Dear Bogdan, Thanks for help...I solved it...It was an asterisk issue
Regards On Wed, Feb 25, 2009 at 4:00 PM, Bogdan-Andrei Iancu <[email protected] > wrote: > 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 >> >> > >
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