Hi Michel, can you post also the SIP trace of the call ? So far, from the log I can see the call is established (I see 200OK and ACK) and then I see a BYE... but the trace will be more helpful.
Regards, Bogdan michel freiha wrote: > Hi all, > > I have an opensips server installed on my network and used for > registration on local call...When a customer dial a PSTN call, it'll > be routed to an asterisk server that route it to a PSTN gateway...Tis > scenario is working smoothly... > > The problem occurs when receiving a call from asterisk...The call is > sent from asterisk to an online endpoint on Opensips...The extension > is ringing but as soon as I open accept the call on the extension > registered on opensips, the call is hanged up direcly... > > I checked logs and found out that asterisk send INVITE packets to > opensips and OpenSip replies by <Call/Transaction Does Not Exist> > > Please chek the opensips log at http://pastebin.com/d27ae4ee9 > > Thanks for help > > Regards > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
