[asterisk-dev] Sip is no longer working in 1.4 trunk

[Loic DIDELOT]

Hello,
I am following the development of the 1.4 branch daily. 

I notice that SIP has been crashing for the first time without a full
asterisk crash. Something we had never had. IAX hat lots of problems in
1.2 and 1.4 but SIP always worked fine.

But now since several revision SIP does no longer work. I can register
with my clients. That works fine. But launching a call does not work. I
see nothing on the cli or debug log message. All I see is this:

[Jan 10 10:06:57] NOTICE[21938] chan_sip.c: Unable to create/find SIP
channel for this INVITE

I haven't change anything on my configuration.

Any idea?

I tracked it down and the last working revision is 50098. 

Best regards,
Loic Didelot.

-- 
Loic DIDELOT (CTO)
voipGATE S.A.
Tel: +352 20 200 223
Fax: +352 20 200 923
E-mail: ldidelot at voipgate.com
Web: http://www.voipgate.com