[asterisk-users] No audio on remote SIP calls

[Jonathan Moore]

On Thu, Aug 6, 2009 at 5:17 PM, SŽébastien
Cramatte<scramatte at zensoluciones.com> wrote:
> 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).

In theory, there should be no firewalls between my asterisk server
and the remote phones.  I've opened a ticket with ATT with that
exact question, as well as a question of rather any NATing is going
on, though, I doubt this is the case, and this is the first time this type
of problem has happened in over 4 years.

The idea of RTP being to blame would make sense though.  I can
still transfer and such, and watching the console, I see when I press
various keys on the phone, so it seems that the SIP traffic is working
out fine.  (I do understand that right?  SIP == control RTP == voice
in a very generic sense?)

I plan to take a packet trace in the morning on the asterisk server and
see what is going on at that level.  Hints as to what I should be looking
for?

-jonathan