I think this has to do with ACK handling. One can probably argue that this
is a bug in the config, but due to a complete config change in 2.0, it was
never fixed.
If I recall correctly, there is a missing relaying of acks without route
headers. I'm on mobile, so I cannot check.
g-)
------- Original message -------
From: Stefan Sayer <stefan.sayer(a)iptego.com>
Cc: serusers(a)lists.iptel.org
Sent: 6.2.'08, 20:59
Hello,
Frank Durda IV wrote:
Thanks for the catch!
Now with 192.168.200.30 re-added to the domain table, things get further
and a test call claims "Connected" at the SIP phone display for exactly
30 seconds before the SIP phone reports that the Call Ended.
Based on the logs it does not appear that SER attempted to contact the
PSTN switch, but SER certainly got closer.
to me it looks like the INVITE is sent
out and retried. an ngrep would
definitely be more revealing about what happens here (ngrep port 5060 -W
byline -d any).
Stefan
--
Stefan Sayer
VoIP Services
stefan.sayer(a)iptego.com
www.iptego.com
iptego GmbH
Am Borsigturm 40
13507 Berlin
Germany
Amtsgericht Charlottenburg, HRB 101010
Geschaeftsfuehrer: Alexander Hoffmann
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