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 <[EMAIL PROTECTED]> 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/ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users