Try removing the Answer() before the Dial... e.g.: [spa2100]
exten => _X.,1,NoOp(SIP Call from SPA2100 to ${EXTEN}) exten => _X.,2,Dial(SIP/netvoice-102) exten => _X.,3,Hangup Regards Julian J. M. On 12/9/05, George Pajari <[EMAIL PROTECTED]> wrote: > Eric "ManxPower" Wieling wrote: > > > T/t/H/h and other options to Dial require Asterisk to stay in the RTP > > stream. > > Understood but already checked as not being the cause. Thanks for the > suggestion, though. > > Here is our entire extensions.conf context: > > [spa2100] > > exten => _X.,1,NoOp(SIP Call from SPA2100 to ${EXTEN}) > exten => _X.,2,Answer > exten => _X.,3,Wait(2) > exten => _X.,4,Dial(SIP/netvoice-102) > exten => _X.,5,Hangup > > where > > [netvoice-102] > accountcode=netvoice-102 > callerid=NETVOICE COMMS <604 484 8647> > username=netvoice-102 > type=friend > host=dynamic > dtmfmode=rfc2833 > nat=no > qualify=no > mailbox=102 > context = netvoice-internal > canreinvite=yes > disallow=all > allow=ulaw > > Here is a "sip show channels" during a call: > > aa.bb.cc.39 netvoice-1 7f6a484c36f 00103/00000 ulaw > aa.bb.cc.40 nvc.test.a 6cfe5077-2f 00103/00102 ulaw > > -- > George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) > Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) > www.netvoice.ca www.ip-centrex.ca > www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca > > _______________________________________________ > --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 > _______________________________________________ --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