You need to answer the line to place audio on the channel. So if you place an answer line before the dials, you should get audio to route back.
On Fri, 2003-12-26 at 23:46, Terry Wilson wrote: > We just switched from three x100p's to a te410p for handling our > incoming/outbound calls. Everything works great, except incoming > callers don't hear a ring while they are waiting for one of us to pick > up. The phones themselves ring fine, but the caller doesn't hear > anything until someone picks up, or it transfers to voicemail. Any > clues as to what may be happening? > > /etc/zaptel.conf > span=1,1,0,esf,b8zs > bchan=1-23 > dchan=24 > defaultzone=us > > /etc/asterisk/zapata.conf > [channels] > usecallerid=yes > adsi=yes > callwaiting=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > echocancel=yes > echocancelwhenbridged=no > echotraining=yes > rxgain=1.5 > txgain=1.5 > immediate=no > switchtype=national > context=incoming > signalling=pri_cpe > group=1 > channel => 1-23 > > /etc/asterisk/extensions.conf > [incoming] > include => sip-phones > exten => _5551212,1,Dial(SIP/6710,12,tr) > exten => _5551212,2,Dial(SIP/6710&SIP/6711&SIP/6712&SIP/6713,20,tr) > exten => _5551212,3,Voicemail2(u6710) > exten => _5551212,4,Hangup > exten => _5551212,103,Voicemail2(b6710) > exten => _5551212,104,Hangup > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
