[asterisk-dev] 1.6.2.0-beta4 - SIP TCP or TLS - Ringing/OK ignored

[Kristijan Vrban]

should be fixed in 1.6.2 svn branch. see:
https://issues.asterisk.org/view.php?id=13865

Kristijan Vrban

2009/8/20 Stefan Tichy <asterisk2 at pi4tel.de>

>
> Asterisk 1.6.2.0-beta4 has udp and tcp enabled for SIP calls.
> If the phone snom360-SIP 7.3.7 uses udp everything seems to work,
> but if I change this to tcp incoming calls do fail. No problem with
> outgoing calls or registration.
>
> Asterisk does send INVITE, ignores 180 Ringing and 200 OK but
> observes BYE at the end of the dialog.
>
> I don't see that there is anything wrong with the first two responses.
>
> Same problem if tls is used instead of tcp.
>
>
> Thanks in advance
>
> --
> Stefan Tichy  ( asterisk2 at pi4tel dot de )
>
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