I have created an application using AudioSocket of asterisk, when I am dialing an agent, I get the audio and it works fine. but when I am dialing an external number, I don’t get any audio.
When I call from agent to external number it works fine.
I don’t understand where the issue is. I am using Asterisk 18
can you check witch IP the SDP header is referring to (200 OK / INVITE)
is RTP through asterisk or send direct to AudioSocket
the INVITE is sent to the IP of my sip trunk.
I am not sure about this, “is RTP through asterisk or send direct to AudioSocket”
as I am using ami to initiate the call, The call connects to the audiosocket and I receive the uid of the call, which is the first packet after that I receive nothing
can you post the INVITE and the 200 OK as they contain a section that describe where to send the RTP
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