My next attempt at this is going to be putting a hub in between the path to the switch. I'm hoping to be able to sniff the packets to see what's going on.<br><br>Also, using the network status page on the hard phones, the transmit and receive counters for the direction of the channel slows way down as if almost no data is being transmitted.
<br><br>How do I send a sip debug?<br><br>Thanks.<br><br><div><span class="gmail_quote">On 8/2/06, <b class="gmail_sendername">Joshua Colp</b> <<a href="mailto:jcolp@digium.com">jcolp@digium.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;">
----- Original Message -----<br>From: Gary Richardson<br>[mailto:<a href="mailto:gary.richardson@gmail.com">gary.richardson@gmail.com</a>]<br>To: Asterisk Users Mailing List -<br>Non-Commercial Discussion [mailto:<a href="mailto:asterisk-users@lists.digium.com">
asterisk-users@lists.digium.com</a>]<br>Sent:<br>Wed, 02 Aug 2006 13:54:04 -0300<br>Subject: [asterisk-users] canreinvite=yes<br>and RTP dropping in and out<br><br><br>> Hey guys,<br>><br>> I'm having yet another strange problem. I've recently set canreinvite=yes,
<br>> allowing the RTP streams to avoid our * server. Now, a few people are<br>> experience one way audio drops on internal calls. External calls are working<br>> fine (they re-invite directly to a Cisco router). Sometimes, if you wait 20
<br>> seconds or more, the stream will resume. Flipping the person on and off hold<br>> won't resume the stream.<br>><br>> We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem<br>> to happen all of the time. There are no sip messages being exchanged when
<br>> the stream stops or restarts.<br>><br>> Any suggestions?<br><br>If the audio is going directly there's not too much you can do to examine it. There may be software out there to sniff the data on your network and examine the RTP stream, maybe even see when it drops out (if it really does drop out, ie: stream actually stops). I know there's some Windows software out there capable of this as I picked a copy up while at Spring VON but you might need to look around. OH - can you also send a sip debug with the reinvites? I'm just curious to see the RTP information in the SDP.
<br><br>> Thanks.<br><br>Joshua Colp<br>Digium<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list
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