[cisco-voip] TIP: CNAM without requiring MGCP

[Zilk, David]

It was a PRI from the telco, but a SIP trunk between the gateway and
Call Manager.

 

David

 

________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com] 
Sent: Saturday, October 14, 2006 7:19 AM
To: Zilk, David; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] TIP: CNAM without requiring MGCP

 

Whom was the telco providing the SIP trunk?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Zilk, David
Sent: Friday, October 13, 2006 8:20 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] TIP: CNAM without requiring MGCP

 

Many of you have run across the problem of not being able to utilize
Caller ID Name (CNAM) information from a telco using H.323.  Since CNAM
is provided by most telcos  within the facility IE field, and this isn't
supported in H.323, the solution has always been to convert to MGCP.
If you needed to stick with H.323 or gateway based call routing for
other reasons, such as needing to support other H.323 devices, or use
TCL scripts, you were out of luck.

 

In one of those "why didn't I think of that before" moments, one of my
colleagues thought to test connectivity using an incoming SIP dial peer
and SIP trunk on call manager.  Lo and behold, CNAM works perfectly!
H.323 dial peers can still be used as needed to match other inbound and
outbound traffic, and since the routing decisions are being done at the
gateway, TCL scripts can still be used.

 

A few SIP specific parameters are required on the gateway including:

 

signaling forward unconditional

sip

  bind all source-interface <interface>

!

!

sip-ua 

  timers buffer-invite 5000

 

As well as adding " session protocol sipv2 " to the incoming voip
dial-peer.

 

 

Configuration on Call Manager includes adding an RFC 2833 DTMF-compliant
MTP resource and setting up a SIP Trunk to the gateway.

 

 

While we have not fully tested this configuration, it appears to solve
the problem of not being able to receive caller-id name information
while still maintaining gateway based call routing. 

 

Can anyone point to any problems we might encounter with this setup?

 

 

 

David Zilk

 

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