[BUG??] Asterisk linphone call dropping by itself

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Well the problem seems to be:
the linphones are listening on port 5062, while * is on port 5060. For
some reason, the INVITEs are received from *, but are forwarded on port
5060 by default.

I “solved” the problem by moving * to port 5062 and moving the linphones
back to port 5060. All is well, but may this be a bug?


Il 03/11/2010 12:48, Matteo Fortini ha scritto:
> hi all, please help… I am calling in the simplest way among two
> linphone clients connected to one asterisk server… the call ends on
> one side without any sign of problem, while on the other side it stays
> connected.
> I checked the SIP dialogue and at some point the server sends a BYE
> message to one party
> I have no timeout set, though the duration of a call is always around 20s.
> the two linphones register with a name which is defined as dynamic in
> sip.conf
> the call terminates on the caller’s side, while the callee is still
> connected, and I have to force the termination on that side.
> I’m using asterisk 1.8.0 and linphone 3.99
> I really don’t know how to investigate further… a capture on sip ports
> just shows that on the 25th ack packet the other side answers with a BYE
> instead of with an OK SDP packet.
> TIA,
> Matteo