Thank you Gareth,
It worked like a charm.
The only problem I am having is now, when I do some changes in my
freepbx and reload it just rewrites my dial play , I will try to fix it
though.
Thanks again
Cheers
Shantanu
On 09/25/2013 05:09 PM, Gareth Blades wrote:
On 25/09/13 11:21, Kumar Shantanu wrote:
Thanks Gareth ,
Try calling Progress() just before the dial command. Without this
Asterisk wont send the SIP/183 Session Progress and send the inband
audio until the call is answered.
Do I need to change something in asterisk dial plan ? I am using
freepbx to mange asterisk graphically.
Yes you will see a section called [macro-dialout-trunk]
Within that there will be a line line :-
exten => s,n,Dial(something....)
Just before that line add :-
exten => s,n,Progress()
You will then need to reload the dialplan (dialplan reload from the
asterisk prompt) and you can give it a go.
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