asterisk users - Aug 2011 - Customizing sip response codes for PBX Sip trunk

Say the PBX is:
Mitel-3300-ICP 10.2.0.26_2
Created SIP trunk to * but PBX doesn't see trunk as unavailable when
its really unavailable. It simply fails the calls..
How can I change the SIP response code to respond with e.g. "All Channels
busy"?
Any suggestions on how to program the Mitel to work?
Tx Shaun

Add qualify=yes to the peer (aka "trunk")  This is not about SIP
response codes.
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Shaun Wingrin
> Sent: Thursday, August 04, 2011 4:21 AM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Customizing sip response codes for PBX Sip trunk
> 
> Say the PBX is:
> Mitel-3300-ICP 10.2.0.26_2
> Created SIP trunk to * but PBX doesn't see trunk as unavailable when
its
> really unavailable. It simply fails the calls..
> How can I change the SIP response code to respond with e.g. "All
Channels
> busy"?
> Any suggestions on how to program the Mitel to work?
> Tx Shaun
> 
> --
> ___________________________________________________________________
> __
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

On 11-08-04 04:21 AM, Shaun Wingrin wrote:
> Say the PBX is:
> Mitel-3300-ICP 10.2.0.26_2
> Created SIP trunk to * but PBX doesn't see trunk as unavailable when
> its really unavailable. It simply fails the calls..
> How can I change the SIP response code to respond with e.g. "All
Channels busy"?
> Any suggestions on how to program the Mitel to work?
 >
I have not done a deployment in the last year, but before that I've done 
multiple installation with the 3300 and ever ran into this issue.

Is this on the same LAN, I assume there is no routing issue?

A SIP debug log from Asterisk would be helpful.

But to answer your question, there is no way to change SIP response 
codes via the dialplan, you would have to modify the source for 
chan_sip.c (something I do not recommend).

-- 
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org