Reply to: [email protected]
Hello everyone. I'd appreciate a lot your help with this issue. I'm running a
very basic script of JS for subscribing my jsSIP User Agent to my local
Asterisk server and making a voice call. I don't get any warnings or errors
from the Asterisk CLI nor the script, but when I make a call to a legacy SIP
phone or SIP trunk well configured, there is no audio on any side although
there is ringing, calls can be answered and they never drop. My Asterisk 12 was
compiled with SRTP and pjproject.
I read at the Asterisk WebRTC
Wiki(https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support) this:
"Starting with Asterisk 12 you need to have pjproject libraries installed,
otherwise you most likely won't have audio in your WebRTC calls and no warning
whatsoever!"
I properly installed it and selected it for the Asterisk compilation, but I
wonder wether I did it wrong, and how can I check it ...
I leave here my Asterisk files: http://pastebin.com/p5euwnTJ
This is a SIP debuging of my jsSIP UA subscribing: http://pastebin.com/KxgB6GYb
This is a SIP debugging of a local call: http://pastebin.com/VQayVYAh
Finally this is what the CLI says about it: http://pastebin.com/9FXAUU6c
... Thanks in advance
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