Configuring Sip-To-Sip Call Forwarding - Cisco CP-7911G-CH1 System Administrator Manual

How to Configure Cisco Unified SIP SRST
Command or Action
Step 9
no vad
Example:
Router(config-register-pool)# no vad
Step 10
codec codec-type [bytes]
Example:
Router(config-register-pool)# codec g729r8
Step 11
end
Example:
Router(config-register-pool)# end

Configuring SIP-to-SIP Call Forwarding

SIP-to-SIP call forwarding (call routing) is available. Call forwarding is provided either by the phone or
by using a back-to-back user agent (B2BUA), which allows call forwarding on any dial peer. Calls into
a SIP device may be forwarded to other SIP or SCCP devices (including Cisco Unity, third-party
voice-mail systems, or an auto attendant or IVR system such as IPCC and IPCC Express). In addition,
SCCP IP phones may be forwarded to SIP phones.
Cisco Unity or other voice messaging systems connected by a SIP trunk or SIP user agent are able to
pass a message-waiting indicator (MWI) when a message is left. The SIP phone then displays the MWI
when indicated by the voice messaging system.
SIP-to-H.323 call forwarding is not supported.
Note
To configure SIP-to-SIP call forwarding, you must first allow connections between specific types of
endpoints in a Cisco IP-to-IP gateway. The allow-connections command grants this capability. Once the
SIP-to-SIP connections are allowed, you can configure call forwarding under an individual SIP phone
pool. Any of the following commands can be used to configure call forwarding, according to your needs:
•
Cisco Unified SCCP and SIP SRST System Administrator Guide
166
Under voice register pool
call-forward b2bua all directory-number
–
call-forward b2bua busy directory-number
–
call-forward b2bua mailbox directory-number
–
call-forward b2bua noan directory-number [timeout seconds]
–
Purpose
Disables voice activity detection (VAD) on the VoIP dial
peer.
• VAD is enabled by default. Because there is no comfort
noise during periods of silence, the call may seem to be
disconnected. You may prefer to set no vad on the SIP
phone pool.
Specifies the codec supported by a single SIP phone or a
VoIP dial peer in a Cisco Unified SIP SRST environment.
The codec-type argument specifies the preferred codec and
can be one of the following:
• g711alaw: G.711 a–law 64,000 bps.
g711ulaw: G.711 mu–law 64,000 bps.
•
g729r8: G.729 8000 bps (default).
•
The bytes argument is optional and specifies the number of
bytes in the voice payload of each frame
Returns to privileged EXEC mode.
Configuring Call Handling
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