[cisco-voip] TIP: CNAM without requiring MGCP

[Fred Nielsen]

Sounded like he was referring to a telco provided ISDN PRI terminating on a
Cisco VoIP gateway router, with SIP running between the customer owned
gateway and the CallManager.
 
TELCO <--PRI--> VOIP GW <--SIP--> CM

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From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons (US)
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?

 

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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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