Does anyone know if there is a way to disable the norefersub for PJSIP?
It appears this is causing problems with a test we're running with Cisco.
A wireshark trace from a system where the transfer with Cisco works versus a
trace with Asterisk/Cisco shows one big difference being the supported:
On 3/25/2019 4:45 PM, Mike Diehl wrote:
>
> > So, I don't think it's their network. I've taken pcaps of both legs of
>
> > example calls. On the provider-side, I see 2-way audio. On the
>
> > client-side, I only hear one side.
>
Mike,
In those pcaps, are you seeing the exact same RTP traffic
Hi, and thank you for your suggestion!
As it turns out, my server didn't even HAVE an rtp.conf file... (No, I don't
know
how that happened...)
So I created one with:
rtpstart=1
rtpend=2
and reloaded chan_sip.
I hope that is sufficient. Or do I need to restart asterisk completely?
