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? Thanks, M 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 > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users