enable rtp debug in your asterisk CLI and check if there's traffic passing. Would be a first approach I think.<br><br><div><span class="gmail_quote">On 1/23/07, <b class="gmail_sendername">Tim Panton</b> <<a href="mailto:tim@mexuar.com">
tim@mexuar.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>On 22 Jan 2007, at 07:28, Troy - Purple Oranges wrote:
<br><br>> I am at a loss, I can terminate and receive calls via any of my<br>> providers with both IAX and SIP. I use GSM, G729a, and ulaw for those<br>> carriers.<br>><br>> If I make an extension to extension call - there is no audio at all in
<br>> either direction.<br>><br>> All my extensions are set to use G729a (I have tried changing that<br>> though to see if it would fix it). I am fairly sure it is not a<br>> transcoding issue - as the server transcodes for the inbound/outbound
<br>> calls.<br>><br><br>You really need to tell us more!<br>At a pure guess however I'd say you have SIP extensions with canreinvite<br>set to true. Your internal network however does not permit rtp<br>traffic between
<br>the handsets.<br><br>Tim Panton<br><br><a href="http://www.mexuar.net">www.mexuar.net</a><br><a href="http://www.westhawk.co.uk/">www.westhawk.co.uk/</a><br><br><br><br>_______________________________________________<br>
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