Simone Ricci
2005-Nov-14 08:28 UTC
[Asterisk-Users] Problem with 827-4v and asterisk as a pstn GW
Hi, I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as sip-to-pstn GW. The issue is that when a call comes in from the pstn, asterisk correctly contacts the router, which in turns send a "183 Session progress". Obviously, asterisk thinks that the telephone is not ringing (because it expects a "180 Ringing") and we have no ringback on the pstn side. Putting a ringing() in the dialplan is not an option. Anyone has suggestions? Cheers, Simone.
FaberK
2006-Mar-15 00:11 UTC
[Asterisk-Users] Problem with 827-4v and asterisk as a pstn GW
Hi Simone, I'm trying to use a 827-4V with SER+Asterisk. My problems are that calling out, from 827, is fine, but on the other way do not works. The PSTN caller ear no ringing and when the 827 answers, there is no outbound voice. Do you reach some results? Best Regards. 2005/11/14, Simone Ricci <simone.ricci@cwnet.it>:> > Hi, > I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as > sip-to-pstn GW. The issue is that when a call comes in from the pstn, > asterisk correctly contacts the router, which in turns send a "183 > Session progress". Obviously, asterisk thinks that the telephone is not > ringing (because it expects a "180 Ringing") and we have no ringback on > the pstn side. Putting a ringing() in the dialplan is not an option. > > Anyone has suggestions? > > Cheers, > Simone. > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060315/3fd92485/attachment.htm