On Thu, 2005-04-14 at 07:23 -0500, Eric Wieling wrote: > Is Asterisk getting a stream of RTP packets from the SIP client? What > happens if you start talking on the SIP device? Does Asterisk then > start sending RTP? It still sounds like VAD and silence supression is > enabled on the SIP device. >
Let me recap since it was spread over a couple emails back and forth between someone and this is a high volume list. Any sip clinet (local or remote) connecting to my asterisk system is met with silence. All sip clients send a RTP stream correct Asterisk reads DTMF from sip clients with no problem and executes the appropriate dialplan entries asterisk will hang on a playback() or background() call and will send only 1 RTP packet to the SIP slient (note the sip client is sending RTP the whole time, thus I do not think its silence detection on the client) If I press a number that causes a different playback() asterisk will send 1 more RTP packet and hang on that playback. If I go through asterisk but the destination is a sip client the RTP stream works perfectly. The *only* time there is a RTP problem is when the destination is asterisk. musiconhold sends no rtp packets, playback 1 packet only. SIP traffic goes back and forth, a sip show debug reveals proper call setup/tear down. It does not matter if the sip client is local or remote. This setup was working a couple days ago, which makes me curious as to what could possibly be happening. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 881 8487 FreeWorldDialup: 635378
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