Prerequisites for Configuring the SIP Registrar
Complete the prerequisites documented in the “Prerequisites for Configuring Cisco Unified SIP SRST” section on page 9 section in “Cisco Unified SRST Feature Overview” section on page 1.
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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 on page 9 section in “Cisco Unified SRST Feature Overview” section on page 1.
See the restrictions documented in the “Restrictions for Configuring Cisco Unified SIP SRST” section on page 10 section in “Cisco Unified SRST Feature Overview” section on page 1.
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
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
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 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 .
Configuration of the required commands as described in the .
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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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,