Hi,
I am integrating OpenSIPS and Asterisk to use Asterisk to play media (typical
media treatment)
I have a softphone registered to OpenSIPS and when i call a specific number, a
simple prompt needs to be played from asterisk. I have the sip configuration
and also extensions.conf file setup.
When i call the specific number, the SIP messages are exchanged but the call
drops stating calling number not found (i have the number configured in
asterisk though). In OpenSIPS.cfg all i am doing is just calling the function
sethostport("<Asterisk IP>:5060") when receiving the call at this number
If i register the endpoints directly with Asterisk, i can hear the announcement
as expected.
Not sure if i am missing something or is there anything that needs to be set
specifically in OpenSIPS for this to work?
Thank you,DK_______________________________________________
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