Steve Totaro a écrit : > On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel < > [email protected]> wrote: > >> Hello. >> >> I've set up and configured an Asterisk server to make SIP phone calls to >> external classic phones. >> >> However, it happens that after 15 or 30 seconds, the phone call drops. >> The SIP session still seems valid, but no sound comes through any more. >> >> How would you go through to troubleshoot this issue? >> >> All the best, >> >> Guillaume Yziquel. > > Make sure you have canreinvite set to no.
It was already set to 'no' > Also, you may need to put an answer() in before your dial, I have dealt with > that strangeness, call always drop at exactly 30 seconds. Putting exten => _X.,n,Answer() in the dialplan doesn't change anything. > That solution worked for me, but I could see how it could mess up CDRs and > billing for some applications. Maybe I'm having a different issue than you've been experiencing. What's rather painful is that nothing appears to show in the Asterisk CLI when this happens since it's obviously not a problem with the SIP connection. How could I monitor the voice going in and out? All the best, Guillaume Yziquel. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
