Greeting,
I’m trying to move old asterisk based proxy from chan_sip to pjsip.
I ran into issue with audio. When direct media is enabled I get one way audio, when I disable direct media I get no audio.
I ran wireshark on asterisk proxy with direct audio disabled I can see rtp flow between sip trunk and asterisk, and asterisk and PBX; but no sounds is coming through.
There is no firewall between a sip trunk and a client PBX. Everything is on public IP’s.
Any help is appreciated.
Thanks!
-Greg
A link to a PBX would be what people call a SIP trunk (although SIP doesn’t have trunks and extensions). You are starting by assuming we know more about your, not completely run of the mill, system, than is reasonable to assume.
Please start by describing the network configuration, in some detail, then providing the working chan_sip configuration, the broken chan_pjsip one and verbose level 3 or more full logs. I think the logs will also need pjsip set logger on setting.
In terms of the sound going in and out, you have no end nodes in your description, so where are you expecting to originate the sound and where to listen to it?
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