[asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2

[Kevin P. Fleming]

On 02/14/2012 09:30 AM, Karsten Wemheuer wrote:
> Hi,
>
> I got a problem with asterisk 1.8.9.2. The same scenario is working fine
> in 1.8.8.2.
>
> Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
> the same LAN, no NAT.
>
> Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the
> phone. The phone sends 180 RINGING back to the proxy. The proxy sends
> 180 RINGING to asterisk. So far so good. If the calling side decides to
> cancel the call, asterisk sends the CANCEL directly to the phone. The
> phone doesn't find the call and answers 404. In asterisk 1.8.8.2
> asterisk sends the CANCEL to the proxy, which sends the CANCEL to the
> phone and all ist fine.
>
> I think, the new behavior comes from the lines
>   		parse_ok_contact(p, req);
> 		if (!reinvite) {
> 			build_route(p, req, 1);
> 		}
> which are inserted in the handling of provisional SIP response.
>
> Am I doing something wrong or is this a bug?

It's impossible to answer that question without seeing the SIP 
signaling. The answer will depend on what the proxy did to insert itself 
in the path (or not) when it forwarded the 180 RINGING response to Asterisk.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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