Dears
Asterisk is a median server between the caller and the terminations gateway
The caller send the call to asterisk à asterisk will play music on hold
untill the termination gateway send 200 OK and the RTP establish
My problem that, Asterisk is not forwarding the 180 ringing from the
termination gateway to the user
How can I forward the sip message 180 to the caller or let the music on hold
stop playing and the caller hears the Ring Back Tone which when 180
ringing from the termination gateway.
What I am using now to stop the musinc on hold when the RTP established is
exten => _X.,n,Dial(SIP/Temination_Gateway/${EXTEN}|300|m)
NB: I tried to edit chan_sip.c but I did not find the solution
Please Advice
Regards
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