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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple style='word-wrap: break-word;-webkit-nbsp-mode: space;-webkit-line-break: after-white-space'><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>The exited non-zero is typical when a call has ended. What I would recommend (easiest method) is for you to enter the CLI using: asterisk –rvvvvvvvvvvvvvvvvvvvdddd<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>The v’s will provide more verbose logging, the 4 d’s will place the core in debug mode(4). Once in the CLI, pick a phone you will use as a test unit and issue a <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>sip set debug peer XXXXXX (X=peer device id, such as 10001)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>This will seriously increase the size of your logging – but should provide you with a very thorough trace of the call as its in flight, including the SIP dialog between the phone/server. <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Perhaps you can enable the above, place a call that drops, then snip that section of the full log and send it to the list for parsing. It’s the best way to nail down an issue like this.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>JT<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Mark Rosedale [mailto:mrosedale@oreilly.com] <br><b>Sent:</b> Friday, July 01, 2011 2:17 PM<br><b>To:</b> jonathan.thomas@us.patersons.net<br><b>Cc:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br><b>Subject:</b> Re: [asterisk-users] Dropping Conference calls<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>So I didn't have sip debug set. So I don't have any SIP TIMER's in my log. I have that set now. <o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I would be interested in the debut/logs if you have them.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I do have <span class=apple-style-span><span style='font-size:9.0pt;font-family:"Verdana","sans-serif"'>Spawn extension...exited non-zero on 'SIP/'</span></span><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><span style='font-family:"Verdana","sans-serif"'>Here is the specifics </span><o:p></o:p></p></div><div><p class=MsoNormal><span style='font-family:"Verdana","sans-serif"'>VERBOSE[10928] pbx.c: == Spawn extension (from-sip, 1***, 1) exited non-zero on 'SIP/7XXX-000009d7'</span><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><span style='font-family:"Verdana","sans-serif"'>Not sure if that relates or not, but it is the only hit for the connection between my sip client and the PRI going outbound right before the hangup. </span><o:p></o:p></p><div><div><p class=MsoNormal>On Jul 1, 2011, at 11:21 AM, Jonathan Thomas wrote:<o:p></o:p></p></div><p class=MsoNormal><br><br><o:p></o:p></p><div><p class=MsoNormal>The key item in my logs, which would preface the call dropping, was: <br>[2011-06-28 09:43:49] DEBUG[25563] chan_sip.c: ** SIP TIMER: Cancelling<br>retransmit of packet (reply received) Retransid #858<br><br>For instance - a call would be connected. SIP debug/core debug on. At the<br>14:30 mark I would begin tailing the full log. Once I saw the SIP TIMER<br>notice, it would be followed by a new INVITE (re-invite) SIP transmission<br>that would be sent to the phone currently on call. This re-invite was odd<br>in that it would be on a different port to the phone than was already<br>established (for example the NAT outgoing SIP OPTIONS would be sent to the<br>phone on port 27608 - and this re-invite might go out on port 35780). The<br>behavior following would be: Asterisk would hang up as though the parties<br>disconnected - however the phone would show the call was still going and<br>would continue sending SIP responses to asterisk indicating as such. When<br>the phone was manually hung up it would send a SIP BYE (as normal) to<br>asterisk - indicating it had no notice that Asterisk dropped the call.<br><br>Adding to sip.conf<br><span class=apple-tab-span> </span>session-timers=refuse<br>Resolved the issue by stopping Asterisk from sending these re-invites during<br>a live call.<br><br>Hope that helps! I have more SIP debugs/logs if they're useful to ya.<br><br>JT<br><br><br>-----Original Message-----<br>From: Mark Rosedale [mailto:mrosedale@oreilly.com] <br>Sent: Friday, July 01, 2011 10:45 AM<br>To: <a href="mailto:jonathan.thomas@us.patersons.net">jonathan.thomas@us.patersons.net</a>; Asterisk Users Mailing List -<br>Non-Commercial Discussion<br>Subject: Re: [asterisk-users] Dropping Conference calls<br><br>What would I be looking for in the logs to indicate that time? <br><br>I'm looking into the sip session timers. I believe the issue lies there, but<br>haven't confirmed that just yet. <br>On Jul 1, 2011, at 10:31 AM, Jonathan Thomas wrote:<br><br><br><o:p></o:p></p><p class=MsoNormal>900ms?<o:p></o:p></p><p class=MsoNormal><br><br><br><br>Email has been scanned for viruses<o:p></o:p></p></div></div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal><br>Email has been scanned for viruses<o:p></o:p></p></div><BR>
Email has been scanned for viruses<BR>
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