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.
I just changed extensions.conf to read:
/etc/asterisk/extensions.conf [incoming] include => sip-phones exten => _5551212,1,Answer exten => _5551212,2,Dial(SIP/6710,12,tr) exten => _5551212,3,Dial(SIP/6710&SIP/6711&SIP/6712&SIP/6713,20,tr) exten => _5551212,4,Voicemail2(u6710) exten => _5551212,5,Hangup exten => _5551212,104,Voicemail2(b6710) exten => _5551212,105,Hangup
and restarted asterisk for good measure. I am still not getting any ring on the calling end. Anything else I should check?
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