Prerequisites for Cisco Unified SIP SRST Features Using Redirect Mode
Complete the prerequisites documented in the Cisco Unified SRST Feature Overview chapter.
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This chapter describes Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) features using redirect mode.
Note |
This chapter applies to version 3.0 only. |
Complete the prerequisites documented in the Cisco Unified SRST Feature Overview chapter.
See the restrictions documented 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 redirect services. These services are used by a SIP IP phone if 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.
To make maximum use of the Cisco Unified SIP SRST service, the local SIP IP phones should support dual (concurrent) registration with both their primary SIP proxy or registrar and the Cisco Unified SIP SRST backup registrar. 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.
Other 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 Cisco Unified SIP SRST Features Using Redirect Mode
The call redirect enhancement supports calls from a local SIP phone to another local SIP phone through the Cisco IOS voice gateway. Before this enhancement, an attempt by a SIP phone to contact another local SIP phone using the Cisco IOS voice gateway as if it were a SIP proxy or redirect server would fail. However, the Cisco IOS voice gateway can now act as a SIP redirect server. The voice gateway responds to the originator with a SIP Redirect message, allowing the SIP phone that originated the call to establish a call to its destination.
The redirect ip2ip (voice service) and redirect ip2ip (dial-peer) commands allow you to enable the SIP functionality, globally or on a specific inbound dial peer. The default application on Cisco Unified SIP SRST supports IP-to-IP redirection.
Note |
When IP-to-IP redirection is configured in dial-peer configuration mode, the configuration for the specific dial peer takes precedence over the global configuration entered under voice service configuration mode. |
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 service voip Example:
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Enters voice service configuration mode. |
Step 4 |
redirect ip2ip Example:
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Configures a video codec at the dial peer level. Redirects SIP phone calls to SIP phone calls globally on a gateway using the Cisco IOS voice gateway. |
Step 5 |
end Example:
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Returns to privileged EXEC mode. |
To enable IP-to-IP call redirection for a specific VoIP dial peer, configure it on an inbound dial peer in dial-peer configuration mode. The default application on Cisco Unified SIP SRST supports IP-to-IP redirection.
Note |
When IP-to-IP redirection is configured in dial-peer configuration mode, the configuration for the specific dial peer takes precedence over the global configuration entered under voice service configuration mode. |
Command or Action | Purpose | |
---|---|---|
Step 1 |
enable Example:
|
Enables privileged EXEC mode.
|
Step 2 |
configure terminal Example:
|
Enters global configuration mode. |
Step 3 |
dial-peer voice tag voip Example:
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Enters dial-peer configuration mode.
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Step 4 |
application application-name Example:
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Enables a specific application on a dial peer.
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Step 5 |
redirect ip2ip Example:
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Redirects SIP phone calls to SIP phone calls on a specific VoIP dial peer using the Cisco IOS voice gateway. |
Step 6 |
end Example:
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Returns to privileged EXEC mode. |
Before Cisco IOS Release 12.2(15)ZJ, when a call was redirected, the SIP gateway would send a 302 Moved Temporarily message. The first longest match route on a gateway (dial-peer destination pattern) was used in the Contact header of the 302 message. With Release 12.2(15)ZJ, if multiple routes to a destination exist for a redirected number (multiple dial peers are matched), the SIP gateway sends a 300 Multiple Choice message, and the multiple routes in the Contact header are listed.
The configuration below allows users to choose the order in which the routes appear in the Contact header.
Command or Action | Purpose | |
---|---|---|
Step 1 |
enable Example:
|
Enables privileged EXEC mode.
|
Step 2 |
configure terminal Example:
|
Enters global configuration mode. |
Step 3 |
voice service voip Example:
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Enters voice service configuration mode. |
Step 4 |
sip Example:
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Enters SIP configuration mode. |
Step 5 |
redirect contact order [best-match | longestmatch] Example:
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Sets the order of contacts in the 300 Multiple Choice message. The keywords are defined as follows:
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Step 6 |
end Example:
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Returns to privileged EXEC mode. |
This section provides the following configuration example:
This section provides a configuration example to match the configuration tasks in the previous sections.
!
! Sets up the registrar server and enables IP-to-IP redirection and 300
! Multiple Choice support.
!
voice service voip
redirect ip2ip
sip
registrar server expires max 600 min 60
redirect contact order best-match
!
! Configures the voice-class codec with G.711uLaw and G729 codecs. The codecs are
! applied to the voice register pools.
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729br8
!
! The voice register pools define various pools that are used to match
! incoming REGISTER requests and create corresponding dial peers.
!
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
!
voice register pool 2
id ip 192.168.0.3 mask 255.255.255.255
preference 5
cor outgoing call95 1 91021
proxy 10.2.161.187 preference 1
voice-class codec 1
!
voice register pool 3
id network 10.2.161.0 mask 255.255.255.0
number 1 95... preference 1
preference 5
cor incoming call95 1 95011
cor outgoing call95 1 95011
proxy 10.2.161.187 preference 1 monitor probe icmp-ping
max registrations 5
voice-class codec 1
!
voice register pool 4
id network 10.2.161.0 mask 255.255.255.0
number 1 94... preference 1
preference 5
cor incoming everywhere default
cor outgoing everywhere default
proxy 10.2.161.187 preference 1
max registrations 2
voice-class codec 1
!
! Configures translation rules to be applied in the voice register pools.
!
translation-rule 1
Rule 0 94 91
!
! Sets up proxy monitoring.
!
call fallback active
!
dial-peer cor custom
name 95
name 94
name 91
!
! Configures COR values to be applied to the voice register pool.
!
dial-peer cor list call95
member 95
!
dial-peer cor list call94
member 94
!
dial-peer cor list call91
member 91
!
dial-peer cor list everywhere
member 95
member 94
member 91
!
! Configures a voice port and a POTS dial peer for calls to and from the PSTN endpoints.
voice-port 1/0/0
!
dial-peer voice 91500 pots
corlist incoming call91
corlist outgoing call91
destination-pattern 91500
port 1/0/0
!