Yes. Add a Ringing command.
exten = _5551212,1,Answer
exten = _5551212,2,Ringing
exten = _5551212,3,Dial(SIP/6710,12,tr)
Ok, extensions.conf now contains:
[incoming]
include = sip-phones
exten = _5551212,1,Answer
exten = _5551212,2,Ringing
exten = _5551212,2,Dial(SIP/6710,12,tr)
... etc. and
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 =
