip registrar server expires max 600 min 60 !

Related Commands

Command
Description

voice service

Enters voice-service configuration mode.

notify redirect (dial peer)

Enables, on a per dial peer basis, IP-to-IP or IP-to-POTS notify redirection on the Cisco IOS voice gateway.


number (voice register pool)

To indicate the E.164 phone numbers that the registrar permits to handle the Register message from the Cisco SIP IP phone, use the number command in voice register pool configuration mode. To disable number registration, use the no form of this command.

number tag number-pattern [preference value] [huntstop]

no number tag number-pattern

Syntax Description

tag

Number from 1 to 10 and the distinguishing factor when there are multiple number commands.

number-pattern

Phone numbers (including wildcards and patterns) that are permitted by the registrar to handle the Register message from the Cisco SIP IP phone.

preference value

(Optional) Defines the number list preference order. Range is from 0 to 10. The highest preference is 0. There is no default.

huntstop

(Optional) Stops hunting if the dial peer is busy.


Defaults

None (see the syntax description for syntax-level defaults)

Command Modes

Voice register pool configuration

Command History

Cisco IOS Release
Version
Modification

12.2(15)ZJ

Cisco SIP SRST 3.0

This command was introduced.

12.3(4)T

Cisco SIP SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was added to Cisco CME.


Usage Guidelines

The number command indicates the phone numbers that are permitted by the registrar to handle the Register message from the Cisco SIP IP phone. The keywords and arguments of this command allow for more explicit setting of user preferences regarding what number patterns should match the voice register pool.


Note Configure the id (voice register pool) command before any other voice register pool commands, including the number command. The id command identifies a locally available individual Cisco SIP IP phone or set of Cisco SIP IP phones.


Examples

The following is partial sample output from the show running-config command showing the number command set to the telephone number pattern 95. Thus, all numbers beginning with 95 are permitted by the registrar to handle the Register message.

voice register pool 3
id network 10.2.161.0 mask 255.255.255.0
number 1 95... preference 1
 cor incoming call95 1 95011

Related Commands

Command
Description

id (voice register pool)

Explicitly identifies a locally available individual Cisco SIP IP phone, or when running Cisco SIP SRST, set of Cisco SIP IP phones.

voice register dn

Enters voice register dn configuration mode to define an extension for a SIP phone line.

voice register pool

Enters voice register pool configuration mode for SIP phones.


preference (voice register pool)

To set the preference order for creating the VoIP dial peers created for a number associated with a voice pool, use the preference command in voice register pool configuration mode. To put the number in default preference order, use the no form of this command.

preference preference-order

no preference

Syntax Description

preference-order

Preference order for the extension or telephone number associated with a directory number. Range is 0 to 10. Default is 0, which is the highest preference.


Defaults

0 (highest preference order)

Command Modes

Voice register pool configuration

Command History

Cisco IOS Release
Version
Modification

12.2(15)ZJ

Cisco SIP SRST 3.0

This command was introduced.

12.3(4)T

Cisco SIP SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was added to Cisco CME.


Usage Guidelines

When you create a voice register pool for a SIP phone or a group of SIP phones in a Cisco CallManager Express (Cisco CME) or Cisco SRST environment, you automatically create a virtual voice port and one to four virtual dial peers to be used by the number associated with that pool. This command sets a preference value for the number that you are creating. The preference value is passed transparently to dial peers created for the number. The preference value allows you to control the selection of a desired dial peer when multiple dial peers are matched on the same destination pattern (extension or telephone number) associated with the pool. In this way, the preference command can be used to establish a hunt strategy for incoming calls.


Note Configure the id (voice register pool) command before any other voice register pool commands, including the preference command. The id command identifies a locally available individual SIP phone or set of Cisco SIP phones.


Examples

The following example sets a preference of 2 for extension number 3000:

voice register pool 1
 number 3000
 preference 2

In the following example, extension number 1222 under voice register dn 4 has a higher preference than number 1222 under voice register pool 5.

voice register pool 4
 number 1222
 preference 0
!
!
voice register dn 5
 number 1222
 preference 1

Related Commands

Command
Description

id (voice register pool)

Explicitly identifies a locally available individual Cisco SIP IP phone, or when running Cisco SIP SRST, set of Cisco SIP IP phones.

voice register pool

Enters voice register pool configuration mode for SIP phones.


proxy (voice register pool)

To autogenerate additional VoIP dial peers to reach the main proxy whenever a Cisco SIP IP phone registers with a SIP SRST gateway, use the proxy command in voice register pool configuration mode. To disable a dial peer as a SIP proxy, use the no form of this command.

proxy ip-address [preference value] [monitor probe {icmp-ping | rtr} [alternate-ip-address]]

no proxy

Syntax Description

ip-address

IP address of the SIP proxy.

preference value

(Optional) Defines the preference of the proxy dial peers that are created. Range is from 0 to 10. The highest preference is 0. There is no default.

monitor probe

(Optional) Enables monitoring of proxy dial peers.

icmp-ping—Enables monitoring of proxy dial peers using ICMP ping.

rtr—Enables monitoring of proxy dial peers using RTR probes.

alternate-ip-address—(Optional) Enables monitoring of alternate IP addresses other than the proxy address. For example, to monitor a gateway front end to a SIP proxy.


Defaults

Proxy dial peer is disabled.

Command Modes

Voice register pool configuration

Command History

Release
Modification

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 proxy command autogenerates additional VoIP dial peers to reach the main proxy whenever a Cisco SIP IP phone registers with a SIP SRST gateway. This autogeneration process enables all PSTN calls to be routed first to the main proxy before the backup dial peers for local Cisco SIP IP phones are tried.

Proxy dial peers can be monitored using ICMP ping or RTR probes, in case of WAN failure. If the SIP SRST gateway loses probes to the main proxy, the proxy dial peers are temporarily set to an operational down state. Then the backup dial peers can be selected faster to lower the call setup time. In addition, the proxy dial peers can be monitored using an alternate location other than the main proxy to monitor the status of the WAN link.

Only one proxy address can be set per voice register pool.

For proxy monitoring to work, the call fallback active command must be configured.


Note The id (voice register pool) command must be configured before any other voice register pool commands, including the proxy command. The id command identifies a locally available individual Cisco SIP IP phone or sets of Cisco SIP IP phones.


Examples

The following partial sample output from the show running-config command shows that voice register pool 1 has defined 10.2.161.187 as the SIP proxy and that it is monitored by ICMP ping:

voice register pool 1
id mac 0030.94C2.A22A
preference 5
cor incoming call91 1 91011
translate-outgoing called 1
proxy 10.2.161.187 preference 1 monitor probe icmp-ping
alias 1 94... to 91011 preference 8
voice-class codec 1

Related Commands

Command
Description

call fallback active

Enables a call request to fall back to alternate dial peers in case of network congestion.

id (voice register pool)

Explicitly identifies a locally available individual Cisco SIP IP phone or set of Cisco SIP IP phones.

voice register pool

Enables SIP SRST voice register pool configuration commands.


registrar server (SIP)

To enable SIP registrar functionality, use the registrar server command in SIP configuration mode. To disable SIP registrar functionality, use the no form of the command.

registrar server [expires [max sec] [min sec] ]

no registrar server

Syntax Description

expires

(Optional) Sets the active time for an incoming registration.

max sec

(Optional) Maximum expires time for a registration, in seconds. The range

is from 600 to 86400. The default is 3600.

min sec

(Optional) Minimum expires time for a registration, in seconds. The range is

from 60 to 3600. The default is 60.


Defaults

Disabled

Command Modes

SIP configuration

Command History

Cisco IOS Release
Version
Modification

12.2(15)ZJ

Cisco SIP SRST 3.0

This command was introduced.

12.3(4)T

Cisco SIP SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was added to Cisco CME.


Usage Guidelines

When this command is entered, the router accepts incoming SIP Register messages. If SIP Register message requests are for a shorter expiration time than what is set with this command, the SIP Register message expiration time is used.

This command is mandatory for Cisco SIP SRST or Cisco CME and must be entered before any voice register pool or voice register global commands are configured.

If the WAN is down and you reboot your Cisco CME or Cisco SIP SRST router, when the router reloads it will have no database of SIP phone registrations. The SIP phones will have to register again, which could take several minutes, because SIP phones do not use a keepalive functionality. To shorten the time before the phones re-register, the registration expiry can be adjusted with this command. The default expiry is 3600 seconds; an expiry of 600 seconds is recommended.

Examples

The following partial sample output from the show running-config command shows that SIP registrar functionality is set:

voice service voip

 allow-connections sip-to-sip

 sip

  registrar server expires max 1200 min 300


Related Commands

Command
Description

sip

Enters SIP configuration mode from voice service VoIP configuration mode.

voice register global

Enters voice register global configuration mode in order to set global parameters for all supported Cisco SIP phones in a Cisco CME or Cisco SIP SRST environment.

voice register pool

Enters voice register pool configuration mode for SIP phones.


show sip-ua status registrar

To display all the SIP endpoints that are currently registered with the contact address, use the show sip-ua status registrar command in privileged EXEC mode.

show sip-ua status registrar

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC

Command History

Release
Modification

12.2(15)ZJ

This command was introduced.

12.3(4)T

This command was integrated into Cisco IOS Release 12.3(4)T.


Examples

The following is sample output from this command:

Router# show sip-ua status registrar

Line destination expires(sec) contact
              call-id
              peer
============ =============== ============ ===============
2800          10.0.2.50        29            10.0.2.50
              00127f3b-90250002-23b0fa96-5900e864@10.0.2.50
              40003
1000          10.0.2.52        372           10.0.2.52
              00127f57-60aa0002-23a74903-48d80f79@10.0.2.52
              40001
2801          10.0.2.51        31            10.0.2.51
              00127f57-628f0002-23b08297-5f020ea1@10.0.2.51
              40002

Table 3 describes significant fields shown in this output.

Table 3 show sip-ua status registrar Field Descriptions

Field
Description

Line

The phone number that maintains registration of SIP devices.

destination

The destination IP address.

call-id

A unique ID assigned for each call.

peer

When a SIP IP phone registers, an associated VoIP dial peer is automatically generated. This dial peer contains general information on how to contact the phone. The information includes the directory number or numbers associated with the phone, and the IP address and protocol of the phone.

expires (sec)

The amount of time, in seconds, until registration expires.

contact

The contact IP address provided by the Cisco SIP IP phone.


Related Commands

Command
Description

registrar server

Enables SIP registrar functionality.


show voice register all

To display all Cisco SIP SRST and Cisco CME configurations and register information, use the show voice register all command in privileged EXEC mode.

show voice register all

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC

Command History

Cisco IOS Release
Version
Modification

12.2(15)ZJ

Cisco SIP SRST 3.0

This command was introduced.

12.3(4)T

Cisco SIP SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was added to Cisco CME.


Examples

Cisco SIP SRST

The following is sample output from this command displaying all register information:

Router# show voice register all

Pool Tag 1
Config:
Network address is 192.168.0.0, Mask is 255.255.0.0
Number list 1 : Pattern is 50.., Preference is 2
Proxy Ip address is 0.0.0.0
Default preference is 2
Incoming called number is
Translate outgoing called tag is 1
Class of Restriction List Tag: default
Incoming corlist name is allowall
Application is default.new

Dialpeers created:

dial-peer voice 40007 voip
application default.new
corlist incoming allowall
preference 2
incoming called-number 5001
destination-pattern 5001
redirect ip2ip
session target ipv4:192.168.0.3
session protocol sipv2
translate-outgoing called 1
voice-class codec 1

Statistics:
Active registrations : 2

Total Registration Statistics
Registration requests : 47
Registration success : 47
Registration failed : 0
unRegister requests : 45
unRegister success : 45
unRegister failed : 0

Cisco CME

The following is sample output from this command displaying all register information:

Router# show voice register all

VOICE REGISTER GLOBAL ===================== CONFIG [Version=4.0(0)] ======================== Version 4.0(0) Mode is cme Max-pool is 24 Max-dn is 72 Source-address is 172.18.202.243 port 5060 Load ata ATA030200SIP041111A.zup Load 7960-40 is P0S3-07-4-00 Time-format is 12 Date-format is YY-M-D Time-zone is 5 Hold-alert is enabled Mwi stutter is enabled Mwi registration for full E.164 is enabled Forwarding local is enabled Dst auto adjust is enabled start at Apr week 1 day Sun time 02:00 stop at Oct week 8 day Sun time 02:00 Voicemail number is 7788 Max redirect number is 20 Telnet Level: 2 Tftp path is system:/cme/sipphone Generate text file is enabled Tftp files are created, current syncinfo 0002917733516824 OS79XX.TXT is not created VOICE REGISTER DN ================= Dn Tag 1 Config: Number is 7001 Preference is 0 Huntstop is disabled Name christoper robert Auto answer is disabled Label is jennifer nicole Dn Tag 2 Config: Number is 7002 Preference is 0 Huntstop is disabled Name Jenny Auto answer is disabled Dn Tag 3 Config: Number is 7003 Preference is 0 Huntstop is disabled Name nino Auto answer is disabled Dn Tag 4 Config: Number is 7004 Preference is 0 Huntstop is disabled Auto answer is disabled Dn Tag 5 Config: Number is 7005 Preference is 0 Huntstop is disabled Name ABBY Auto answer is disabled Dn Tag 6 Config: Number is 7006 Preference is 0 Huntstop is disabled Name jayce Auto answer is disabled MWI registration is enabled. Dn Tag 7 Config: Number is 7007 Preference is 0 Huntstop is disabled Name bugs Auto answer is enabled Label is daffy Dn Tag 8 Config: Number is 7008 Preference is 0 Huntstop is disabled Name Bob Auto answer is disabled VOICE REGISTER TEMPLATE ======================= Temp Tag 1 Config: Attended Transfer is enabled Blind Transfer is enabled Semi-attended Transfer is enabled Conference is enabled Caller-ID block is disabled DnD control is enabled Anoymous call block is disabled Temp Tag 2 Config: Attended Transfer is enabled Blind Transfer is enabled Semi-attended Transfer is enabled Conference is disabled Caller-ID block is disabled DnD control is enabled Anoymous call block is disabled Voicemail is 7788, timeout 5 Temp Tag 3 Config: Attended Transfer is enabled Blind Transfer is enabled Semi-attended Transfer is enabled Conference is enabled Caller-ID block is disabled DnD control is enabled Anoymous call block is disabled Temp Tag 5 Config: Attended Transfer is enabled Blind Transfer is enabled Semi-attended Transfer is enabled Conference is enabled Caller-ID block is disabled DnD control is enabled Anoymous call block is disabled VOICE REGISTER POOL =================== Pool Tag 1 Config: Mac address is 000D.ED22.EDFE Type is 7960 Number list 1 : DN 1 Proxy Ip address is 0.0.0.0 Default preference is 1 DTMF Relay is disabled Call Waiting is disabled DnD is disabled keep-conference is enabled template is 1 Dialpeers created: Statistics: Active registrations : 0 Total Registration Statistics Registration requests : 0 Registration success : 0 Registration failed : 0 unRegister requests : 0 unRegister success : 0 unRegister failed : 0 Pool Tag 2 Config: Mac address is 000D.ED23.CBA0 Type is 7960 Number list 1 : DN 2 Number list 2 : DN 2 Proxy Ip address is 0.0.0.0 Default preference is 1 DTMF Relay is enabled, rtp-nte Call Waiting is enabled DnD is disabled speed-dial 3 7001 speed-dial 4 7701 keep-conference is enabled template is 1 Dialpeers created: dial-peer voice 40003 voip <----------------------------------------------------------------------------------------- --------------------------------- destination-pattern 7002 redirect ip2ip session target ipv4:172.18.202.251:5060 session protocol sipv2 dtmf-relay rtp-nte after-hours-exempt FALSE Statistics: Active registrations : 2 Total Registration Statistics Registration requests : 2 Registration success : 2 Registration failed : 0 unRegister requests : 0 unRegister success : 0 unRegister failed : 0 Pool Tag 3 Config: Mac address is 0030.94C3.035E Type is 7960 Number list 1 : DN 3 Number list 3 : DN 3 Proxy Ip address is 0.0.0.0 DTMF Relay is disabled Call Waiting is enabled DnD is disabled keep-conference is enabled template is 2 Dialpeers created: Statistics: Active registrations : 0 Total Registration Statistics Registration requests : 0 Registration success : 0 Registration failed : 0 unRegister requests : 0 unRegister success : 0 unRegister failed : 0 Pool Tag 5 Config: Mac address is 0012.019B.3FD8 Type is ATA Number list 1 : DN 5 Proxy Ip address is 0.0.0.0 Default preference is 1 DTMF Relay is disabled Call Waiting is enabled DnD is disabled keep-conference is enabled Dialpeers created: Statistics: Active registrations : 0 Total Registration Statistics Registration requests : 0 Registration success : 0 Registration failed : 0 unRegister requests : 0 unRegister success : 0 unRegister failed : 0 Pool Tag 6 Config: Mac address is 0012.019B.3E88 Type is ATA Number list 1 : DN 6 Number list 2 : DN 7 Proxy Ip address is 0.0.0.0 Default preference is 1 DTMF Relay is enabled, rtp-nte Call Waiting is enabled DnD is disabled call-forward b2bua all 7788 keep-conference is enabled template is 2 Dialpeers created: dial-peer voice 40001 voip <----------------------------------------------------------------------------------------- ------------------ destination-pattern 7006 redirect ip2ip session target ipv4:172.18.202.32:5060 session protocol sipv2 dtmf-relay rtp-nte call-fwd-all 7788 after-hours-exempt FALSE dial-peer voice 40002 voip destination-pattern 7007 redirect ip2ip session target ipv4:172.18.202.32:5060 session protocol sipv2 dtmf-relay rtp-nte call-fwd-all 7788 after-hours-exempt FALSE Statistics: Active registrations : 2 Total Registration Statistics Registration requests : 2 Registration success : 2 Registration failed : 0 unRegister requests : 0 unRegister success : 0 unRegister failed : 0 Nothing configured yet Pool Tag 8 Config: Mac address is 0006.D737.CC42 Type is 7940 Number list 1 : DN 8 Proxy Ip address is 0.0.0.0 Default preference is 1 DTMF Relay is disabled Call Waiting is enabled DnD is disabled keep-conference is enabled template is 5 Dialpeers created: Statistics: Active registrations : 0 Total Registration Statistics Registration requests : 0 Registration success : 0 Registration failed : 0 unRegister requests : 0 unRegister success : 0 unRegister failed : 0 Pool Tag 9 Config: Mac address is 0030.94C3.0831 Proxy Ip address is 0.0.0.0 DTMF Relay is disabled Call Waiting is enabled DnD is disabled keep-conference is enabled Dialpeers created: Statistics: Active registrations : 0 Total Registration Statistics Registration requests : 0 Registration success : 0 Registration failed : 0 unRegister requests : 0 unRegister success : 0 unRegister failed : 0 Pool Tag 10 Config: Mac address is 000D.ED22.EDFE Proxy Ip address is 0.0.0.0 DTMF Relay is disabled Call Waiting is disabled DnD is disabled call-forward b2bua all 1234 keep-conference is enabled Dialpeers created: Statistics: Active registrations : 0 Total Registration Statistics Registration requests : 0 Registration success : 0 Registration failed : 0 unRegister requests : 0 unRegister success : 0 unRegister failed : 0 Nothing configured yet

Table 4 describes significant fields shown in this output.

Table 4 show voice register all Field Descriptions 

Field
Description

Pool Tag

Used with the all and pool keywords. Shows the assigned tag number of the current pool.

Config:

Used with the all and pool keywords. Shows the voice register pool.

Network address and Mask

Used with the all and pool keywords. Shows network address and mask information if the id command is configured.

Number list, Pattern, and Preference

Used with the all and pool keywords. Shows the number command configuration.

Proxy Ip address

Used with the all and pool keywords. Shows the proxy command configuration.

Default preference

Used with the all and pool keywords. Shows the default preference value of this pool.

Incoming called number

Used with the all and pool keywords. Shows the incoming called-number command configuration.

Translate outgoing called tag

Used with the all and pool keywords. Shows the translate-outgoing command configuration.

Class of Restriction List Tag

Used with the all and pool keywords. Shows the COR tag.

Incoming corlist name

Used with the all and pool keywords. Shows the cor command configuration.

Application

Used with the all and pool keywords.  Shows the application command configuration for this pool.

Dialpeers created:

Used with the all and pool keywords. What follows is a list of all dial peers created and their contents. Dial-peer contents differ per application and are not described here.

Statistics:

Used with the all, pool, and statistics keywords. Shows the registration statistics for this pool.

Active registrations

Used with the all, pool, and statistics keywords. Shows the current active registrations.

Total Registration Statistics

Used with the all, pool, and statistics keywords. Shows the total registration statistics for this pool.

Registration requests

Used with the all, pool, and statistics keywords. Shows the incoming registration requests.

Registration success

Used with the all, pool, and statistics keywords. Shows the successful registrations.

Registration failed

Used with the all, pool, and statistics keywords. Shows the failed registrations.

unRegister requests

Used with the all, pool, and statistics keywords. Shows the incoming unregister/registration expire requests.

unRegister success

Used with the all, pool, and statistics keywords. Reports the number of successful unregisters.

unRegister failed

Used with the all, pool, and statistics keywords. Reports the number of failed unregisters.


Related Commands

Command
Description

show sip-ua status registrar

Displays all the SIP endpoints currently registered with the contact address.

show voice register all

Displays all Cisco SIP SRST and Cisco CME configurations and register information.

show voice register dial-peer

Displays details of all dynamically created VoIP dial peers associated with the Cisco SIP SRST or Cisco CME register event

show voice register pool

Displays all configuration information associated with a particular voice register pool.


show voice register dial-peers

To display details of all dynamically created VoIP dial peers associated with the Cisco SIP SRST or Cisco CME register event, use the show voice register dial-peers command in privileged EXEC mode.

show voice register dial-peers

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC

Command History

Cisco IOS Release
Version
Modification

12.2(15)ZJ

Cisco SIP SRST 3.0

This command was introduced.

12.3(4)T

Cisco SIP SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was added to Cisco CME.


Examples

Cisco SIP SRST

The following is sample output from this command displaying all dial peers:

Router# show voice register dial-peers

dial-peer voice 40024 voip
corlist incoming call91
preference 5
destination-pattern 91011
redirect ip2ip
session target ipv4:192.168.0.2
session protocol sipv2
translate-outgoing called 1
voice-class codec 1

dial-peer voice 40025 voip
destination-pattern 40891011
redirect ip2ip
session target ipv4:192.168.0.2
session protocol sipv2
translate-outgoing called 1
voice-class codec 1

dial-peer voice 40026 voip
preference 8
destination-pattern 94...
redirect ip2ip
session target ipv4:192.168.0.2
session protocol sipv2
translate-outgoing called 1
voice-class codec 1

dial-peer voice 40027 voip
preference 1
destination-pattern 91011
redirect ip2ip
session target ipv4:10.2.161.187
session protocol sipv2
voice-class codec 1
monitor probe icmp-ping 10.2.161.187

Related Commands

Command
Description

show sip-ua status registrar

Displays all the SIP endpoints currently registered with the contact address.

show voice register all

Displays all Cisco SIP SRST and Cisco CME configurations and register information.

show voice register pool

Displays all configuration information associated with a particular voice register pool.


show voice register global

To display all global configuration parameters associated with SIP phones, use the show voice register global command in privileged EXEC mode.

show voice register global

Syntax Description

This command has no arguments or keywords.

Command Default

Privileged EXEC

Command History

Cisco IOS Release
Version
Modification

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was introduced.


Examples

Cisco CME

The following is sample output from this command:

Router# show voice register global
CONFIG [Version=3.4(0)]
========================
Version 3.4(0)
Mode is cme
Max-pool is 48
Max-dn is 48
Source-address is 10.0.2.4 port 5060
Load 7960-40 is P0S3-07-4-07
Time-format is 12
Date-format is M/D/Y
Time-zone is 5
Hold-alert is disabled
Mwi stutter is disabled
Mwi registration for full E.164 is disabled
Forwarding local is enabled
Dst auto adjust is enabled
 start at Apr week 1 day Sun time 02:00
 stop at Oct week 8 day Sun time 02:00
Max redirect number is 5
Telnet Level: 2
Tftp path is system:/cme/sipphone
Generate text file is disabled
Tftp files are created, current syncinfo 0002830590524159
OS79XX.TXT is not created
Router#

Cisco SIP SRST

Router# show voice register global

CONFIG [Version=3.4(0)]
======================================================
   Version 3.4(0)
   Mode is SIP SRST
   Max-pool is 10
   Max-dn is 10

Table 5 contains descriptions of significant fields shown in this output, listed in alphabetical order.

Table 5 show voice register global Field Descriptions 

Field
Description

Date-format

Value of date-format command.

DST auto adjust

Setting of dst auto-adjust command.

Forwarding local

Setting of forwarding local command.

Generate text file

Setting of text file command.

Hold-alert

Setting of hold-alert command.

Load

Value of load command.

Max-dn

Reports the maximum number of SIP voice register directory numbers (dns) supported by the Cisco SIP CME or Cisco SIP SRST router as configured with the max-dn command. The maximum possible number is platform-dependent.

Max-pool

Reports the maximum number of SIP voice register pools supported by the Cisco SIP SRST or Cisco CME router as configured with the max-pool command. The maximum possible number is platform-dependent.

Max redirect number

Maximum number of redirects set with the max-redirect command.

Mode

Reports the mode as configured with the mode command. Value can be either Cisco CME or Cisco SIP SRST.

MWI registration

Setting of mwi command.

MWI stutter

Setting of mwi stutter command.

Time-format

Value of time-format command.

Time-zone

Number of the timezone selected with the timezone command.

TFTP path

Directory location of provisioning files for SIP phones that is specified with the tftp-path command.

Version

Reports the Cisco SIP SRST or Cisco CME version number.


Related Commands

Command
Description

show sip-ua status registrar

Displays all the SIP endpoints currently registered with the contact address.

show voice register all

Displays all Cisco SIP SRST and Cisco CME configurations and register information.

show voice register dial-peer

Displays details of all dynamically created VoIP dial peers associated with the Cisco SIP SRST or Cisco CME register event.

voice register global

Enters voice register global configuration mode in order to set global parameters for all supported Cisco SIP phones in a Cisco CME or Cisco SIP SRST environment.


show voice register pool

To display all configuration information associated with a particular voice register pool, use the show voice register pool command in privileged EXEC mode.

show voice register pool pool-tag

Syntax Description

pool-tag

Tag number of the voice register pool for which to display information. The maximum number of pools is version and platform dependent; refer to Cisco IOS command-line interface (CLI) help.


Command Modes

Privileged EXEC

Command History

Cisco IOS Release
Version
Modification

12.2(15)ZJ

Cisco SIP SRST

This command was introduced.

12.3(4)T

Cisco SIP SRST

This command was integrated into Cisco IOS Release 12.3(4)T.

12.4(4)T

Cisco CME 3.4

This command was introduced.


Examples

Cisco CME

The following is sample output from this command displaying information for voice register pool 33:

Router# show voice register pool 33
Pool Tag 33
Config:
 Mac address is 0009.B7F7.532E
 Type is 7960
 Number list 1 : DN 1
 Number list 2 : DN 2
 Number list 3 : DN 3
 Number list 4 : DN 4
 Number list 5 : DN 5
 Number list 6 : DN 6
 Proxy Ip address is 0.0.0.0
 DTMF Relay is disabled
 Call Waiting is enabled
 keep-conference is enabled
 template is 1

Cisco SIP SRST

The following is sample output from this command displaying all information for voice register pool 1:

Router# show voice register pool 1

Pool Tag 1
Config:
Network address is 192.168.0.0, Mask is 255.255.0.0
Number list 1 : Pattern is 50.., Preference is 2
Proxy Ip address is 0.0.0.0
Default preference is 2
Incoming called number is
Translate outgoing called tag is 1
Class of Restriction List Tag: default
Incoming corlist name is allowall
Application is default.new

Dialpeers created:

dial-peer voice 40007 voip
application default.new
corlist incoming allowall
preference 2
incoming called-number 5001
destination-pattern 5001
redirect ip2ip
session target ipv4:192.168.0.3
session protocol sipv2
translate-outgoing called 1
voice-class codec 1

Statistics:
Active registrations : 2

Total Registration Statistics
Registration requests : 48
Registration success : 48
Registration failed : 0
unRegister requests : 46
unRegister success : 46
unRegister failed : 0

Table 6 contains descriptions of significant fields shown in the Cisco SIP SRST and Cisco CME output, listed in alphabetical order.

Table 6 show voice register pool Field Descriptions 

Field
Description

Active registrations

Used with the all, pool, and statistics keywords. Shows the current active registrations.

Application

Used with the all and pool keywords.  Shows the application command configuration for this pool.

Call Waiting

Setting of call-waiting command.

Config:

Used with the all and pool keywords. Shows the voice register pool.

Class of Restriction List Tag

Used with the all and pool keywords. Shows the COR tag.

Default preference

Used with the all and pool keywords. Shows the default preference value of this pool.

Dialpeers created:

Used with the all and pool keywords. What follows is a list of all dial peers created and their contents. Dial-peer contents differ per application and are not described here.

DnD

Setting of dnd-control command.

DTMF Relay

Setting of dtmf-relay command.

Incoming called number

Used with the all and pool keywords. Shows the incoming called-number command configuration.

Incoming corlist name

Used with the all and pool keywords. Shows the cor command configuration.

keep-conference

Status of keep-conference command.

Mac address

MAC address of this SIP phone as defined by using the id command.

Network address and Mask

Used with the all and pool keywords. Shows network address and mask information if the id command is configured.

Number list, Pattern, and Preference

Used with the all and pool keywords. Shows the number (voice register pool) command configuration.

Pool Tag

Used with the all and pool keywords. Shows the assigned tag number of the current pool.

Proxy Ip address

Used with the all and pool keywords. Shows the proxy command configuration; that is, the IP address of external SIP server.

Registration failed

Used with the all, pool, and statistics keywords. Shows the failed registrations.

Registration requests

Used with the all, pool, and statistics keywords. Shows the incoming registration requests.

Registration success

Used with the all, pool, and statistics keywords. Shows the successful registrations.

Statistics:

Used with the all, pool, and statistics keywords. Shows the registration statistics for this pool.

Template

Template-tag number for template that is applied to this SIP phone.

Translate outgoing called tag

Used with the all and pool keywords. Shows the translate-outgoing command configuration.

Total Registration Statistics

Used with the all, pool, and statistics keywords. Shows the total registration statistics for this pool.

Type

Phone type identified for this SIP phone using the type command.

unRegister requests

Used with the all, pool, and statistics keywords. Shows the incoming unregister/registration expire requests.

unRegister success

Used with the all, pool, and statistics keywords. Reports the number of successful unregisters.

unRegister failed

Used with the all, pool, and statistics keywords. Reports the number of failed unregisters.

Username Password

Values within authentication credential.


Related Commands

Command
Description

show sip-ua status registrar

Displays all the SIP endpoints currently registered with the contact address.

show voice register all

Displays all Cisco SIP SRST and Cisco CME configurations and register information.

show voice register dial-peer

Displays details of all dynamically created VoIP dial peers associated with the Cisco SIP SRST or Cisco CME register event.

voice register pool

Enters voice register pool configuration mode for SIP phones.


show voice register statistics

To display statistics associated with the register event, use the show voice register statistics command in privileged EXEC mode.

show voice register statistics

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC

Command History

Cisco IOS Release
Version
Modification

12.2(15)ZJ

Cisco SIP SRST 3.0

This command was introduced.

12.3(4)T

Cisco SIP SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was added to Cisco CME.


Usage Guidelines

When using the show voice register statistics command, you can verify that the number of Registration and unRegister successes for global statistics are the sum of the values in the individual pools. Because some Registrations fail even before matching a voice register pool, for Registration and unRegister failed statistics the value is not the sum of the values in the individual pools. Immediate failures are accounted in the global statistics.

Examples

The following is sample output from this command displaying all statistical information:

Router# show voice register statistics

Global statistics
Active registrations : 3
Total Registration Statistics
Registration requests : 7
Registration success : 4
Registration failed : 3
unRegister requests : 1
unRegister success : 1
unRegister failed : 0
Register pool 1 statistics
Active registrations : 1
Total Registration Statistics
Registration requests : 3
Registration success : 2
Registration failed : 1
unRegister requests : 1
unRegister success : 1
unRegister failed : 0
Register pool 2 statistics
Active registrations : 2
Total Registration Statistics
Registration requests : 2
Registration success : 2
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0

Table 7 describes significant fields shown in this output.

Table 7 show voice register statistics Field Descriptions 

Field
Description

Statistics:

Used with the all, pool, and statistics keywords. Shows the registration statistics for this pool.

Active registrations

Used with the all, pool, and statistics keywords. Shows the current active registrations.

Total Registration Statistics

Used with the all, pool, and statistics keywords. Shows the total registration statistics for this pool.

Registration requests

Used with the all, pool, and statistics keywords. Shows the incoming registration requests.

Registration success

Used with the all, pool, and statistics keywords. Shows the successful registrations.

Registration failed

Used with the all, pool, and statistics keywords. Shows the failed registrations.

unRegister requests

Used with the all, pool, and statistics keywords. Shows the incoming unregister/registration expire requests.

unRegister success

Used with the all, pool, and statistics keywords. Reports the number of successful unregisters.

unRegister failed

Used with the all, pool, and statistics keywords. Reports the number of failed unregisters.

Global statistics

Used with the statistics keyword. Details all active registrations.

Register pool number statistics

Used with the statistics keyword. Details specific pool statistics.


Related Commands

Command
Description

show sip-ua status registrar

Displays all the SIP endpoints currently registered with the contact address.

show voice register all

Displays all Cisco SIP SRST and Cisco CME configurations and register information.

show voice register dial-peers

Displays details of all dynamically created VoIP dial peers associated with the Cisco SIP SRST or Cisco CME register event.

show voice register pool

Displays all configuration information associated with a particular voice register pool.


translate-outgoing (voice register pool)

To allow an explicit setting of translation rules on the VoIP dial peer in order to modify a phone number dialed by any Cisco IP phone user, use the translate-outgoing command in voice register pool configuration mode. To disable translation rules, use the no form of this command.

translate-outgoing {called | calling} rule-tag

no translate-outgoing {called | calling}

Syntax Description

called

Called party requires translation.

calling

Calling party requires translation.

rule-tag

The rule-tag is an arbitrarily chosen number by which the rule set is

referenced. The range is from 1 to 2147483.


Defaults

None

Command Modes

Voice register pool configuration

Command History

Cisco IOS Release
Version
Modification

12.2(15)ZJ

Cisco SIP SRST 3.0

This command was introduced.

12.3(4)T

Cisco SIP SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was added to Cisco CME.


Usage Guidelines

Translation rules are a powerful general-purpose number-manipulation mechanism that perform operations such as automatically adding telephone area and prefix codes to dialed numbers. The translation rules are applied to VoIP dial peers created by Cisco SIP SRST or Cisco CME.

During registration, a dial peer is created, and that dial peer includes a default translation rule. The translate-outgoing command allows you to change the translation rule, if desired. The translate-outgoing command allows you to select a preconfigured number translation rule to modify the number dialed by a specific extension.


Note Translation rules must be set by using the translate-outgoing command before the alias command is configured in Cisco SIP SRST.

Configure the id (voice register pool) command before any other voice register pool commands, including the translate-outgoing command. The id command identifies a locally available individual SIP phone or set of SIP phones.


Examples

Cisco CME

The following is partial sample output from the show running-config command showing that called-party 1 requires translation.

voice register pool 1
 id mac 0030.94C2.A22A
 preference 5
 cor incoming call91 1 91011
 translate-outgoing called 1

Cisco SIP SRST

The following is partial sample output from the show running-config command showing that called-party 1 requires translation.

voice register pool 1
id mac 0030.94C2.A22A
preference 5
cor incoming call91 1 91011
translate-outgoing called 1
proxy 10.2.161.187 preference 1 monitor probe icmp-ping
alias 1 94... to 91011 preference 8
voice-class codec 1

Related Commands

Command
Description

alias (voice register pool)

Allows Cisco SIP IP phones to handle inbound PSTN calls to telephone numbers that are unavailable when the main proxy is not available.

id (voice register pool)

Explicitly identifies a locally available individual Cisco SIP IP phone, or when running Cisco SIP SRST, set of Cisco SIP IP phones.

translate-outgoing (dial-peer)

Applies a translation rule to manipulate dialed digits on an outbound POTS or VoIP call leg.

voice register pool

Enters voice register pool configuration mode for SIP phones.


vad (voice register pool)

To enable voice activity detection (VAD) on a VoIP dial peer, use the vad command in voice register pool configuration mode. To disable VAD, use the no form of this command.

vad

no vad

Syntax Description

This command has no arguments or keywords.

Defaults

Enabled

Command Modes

Voice register pool configuration

Command History

Cisco IOS Release
Version
Modification

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was introduced.


Usage Guidelines

VAD detects periods of silence in the voice signal and temporarily discontinues transmission of the signal during these periods to save bandwidth. Because VAD is enabled by default, there is no comfort noise during periods of silence. As a result, the call may seem to be disconnected and you may prefer to set no vad on the SIP phone pool.

Examples

The following example disables VAD for pool 1:

Router(config)# voice register pool 1
Router(config-register-pool)# no vad

Related Commands

Command
Description

voice register pool

Enters voice register pool configuration mode for SIP phones.


voice-class codec (voice register pool)

To assign a previously configured codec selection preference list, use the voice-class codec command in voice register pool configuration mode. To remove the codec preference assignment from the voice register pool, use the no form of this command.

voice-class codec tag

no voice-class codec

Syntax Description

tag

Unique number assigned to the voice class. Range is from 1 to 10000. The tag number maps to the tag number created by using the voice class codec (dial peer) command.


Command Default

None

Command Modes

Voice register pool configuration

Command History

Cisco IOS Release
Version
Modification

12.2(15)ZJ

Cisco SIP SRST 3.0

This command was introduced.

12.3(4)T

Cisco SIP SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

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 includes codec g729r8 by default. The voice-class codec command allows you to change the automatically selected default codec, if desired.

You can assign one voice class to each voice register pool. If you assign another voice class to a pool, the last voice class assigned replaces the previous voice class.


Note The id (voice register pool) command is required and must be configured before any other voice register pool commands. The id command identifies a locally available individual Cisco SIP IP phone or set of Cisco SIP IP phones.


Examples

The following partial sample output from the show running-config command shows that voice register pool 1 has been set up to use the previously configured codec voice class 1:

voice register pool 1
id mac 0030.94C2.A22A
preference 5
cor incoming call91 1 91011
translate-outgoing called 1
proxy 10.2.161.187 preference 1 monitor probe icmp-ping
alias 1 94... to 91011 preference 8
voice-class codec 1

Related Commands

Command
Description

codec (voice register pool)

Specifies the codec supported by a single Cisco SIP phone or a VoIP dial peer in a Cisco SIP SRST or a Cisco CME environment.

id (voice register pool)

Explicitly identifies a locally available individual Cisco SIP IP phone, or when running Cisco SIP SRST, set of Cisco SIP IP phones.

voice register pool

Enters voice register pool configuration mode for SIP phones.

voice class codec (dial-peer)

Assigns a previously configured codec selection preference list (codec voice class) to a VoIP dial peer.


voice register global

To enter voice register global configuration mode in order to set global parameters for all supported Cisco SIP phones in a Cisco CME or Cisco SIP SRST environment, use the voice register global command in global configuration mode. To remove the configuration, use the no form of this command.

voice register global

no voice register global

Syntax Description

This command has no arguments or keywords.

Defaults

No default behavior or values

Command Modes

Global configuration

Command History

Cisco IOS Release
Version
Modification

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was introduced.


Usage Guidelines

Cisco CME

Use this command to set provisioning parameters for all supported SIP phones in a Cisco CME system.

Cisco SIP SRST

Use this command to set provisioning parameters for multiple pools; that is, all supported Cisco SIP IP phones in a SIP SRST environment.

Examples

Cisco CME

The following is partial sample output from the show voice register global command. All of the parameters listed were set under voice register global configuration mode:

Router# show voice register global
CONFIG [Version=4.0(0)]
========================
Version 4.0(0)
Mode is cme
Max-pool is 48
Max-dn is 48
Source-address is 10.0.2.4 port 5060
Load 7960-40 is P0S3-07-4-07
Time-format is 12
Date-format is M/D/Y
Time-zone is 5
Hold-alert is disabled
Mwi stutter is disabled
Mwi registration for full E.164 is disabled
Dst auto adjust is enabled
 start at Apr week 1 day Sun time 02:00
 stop at Oct week 8 day Sun time 02:00

Related Commands

Command
Description

allow connections sip to sip

Allows connections between SIP endpoints in a Cisco multiservice IP-to-IP gateway.

application (voice register global)

Selects the session-level application for all dial peers associated with SIP phones.

mode (voice register global)

Enables the mode for provisioning SIP phones in a Cisco CallManager Express (Cisco CME) system.

voice register dn

Enters voice register dn configuration mode to define an extension for a SIP phone line.

voice register pool

Enters voice register pool configuration mode for Cisco SIP IP phones.

voice register template

Enters voice register template configuration mode and defines a template of common parameters for SIP phones.


voice register pool

To enter voice register pool configuration mode for SIP phones, use the voice register pool command in global configuration mode. To remove the pool configuration, use the no form of this command.

voice register pool pool-tag

no voice register pool pool-tag

Syntax Description

pool-tag

Unique number assigned to the pool. Range is 1 to 100.

Note For Cisco CME systems, the upper limit for this argument is defined by the max-pool command.


Defaults

No default behavior or values

Command Modes

Global configuration

Command History

Cisco IOS Release
Version
Modification

12.2(15)ZJ

Cisco SIP SRST 3.0

This command was introduced.

12.3(4)T

Cisco SIP SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was added to Cisco CME.


Usage Guidelines

Cisco CME

Use this command to set phone-specific parameters for SIP phones in a Cisco CallManager Express (Cisco CME) system. Before using this command, enable the mode cme command and set the maximum number of SIP phones supported in your system by using the max-pool command.

Cisco SIP SRST

Use this command to enable user control on which registrations are to be accepted or rejected by a SIP SRST device. The voice register pool command mode can be used for specialized functions and to restrict registrations on the basis of MAC, IP subnet, and number range parameters.

Examples

Cisco CME

The following example enters voice register pool configuration mode and forwards calls to extension 9999 when extension 2001 is busy:

Router(config)# voice register pool 10
Router(config-register-pool)# type 7960
Router(config-register-pool)# number 1 2001
Router(config-register-pool)# call-forward busy 9999 mailbox 1234

Cisco SIP SRST

The following partial sample output from the show running-config command shows that several voice register pool commands are configured within voice register pool 3:

voice register pool 3
id network 10.2.161.0 mask 255.255.255.0
number 1 95... preference 1
 cor outgoing call95 1 95011
 max registrations 5
voice-class codec 1

Related Commands

Command
Description

max-pool (voice register global)

Sets the maximum number of SIP phones that are supported by a Cisco CallManager Express (Cisco CME) system.

mode (voice register global)

Enables the mode for provisioning SIP phones in a Cisco CME system.

number (voice register pool)

Configures a valid number for a SIP phone.

type (voice register pool)

Defines a Cisco IP phone type.

voice register dn

Enters voice register dn configuration mode to define an extension for a SIP phone line.

voice register global

Enters voice register global configuration mode in order to set global parameters for all supported Cisco SIP phones in a Cisco CME or Cisco SIP SRST environment.



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Posted: Tue Dec 13 12:26:31 PST 2005
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