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:
n
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 be
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?
A