[asterisk-users] No Audio for Extension to Extension

[Marco Mouta]

enable rtp debug in your asterisk CLI and check if there's traffic passing.
Would be a first approach I think.

On 1/23/07, Tim Panton <tim at mexuar.com> wrote:
>
>
> On 22 Jan 2007, at 07:28, Troy - Purple Oranges wrote:
>
> > I am at a loss, I can terminate and receive calls via any of my
> > providers with both IAX and SIP.  I use GSM, G729a, and ulaw for those
> > carriers.
> >
> > If I make an extension to extension call - there is no audio at all in
> > either direction.
> >
> > All my extensions are set to use G729a (I have tried changing that
> > though to see if it would fix it).  I am fairly sure it is not a
> > transcoding issue - as the server transcodes for the inbound/outbound
> > calls.
> >
>
> You really need to tell us more!
> At a pure guess however I'd say you have SIP extensions with canreinvite
> set to true. Your internal network however does not permit rtp
> traffic between
> the handsets.
>
> Tim Panton
>
> www.mexuar.net
> www.westhawk.co.uk/
>
>
>
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