Appendix A: Configuring Cisco Unified SIP SRST Features Using Redirect Mode
Information About Cisco Unified SIP SRST Features Using Redirect Mode
274
Cisco Unified SCCP and SIP SRST System Administrator Guide
OL-13143-04
Information About Cisco Unified SIP SRST Features Using
Redirect Mode
Cisco Unified SIP SRST provides backup to an external SIP proxy server by providing basic registrar
and redirect 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.
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.
•
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 Cisco Unified SIP SRST Features Using
Redirect Mode
This section contains the following procedures:
•
Configuring Call Redirect Enhancements to Support Calls Between SIP IP Phones for Cisco Unified
SIP SRST, page 274
(required)
•
Configuring Sending 300 Multiple Choice Support, page 277
(required)
Configuring Call Redirect Enhancements to Support Calls Between SIP IP
Phones for Cisco Unified SIP SRST
The call redirect enhancement supports calls from a local SIP phone to another local SIP phone through
the Cisco IOS voice gateway. Prior to 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.
•
Configuring Call Redirect Enhancements to Support Calls Globally, page 275
•
Configuring Call Redirect Enhancements to Support Calls on a Specific VoIP Dial Peer, page 276