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

[Karsten Wemheuer]

Hi Kevin,
Am Dienstag, den 14.02.2012, 09:46 -0600 schrieb 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.
> 

I shorten the trace to (hopefully) the relevant things. Asterisk is on
192.168.10.72, port 25060, proxy is opnesips on the same machine with
port 5060, the phone which is ringing is on 192.168.10.221.

Asterisk => Proxy:
U 192.168.10.72:25060 -> 192.168.10.72:5060
INVITE sip:arthur at 192.168.10.72 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
Max-Forwards: 70.
From: "Max M..ller" <sip:245 at 192.168.10.72>;tag=as3cafd135.
To: <sip:arthur at 192.168.10.72>.
Contact: <sip:245 at 192.168.10.72:25060>.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce at 192.168.10.72.
CSeq: 102 INVITE.
... sdp cut of ...

Proxy => Asterisk
U 192.168.10.72:5060 -> 192.168.10.72:25060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: "Max M..ller" <sip:245 at 192.168.10.72>;tag=as3cafd135.
To: <sip:arthur at 192.168.10.72>.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce at 192.168.10.72.
CSeq: 102 INVITE.

Proxy => phone
U 192.168.10.72:5060 -> 192.168.10.221:34381
INVITE sip:arthur at 192.168.10.221:34381;line=478vzxb3 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72;branch=z9hG4bK24be.5163d992.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
Max-Forwards: 69.
From: "Max M..ller" <sip:245 at 192.168.10.72>;tag=as3cafd135.
To: <sip:arthur at 192.168.10.72>.
Contact: <sip:245 at 192.168.10.72:25060>.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce at 192.168.10.72.
CSeq: 102 INVITE.
... sdp cut of ...

Phone => Proxy
U 192.168.10.221:34381 -> 192.168.10.72:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.10.72;branch=z9hG4bK24be.5163d992.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: "Max M..ller" <sip:245 at 192.168.10.72>;tag=as3cafd135.
To: <sip:arthur at 192.168.10.72>;tag=cvovqkf6i5.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce at 192.168.10.72.
CSeq: 102 INVITE.
Contact: <sip:arthur at 192.168.10.221:34381;line=478vzxb3>;reg-id=1.

Proxy => Asterisk
U 192.168.10.72:5060 -> 192.168.10.72:25060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: "Max M..ller" <sip:245 at 192.168.10.72>;tag=as3cafd135.
To: <sip:arthur at 192.168.10.72>;tag=cvovqkf6i5.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce at 192.168.10.72.
CSeq: 102 INVITE.
Contact: <sip:arthur at 192.168.10.221:34381;line=478vzxb3>;reg-id=1.

When canceling the call, asterisk sends

Asterisk => Phone
U 192.168.10.72:25060 -> 192.168.10.221:34381
CANCEL sip:arthur at 192.168.10.72 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
Max-Forwards: 70.
From: "Max M..ller" <sip:245 at 192.168.10.72>;tag=as3cafd135.
To: <sip:arthur at 192.168.10.72>.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce at 192.168.10.72.
CSeq: 102 CANCEL.

The Phone responds:

U 192.168.10.221:34381 -> 192.168.10.72:25060
SIP/2.0 404 Not found.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: "Max M..ller" <sip:245 at 192.168.10.72>;tag=as3cafd135.
To: <sip:arthur at 192.168.10.72>.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce at 192.168.10.72.
CSeq: 102 CANCEL.

As noted in the earlier mail, this scenario is working in previous
versions (1,4.x up to asterisk 1.8.8.2). 

Do You have any idea where the failure happens? Is it the proxy
configuration or is it at the asterisk side (maybe config or bug)?

Thanks,

Karsten