SIP Trunk Features
Call Forwarding over SIP Networks
Call forwarding over SIP networks uses the 302 Moved Temporarily SIP response, which
works in a manner similar to the way in which the H.450.3 standard is used for H.323
networks. To enable call forwarding, use the call-forward pattern command and specify
a pattern that matches the calling-party numbers of the calls that you want to be able to
forward. Use the call-forward pattern command with the .T pattern to allow all calls for
all possible SIP calling parties to be forwarded using the SIP 302 response.
Call Transfer over SIP Networks
Cisco Unified CME supports all SIP Refer method call transfer scenarios, but you must
ensure that call transfer is enabled using H.450.2 standards. Note that the transfer-
system command must be configured with the full-blind or full-consult keyword for SIP
Refer to be invoked.
DTMF Relay
SCCP phones used with Cisco Unified CME systems relay dual tone multifrequency
(DTMF) digits out of band. To interwork with SIP applications that expect in-band
DTMF digits, you must enable a conversion. Two types of conversions are possible:
- For remote SIP-based IVR or voice-mail application
- For Cisco Unity Express on a SIP network
SIP Register Support
SIP register support enables a SIP gateway to register E.164 numbers with a SIP proxy or
SIP registrar, similar to the way that H.323 gateways can register E.164 numbers with a
gatekeeper. SIP gateways allow registration of E.164 numbers to a SIP proxy or registrar
on behalf of analog telephone voice ports (FXS) and IP phone virtual voice ports (EFXS)
for local SCCP phones. This support is enabled using the register command in SIP UA
configuration mode.
When registering E.164 numbers in dial peers with an external registrar, you can also
register them with a secondary SIP proxy or registrar to provide redundancy. The
secondary registration can be used if the primary registrar fails.
Examples
Call Forwarding over SIP Networks:
Example
The following example enables call forwarding using the H.450.3 standard or SIP 302 response:
dial-peer voice 100 pots
destination-pattern 9.T
port 1/0/0
!
dial-peer voice 4000 voip
destination-pattern 4...
Call Transfer over SIP Networks:
Example
The following example specifies transfer with consultation using the H.450.2 standard for all IP phones serviced by the router:
dial-peer voice 100 pots
destination-pattern 9.T
port 1/0/0
!
dial-peer voice 4000 voip
destination-pattern 4...
session protocol sipv2
session-target ipv4:10.1.1.1
!
telephony-service
transfer-pattern 4...
transfer-system full-consult
DTMF Relay using RFC 2833:
Example
The following example specifies use of the RFC 2833 method for in-band DTMF relay for calls using dial peer 2.
dial-peer voice 2 voip
dtmf-relay rtp-nte
sip-ua
notify telephone-event max-duration 2000
DTMF Relay using SIP Notify:
Example
The following example specifies use of the SIP notify method for in-band DTMF relay for calls using dial peer 4.
dial-peer voice 4 voip
dtmf-relay sip-notify
sip-ua
notify telephone-event max-duration 2000
SIP Register Support:
Example
The following example sets up the gateway to register the gateway’s E.164 telephone numbers with an external SIP
registrar.
sip-ua
registrar ipv4:10.8.17.40 expires 3600 secondary
retry register 10
timers register 500
|
