should be fixed in 1.6.2 svn branch. see: <a href="https://issues.asterisk.org/view.php?id=13865">https://issues.asterisk.org/view.php?id=13865</a><br><br>Kristijan Vrban<br><br><div class="gmail_quote">2009/8/20 Stefan Tichy <span dir="ltr"><<a href="mailto:asterisk2@pi4tel.de">asterisk2@pi4tel.de</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
Asterisk 1.6.2.0-beta4 has udp and tcp enabled for SIP calls.<br>
If the phone snom360-SIP 7.3.7 uses udp everything seems to work,<br>
but if I change this to tcp incoming calls do fail. No problem with<br>
outgoing calls or registration.<br>
<br>
Asterisk does send INVITE, ignores 180 Ringing and 200 OK but<br>
observes BYE at the end of the dialog.<br>
<br>
I don't see that there is anything wrong with the first two responses.<br>
<br>
Same problem if tls is used instead of tcp.<br>
<br>
<br>
Thanks in advance<br>
<font color="#888888"><br>
--<br>
Stefan Tichy ( asterisk2 at pi4tel dot de )<br>
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