Prerequisites for Configuring the SIP Registrar
Complete the prerequisites documented in the Prerequisites for Configuring Cisco Unified SIP SRST section in Cisco Unified SRST Feature Overview chapter.
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Session Initiation Protocol (SIP) registrar functionality in Cisco IOS software is an essential part of Cisco Unified SIP Survivable Remote Site Telephony (SRST). According to RFC 3261, a SIP registrar is a server that accepts Register requests and is typically collocated with a proxy or redirect server. A SIP registrar may also offer location services.
Complete the prerequisites documented in the Prerequisites for Configuring Cisco Unified SIP SRST section in Cisco Unified SRST Feature Overview chapter.
See the restrictions documented in the Restrictions for Configuring Cisco Unified SIP SRST section in the Cisco Unified SRST Feature Overview chapter.
Cisco Unified SIP SRST provides backup to an external SIP call control (IP-PBX) by providing basic registrar and call handling services. These services are used by a SIP IP phone in the event of a WAN connection outage when the SIP phone is unable to communicate with its primary SIP proxy. The Cisco Unified SIP SRST device also provides PSTN gateway access for placing and receiving PSTN calls.
Cisco Unified SIP SRST works for the following types of calls:
Local SIP IP phone to local SIP phone, if the main proxy is unavailable.
Additional services like class of restriction (COR) for local SIP IP phones to the outgoing PSTN. For example, to block outgoing 1-900 numbers.
How to Configure the SIP Registrar
The local SIP gateway that becomes the SIP registrar acts as a backup SIP proxy and accepts SIP Register messages from SIP phones. It becomes a location database of local SIP IP phones.
A registrar accepts SIP Register requests and dynamically builds VoIP dial peers, allowing the Cisco IOS voice gateway software to route calls to SIP phones.
If a SIP Register request has a Contact header that includes a DNS address, the Contact header is resolved before the contact is added to the SIP registrar database. This is done because during a WAN failure (and the resulting Cisco Unified SIP SRST functionality), DNS servers may not be available.
SIP registrar functionality is enabled with the following configuration. By default, Cisco Unified SIP SRST is not enabled and cannot accept SIP Register messages. The following configuration must be set up to accept incoming SIP Register messages.
Command or Action | Purpose | |||
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Step 1 |
enable Example:
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Enables privileged EXEC mode.
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Step 2 |
configure terminal Example:
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Enters global configuration mode. |
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Step 3 |
voice service voip Example:
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Enters voice service configuration mode. |
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Step 4 |
allow-connections sip to sip Example:
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Allows connections from SIP to SIP endpoints. |
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Step 5 |
sip Example:
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Enters SIP configuration mode. |
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Step 6 |
registrar server [ expires [ maxsec] [minsec] ] Example:
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Enables SIP registrar functionality. The keywords and arguments are defined as follows:
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Step 7 |
end Example:
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Returns to privileged EXEC mode. |
For incoming SIP Register messages to be successfully accepted, users must also set up a voice register pool. See the section Configuring Backup Registrar Service to SIP Phones.
Backup registrar service to SIP IP phones can be provided by configuring a voice register pool on SIP gateways. The voice register pool configuration provides registration permission control and can also be used to configure some dial-peer attributes that are applied to the dynamically created VoIP dial peers when SIP phone registrations match the pool. The following call types are supported:
SIP IP phone to or from:
Local PSTN
Local analog FXS phones
Local SIP IP phone
The commands in the configuration below provide registration permission control and set up a basic voice register pool. The pool gives users control over which registrations are accepted by a Cisco Unified SIP SRST device and which can be rejected. Registrations that match this pool create VoIP SIP dial peers with the dial-peer attributes set to these configurations. Although only the id command is mandatory, this configuration example shows basic functionality.
For command-level information, see the appropriate command page in Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions).
The SIP registrar must be configured before a voice register pool is set up. See the section Configuring the SIP Registrar.
Restrictions
The id command identifies the individual SIP IP phone or sets of SIP IP phones that are to be configured. Thus, theid command configured in Step 5 is required and must be configured before any other voice register pool commands. When themacaddress 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.
Proxy dial peers are autogenerated dial peers that route all calls from the PSTN to Cisco Unified SIP SRST. When a SIP phone registers to Cisco Unified SIP SRST and the proxy command is enabled, two dial peers are automatically created. The first dial peer routes to the proxy, and the second (or fallback) dial peer routes to the SIP phone. The same functionality can also be achieved with the appropriate creation of static dial peers (manually creating dial peers that point to the proxy). Proxy dial peers can be monitored to one proxy IP address, only. That is, only one proxy from a voice registration pool can be monitored at a time. If more than one proxy address needs to be monitored, you must manually create and configure additional dial peers.
If Jabber for desktop clients must register with Unified SRST, ensure thatvoice register pools are configured for all desktop computer networks.
Note |
To monitor SIP proxies, the call fallback active command must be configured, as described in Step 3 |
Command or Action | Purpose | |||
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Step 1 |
enable Example:
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Enables privileged EXEC mode.
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Step 2 |
configure terminal Example:
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Enters global configuration mode. |
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Step 3 |
call fallback active Example:
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Enables a call request to fall back to alternate dial peers in case of network congestion. This command is used if you want to monitor the proxy dial peer and fallback to the next preferred dial peer. For full information on the call fallback active command, see PSTN Fallback Feature. |
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Step 4 |
voice register pool tag Example:
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Enters voice register pool configuration mode for SIP phones. Use this command to control which registrations are accepted or rejected by a Cisco Unified SIP SRST device. |
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Step 5 |
id { network address mask mask | ip address mask mask | mac address } Example:
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Explicitly identifies a locally available individual or set of SIP IP phones. The keywords and arguments are defined as follows:
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Step 6 |
preference preference-order Example:
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Sets the preference order for the VoIP dial peers to be created. Range is from 0 to 10. Default is 0, which is the highest preference. The preference must be greater (lower priority) than the preference configured with the preference keyword in the proxy command. |
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Step 7 |
proxy ip-address [preference value [ monitor probe {icmp-ping | rtr } alternate-ip-address ]] Example:
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Autogenerates additional VoIP dial peers to reach the main SIP proxy whenever a Cisco Unified SIP IP Phone registers with a Cisco Unified SIP SRST gateway. The keywords and arguments are defined as follows:
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Step 8 |
voice-class codec tag Example:
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Sets the voice class codec parameters. The tag argument is a codec group number between 1 and 10000. |
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Step 9 |
(Optional) application application-name Example:
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(Optional)
Selects the session-level application on the VoIP dial peer. Use the application-name argument to define a specific interactive voice response (IVR) application. |
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Step 10 |
end Example:
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Returns to privileged EXEC mode. |
There are several more voice register pool commands that add functionality, but that are not required. See the section Configuring Backup Registrar Service to SIP Phones (Using Optional Commands) for these commands.
The prior configurations set up a basic voice register pool. The configuration in this procedure adds optional attributes to increase functionality.
Prerequisites as described in the Configuring Backup Registrar Service to SIP Phones section.
Configuration of the required commands as described in the Configuring Backup Registrar Service to SIP Phones section .
Before configuring the alias command, translation rules must be set using the translate-outgoing (voice register pool) command.
Command or Action | Purpose | |
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Step 1 |
enable Example:
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Enables privileged EXEC mode.
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Step 2 |
configure terminal Example:
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Enters global configuration mode. |
Step 3 |
voice register pool tag Example:
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Enters voice register pool configuration mode. Use this command to control which registrations are accepted or rejected by a Cisco Unified SIP SRST device. |
Step 4 |
translation-profile outgoing profile-tag Example:
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Use this command to apply the translation profile to a specific directory number or to all directory numbers on a SIP phone. Profile-tag: Translation profile name to handle translation to outgoing calls. |
Step 5 |
alias tag pattern to target [ preference value ] Example:
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Allows Cisco Unified SIP IP Phones to handle inbound PSTN calls to telephone numbers that are unavailable when the main proxy is not available. The keywords and arguments are defined as follows:
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Step 6 |
cor {incoming | outgoing} cor-list-name {cor-list-number starting-number [- ending-number] | default } Example:
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Configures a class of restriction (COR) on the VoIP dial peers associated with directory numbers. COR specifies which incoming dial peers can use which outgoing dial peers to make a call. Each dial peer can be provisioned with an incoming and outgoing COR list. The keywords and arguments are defined as follows:
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Step 7 |
incoming called-number [ number ] Example:
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Applies incoming called parameters to dynamically created dial peers. The number argument is optional and indicates a sequence of digits that represent a phone number prefix. |
Step 8 |
number tag number-pattern { preference value } [huntstop ] Example:
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Indicates the E.164 phone numbers that the registrar permits to handle the Register message from the Cisco Unified SIP IP Phone. The keywords and arguments are defined as follows:
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Step 9 |
dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify] Example:
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Specifies how a SIP gateway relays dual tone multifrequency (DTMF) tones between telephony interfaces and an IP network. The keywords are defined as follows:
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Step 10 |
end Example:
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Returns to privileged EXEC mode. |
The following partial output from the show running-config command shows that voice register pool 12 is configured to accept all registrations from SIP IP phones with extension number 50xx from the 172.16.0.0/16 network. Autogenerated dial peers for registrations that match pool 12 have attributes configured in this pool.
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.
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voice register pool 12
id network 172.16.0.0 mask 255.255.0.0
number 1 50.. preference 2
application SIP.app
preference 2
incoming called-number
cor incoming allowall default
translate-outgoing called 1
voice-class codec 1
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To help you troubleshoot a SIP registrar and voice register pool, perform the following steps.
Command or Action | Purpose | |
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Step 1 |
debug voice register errors Example:
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Use this command to debug errors that happen during registration. If there are no voice register pools configured for a particular registration request, the message Contact doesn’t match any pools is displayed. |
Step 2 |
debug voice register events Example:
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Using the debug voice register events command should suffice to display 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. The phone number 91011 registered successfully, and type 1 is reported, which means there is a pre-existing VoIP dial peer. |
Step 3 |
show sip-ua status registrar Example:
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Use this command to display all the SIP endpoints currently registered with the contact address. |
To use the icmp-ping keyword with the proxy command to assist in troubleshooting proxy dial peers, perform the following steps.
Command or Action | Purpose | |||
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Step 1 |
configure terminal Example:
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Use this command to enter global configuration mode. |
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Step 2 |
voice register pool Example:
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Use this command to enter voice register pool configuration mode. |
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Step 3 |
proxy ip-address[preferencevalue] [monitor probe {icmp-ping|rtr}[alternate-ip-address]] Example:
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Set the proxy command to monitor with icmp-ping . |
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Step 4 |
end Example:
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Returns to privileged EXEC mode. |
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Step 5 |
show voice register dial-peers Example:
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Use this command to verify dial-peer configurations, and notice that icmp-ping monitoring is set. |
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Step 6 |
show dial-peer voice Example:
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Use the show dial-peer voice command on dial peer 40036, and notice the monitor probe status.
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The next step is configuring incoming and outgoing calls for Cisco Unified SRST. For more information, see the Configuring Call Handling section.
Internet Protocol version 6 (IPv6) is the latest version of the Internet Protocol (IP). IPv6 uses packets to exchange data, voice, and video traffic over digital networks. Also, IPv6 increases the number of network address bits from 32 bits in IPv4 to 128 bits. From Unified SRST Release 12.0 onwards, Unified SRST supports IPv6 protocols for SIP IP phones.
IPv6 support in Unified SRST allows the network to behave transparently in a dual-stack (IPv4 and IPv6) environment and provides additional IP address space to SIP IP phones that are connected to the network. If you do not have a dual-stack configuration, configure the CLI command call service stop under voice service voip configuration mode before changing to dual-stack mode. For an example of switching to dual-stack mode, see Examples for Configuring IPv6 Pools for SIP IP Phones.
The Cisco IP Phone 7800 Series and 8800 Series are supported on IPv6 for Unified SRST.
For more information on configuring SIP IP phones for IPv6 source address, see Configure IPv6 Pools for SIP IP Phones.
For an example of configuring IPv6 Support on Unified SRST, see Examples for Configuring IPv6 Pools for SIP IP Phones.
For more details about IPv6 deployment, see IPv6 Deployment Guide for Cisco Collaboration Systems Release 12.0.
The basic feature supported for a IPv6 WAN down scenario is:
Basic SIP Line (IPv4 or IPv6) to SIP Line calls (IPv4 or IPv6) when Unified SRST is in dual-stack no anat mode.
The following supplementary services are supported as part of IPv6 in Unified SRST IP Phones:
Hold/Resume
Call Forward
Call Transfer
Three-way Conference (with BIB conferencing only)
Line to T1/E1 Trunk and Trunk to Line with Supplementary Service Features
Fax to and from PSTN (IPv4 ATA to ISDN T1/E1) for both T.38 Fax Relay and Fax Passthrough
The following are the known restrictions for IPv6 support on Unified SRST:
SIP Trunks are not supported on Unified SRST for IPv6 deployment. PSTN calls are supported only through T1/E1 trunks.
SCCP IP Phones are not supported in a deployment of IPv6 for Unified SRST.
SIP Phones can be either in IPv4 only or IPv6 only mode (no anat ).
Trancoding and Transrating are not supported.
H.323 trunks are not supported.
Secure SIP lines or trunks are not supported.
IPv6 on Unified SRST is not supported on the Cisco IOS platform. The support is restricted to Cisco IOS XE platform with Cisco IOS Release 16.6.1 or later versions.
For IPv6 Support on Unified SRST, all the legacy IP Phones and Voice Gateways must be converted or reconfigured to IPv4-Only SIP signaling from SCCP signaling, if applicable.
Unified SRST 12.0 or a later version.
IPv6 option only appears if protocol mode is dual-stack configured under sip-ua configuration mode or IPv6.
Cisco Unified SRST License must be configured for the gateway to function as a Unified SRST gateway to support IPv6 functionality. For more information on licenses, see Licensing.
Cisco Unified Communications Manager (Unified Communications Manager) is provisioned with the IPv6 address of Unified SRST. For information on configuration of Unified SRST on Unified Communications Manager, see Survivable Remote Site Telephony Configurationin Cisco Unified Communications Manager Administration Guide.
Command or Action | Purpose | |
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Step 1 |
enable Example:
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Enables privileged EXEC mode.
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Step 2 |
configure terminal Example:
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Enters global configuration mode. |
Step 3 |
ipv6 unicast-routing Example:
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Enables the forwarding of IPv6 unicast datagrams. |
Step 4 |
voice service voip Example:
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Enters voice-service configuration mode to specify a voice encapsulation type.
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Step 5 |
sip Example:
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Enters SIP configuration mode. |
Step 6 |
no anat Example:
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Disables Alternative Network Address Types (ANAT) on a SIP trunk. |
Step 7 |
call service stop Example:
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Shuts down SIP call service. |
Step 8 |
exit Example:
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Exits SIP configuration mode. |
Step 9 |
exit Example:
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Exits voice service voip configuration mode. |
Step 10 |
sip-ua Example:
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Enters SIP user-agent configuration mode. |
Step 11 |
protocol mode{ ipv4| ipv6| dual-stack[ preference{ ipv4| ipv6} ] } Example:
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Allows phones to interact with phones on IPv6 voice gateways. You can configure phones for IPv4 addresses, IPv6 address es, or for a dual-stack mode.
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Step 12 |
exit Example:
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Exits SIP configuration mode. |
Step 13 |
voice service{ voip} Example:
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Enters voice-service configuration mode to specify a voice encapsulation type.
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Step 14 |
sip Example:
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Enters SIP configuration mode. |
Step 15 |
no call service stop Example:
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Activates SIP call service. |
Step 16 |
exit Example:
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Exits SIP configuration mode. |
Step 17 |
voice register global Example:
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Enters voice register global configuration mode to set parameters for all supported SIP phones in Cisco Unified CME. |
Step 18 |
default mode Example:
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Enables mode for provisioning SIP phones in Unified SRST. The default mode is Unified SRST itself. |
Step 19 |
max-dnmax-directory-numbers Example:
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Limits number of directory numbers to be supported by this router. Maximum number is platform and version-specific. Type ? for value. |
Step 20 |
max-poolmax-voice-register-pools Example:
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Sets maximum number of SIP phones to be supported by the Unified SRST router. |
Step 21 |
exit Example:
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Exits voice register global configuration mode. |
Step 22 |
voice register poolpool-tag Example:
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Enters voice register pool configuration mode to set phone-specific parameters for a SIP phone. |
Step 23 |
id{ networkaddressmaskmask| ip address maskmask| macaddress} Example:
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Explicitly identifies a locally available individual SIP phone to support a degree of authentication. |
Step 24 |
end Example:
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Exits to privileged EXEC mode. |
The following example provides configuration of IPv6 pools for SIP IP Phones:
ipv6 unicast-routing
voice service voip
sip
no anat
call service stop
exit
exit
sip-ua
protocol mode dual-stack
exit
voice service voip
sip
no call service stop
exit
voice register global
default mode
max-dn 50
max-pool 40
exit
voice register pool 1
id network 2001:420:54FF:13::901:0/117
end
The following example provides interface configuration for IPv6 supported on Unified SRST:
configure terminal
interface GigabitEthernet0/0/1
ip address 10.64.86.229 255.255.255.0
negotiation auto
ipv6 address 2001:420:54FF:13::312:82/119
ipv6 enable
The following example provides IP route configuration for IPv6 supported on Unified SRST:
ipv6 route 2001:420:54FF:13::312:0/119 2001:420:54FF:13::312:1
ipv6 route 2001:420:54FF:13::901:0/119 2001:420:54FF:13::312:1
The following example displays output when SIP call service is shut down with the call service stop CLI command:
Router# show sip service
SIP service is shut
under voice service voip, sip submode
The following example displays output when SIP call service is active with the no call service stop CLI command:
Router# show sip-ua service
SIP Service is up
under voice service voip, sip submode