On 02/14/2012 11:19 AM, Karsten Wemheuer wrote:
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@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@192.168.10.72>;tag=as3cafd135.
To:<sip:arthur@192.168.10.72>.
Contact:<sip:245@192.168.10.72:25060>.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@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@192.168.10.72>;tag=as3cafd135.
To:<sip:arthur@192.168.10.72>.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
Proxy => phone
U 192.168.10.72:5060 -> 192.168.10.221:34381
INVITE sip:arthur@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@192.168.10.72>;tag=as3cafd135.
To:<sip:arthur@192.168.10.72>.
Contact:<sip:245@192.168.10.72:25060>.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@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@192.168.10.72>;tag=as3cafd135.
To:<sip:arthur@192.168.10.72>;tag=cvovqkf6i5.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
Contact:<sip:arthur@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@192.168.10.72>;tag=as3cafd135.
To:<sip:arthur@192.168.10.72>;tag=cvovqkf6i5.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
Contact:<sip:arthur@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@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@192.168.10.72>;tag=as3cafd135.
To:<sip:arthur@192.168.10.72>.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@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@192.168.10.72>;tag=as3cafd135.
To:<sip:arthur@192.168.10.72>.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@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)?
This does appear to be a bug in Asterisk; please open an issue in JIRA,
and post the issue number here, so we can get someone looking at this
ASAP. Thanks!
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
--
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