Re: [asterisk-users] Call drop weirdness
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
Re: [asterisk-users] Call drop weirdness
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. I've attached a sanitized text version of the SIP signaling surrounding the time the outbound RTP stream dropped on this particular call. I'm no SIP expert, so there may not be enough info in the file to tell anything. A few notes about the file: 1. X.X.X.X is the public IP our asterisk server is behind. 2. Y.Y.Y.Y is the IP given to us by our provider to use in our SIP trunk through which inbound calls arrive. 3. Z.Z.Z.Z is the IP of our provider's server involved in the RTP stream. 4. DID is our DID. 5. CID is the number of the incoming caller. 6. The outbound RTP stream appears to drop three packets prior to the SIP BYE request. Any thoughts on what might be going wrong? Do I need to post more info? Or am I on the wrong track altogether? Kind Regards, Chris OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK10bfdfb7;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as789d30aa To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 5309893149786df519e2ffa6282fb15d@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK24883b1c;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as395fd02c To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 59e57daf68d31595480ca5927505485f@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK10bfdfb7;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as789d30aa To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 5309893149786df519e2ffa6282fb15d@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK24883b1c;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as395fd02c To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 59e57daf68d31595480ca5927505485f@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 SIP/2.0 503 Unable to load gateways Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK24883b1c;rport=5060 From: Unknown sip:Unknown@X.X.X.X;tag=as395fd02c To: sip:Y.Y.Y.Y;tag=71fd1b189ab888f8d5fb24b00af87228.acb1 Call-ID: 59e57daf68d31595480ca5927505485f@X.X.X.X:5060 CSeq: 102 OPTIONS Server: DFSGW Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK10bfdfb7;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as789d30aa To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 5309893149786df519e2ffa6282fb15d@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 SIP/2.0 503 Unable to load gateways Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK10bfdfb7;rport=5060 From: Unknown sip:Unknown@X.X.X.X;tag=as789d30aa To: sip:Y.Y.Y.Y;tag=71fd1b189ab888f8d5fb24b00af87228.1bf8 Call-ID: 5309893149786df519e2ffa6282fb15d@X.X.X.X:5060 CSeq: 102 OPTIONS Server: DFSGW Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK65801b0b;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as46c7a0b8 To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 0d78056a4c3c4b5d167b013c41450be9@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK65801b0b;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as46c7a0b8 To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 0d78056a4c3c4b5d167b013c41450be9@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:30 GMT Allow: INVITE,
Re: [asterisk-users] Call drop weirdness
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. The thing is, it happens on such an irregular basis (once or twice per day) that I can't get a data dump to see what actually happens. Some times there is a bit of artifacting which takes place just prior to the drop, but mostly nothing: it just drops. I've checked and rechecked firewall settings. Bandwidth consumption on the Inet link varies, but the dropped audio happens even on off-peak times. I'm considering giving the Asterisk box a public IP on one IF and bypassing the FW to rule out NAT weirdness. Any thoughts on things to look at would be greatly appreciated. Kind Regards, Chris I'm not sure if this is any help, but I had a familiar issue to this, except it involved transferring to another internal extension. The symptoms were the same though. Only outbound audio would cut out and it was very sporadic (~10% of transfers). The issue ended up being with the trunking service and their spotty support with UPDATE messages. We had to disable rpid_update in sip.conf and a couple other bits that I can't offhand remember. I'd check with the trunk provider on the issue. Best of luck, Brett Lehrer -- _ -- 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
[asterisk-users] Call drop weirdness
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. The thing is, it happens on such an irregular basis (once or twice per day) that I can't get a data dump to see what actually happens. Some times there is a bit of artifacting which takes place just prior to the drop, but mostly nothing: it just drops. I've checked and rechecked firewall settings. Bandwidth consumption on the Inet link varies, but the dropped audio happens even on off-peak times. I'm considering giving the Asterisk box a public IP on one IF and bypassing the FW to rule out NAT weirdness. Any thoughts on things to look at would be greatly appreciated. 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
