[asterisk-users] No audio on remote SIP calls

[SŽébastien Cramatte]

Hi,

This sounds udp RTP problem.
Might be you have some firewall rules that block this kind of traffic ?
As soon I remember, Asterisk by default use random port between 10000 
and 20000 for rtp traffic (you can adjust this in rtp.conf).

- Sebastien


Jonathan Moore escribió:
> Hi everyone.
>
> We have an asterisk server in our main office and phones at each
> remote site.  The remote offices are connected via a MPLS which, to my
> knowledge has no natting going on.
>
> The problem I have is that any call from a remote phone to a remote
> phone (even on the same remote lan) results in no audio.  If I make a
> call from the same LAN the asterisk server is on, to one of these
> remote sites, I get perfect two way audio.  If I play a call from one
> phone to another at a remote site, there is no audio, however, I do
> hear messages (such as voicemail, things from Playback(), etc) that
> originate on the asterisk server.
>
> I've tried adjusting canreinvite= in sip.conf in hopes in might have
> some effect, but so far nothing.
>
> Suggestions on where else to look, or what the problem might be?
>
> Which configs would be useful in troubleshooting?
>
> Thanks.
>
> -jonathan
>
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