On Wed, Oct 31, 2012 at 10:31 AM, Chris Nighswonger <cnighswon...@foundations.edu> wrote:
>> I'm running Asterisk 10.7.0 with three sip trunks to my call termination >> provider. For the most part everything works great. > >>However, at apparently random times and usually about 20 mins or so into >> the call, the outbound audio stream dies. >>The call stays connected and the inbound audio works fine. >So I've been watching this problem and was finally able to get a pcap >while it happened. <snip> > Any thoughts on what might be going wrong? Do I need to post more >info? Or am I on the wrong track altogether? After lots of grinding through traces and data dumps both on my end and my provider's, it turned out I was on the wrong track altogether. I finally threw together a script to log counter stats from the switchport into which our pbx is plugged, in spite of no noticeable counter activity. From this I found that the port was accumulating align errors at very slow rate; more like small bursts. So wrote a script to log this counter to an RRD and added it to a graph of traffic control rates. This allowed me to associate the bursts of align errors with RTP data flow. The graph here (http://www.screencast.com/t/vMsi3gVke4) contains two bursts of align errors. The first walked all over a call resulting in the outbound RTP stream dropping. As soon as the errors stopped the audio picked back up. The second burst correlates with log entries like this (no calls were placed or received during this burst): [2012-11-09 14:23:11] NOTICE[4199] chan_sip.c: Peer 'didforsale_outbound' is now UNREACHABLE! Last qualify: 84 [2012-11-09 14:23:13] NOTICE[4199] chan_sip.c: Peer 'didforsale_did' is now UNREACHABLE! Last qualify: 84 Interestingly enough, long ping sequences with large packet payloads do not seem to trigger any errors. Having changed cables, ports, as well as for duplex and speed mismatches, the only remaining hardware to be checked is the NIC, which I suspect is bad. So I'm going to switch over to our backup pbx and test that theory. I apologize for the lengthy explanation, but perhaps it will help some other person with a similarly maddening problem. Kind Regards, Chris -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users