----- Original Message ----- From: asterisk [mailto:[EMAIL PROTECTED] To: [email protected] Sent: Mon, 31 Jul 2006 05:37:46 -0300 Subject: [asterisk-users] SIP channel problem
> Hi, > > We have a small callcenter. > We registered with a provider with SIP with username and pass. > If our customer start a call with an IP phone, he can speek. > BUT > If he hang Up the line, he get a callback. > with SIP debug I got the next message: > > --- (24 headers 9 lines)--- > "Using INVITE request as basis request - ....." > > > > > > Customer's PBX -> call -> Provider's PBX -> PSTN > If the customor give a Hang Up > > Provider's PBX -> call -> Customer's PBX > > I think. > > I hope this is the situation because I can't find problem in the dial > plan. :) > > Thx for help. > Kind regards > Szolke > A full sip debug of the dialog would be very helpful so we would know who is at fault and what's going on. Joshua Colp Digium _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
