<html><head><base href="x-msg://86/"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">So I made the change you suggested. That still hasn't worked, but I did manage to grab some logging from a dropped call.<div><br></div><div><div>[Jul 6 16:19:37] DEBUG[25950] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI/i1/18883203585-7e</div><div>[Jul 6 16:19:37] DEBUG[25950] res_rtp_asterisk.c: Setting the marker bit due to a source update</div><div>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Requested indication 20 on channel DAHDI/i1/18883203585-7e</div><div>[Jul 6 16:19:37] DEBUG[25950] channel.c: Bridge stops bridging channels SIP/7531-00000077 and DAHDI/i1/18883203585-7e</div><div>[Jul 6 16:19:37] DEBUG[25950] cdr_mysql.c: Inserting a CDR record.</div><div>[Jul 6 16:19:37] DEBUG[25950] cdr_mysql.c: SQL command as follows: INSERT INTO cdr (`calldate`,`src`,`dst`,`dcontext`,`channel`,`dstchannel`,`lastapp`,`lastdata`,`duration`,`billsec`,`disposition`,`amaflags`,`accountcode`,`uniqueid`) VALUES ('2011-07-06 15:58:57','7531','8883203585','from-sip','SIP/7531-00000077','DAHDI/i1/18883203585-7e','Dial','DAHDI/g1/18883203585','1240','1238','ANSWERED','3','\"Adam Witwer\"','1309982337.338')</div><div>[Jul 6 16:19:37] DEBUG[25950] channel.c: Hanging up channel 'DAHDI/i1/18883203585-7e'</div><div>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: dahdi_hangup(DAHDI/i1/18883203585-7e)</div><div>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on DAHDI/i1/18883203585-7e</div><div>[Jul 6 16:19:37] DEBUG[25950] sig_pri.c: sig_pri_hangup 1</div><div>[Jul 6 16:19:37] DEBUG[25950] sig_pri.c: Not yet hungup... Calling hangup once with icause, and clearing call</div><div>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Disabled echo cancellation on channel 1</div><div>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Set option TDD MODE, value: OFF(0) on DAHDI/i1/18883203585-7e</div><div>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Updated conferencing on 1, with 0 conference users</div><div>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on DAHDI/i1/18883203585-7e</div><div>[Jul 6 16:19:37] VERBOSE[25950] chan_dahdi.c: -- Hungup 'DAHDI/i1/18883203585-7e'</div><div><br></div><div><div>On Jul 1, 2011, at 2:38 PM, Jonathan Thomas wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div 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" style="page: WordSection1; "><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">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></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">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></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">sip set debug peer XXXXXX (X=peer device id, such as 10001)<o:p></o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">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></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">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></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">JT<o:p></o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div><div style="border-right-style: none; border-bottom-style: none; border-left-style: none; border-width: initial; border-color: initial; border-top-style: solid; border-top-color: rgb(181, 196, 223); border-top-width: 1pt; padding-top: 3pt; padding-right: 0in; padding-bottom: 0in; padding-left: 0in; "><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><b><span style="font-size: 10pt; font-family: Tahoma, sans-serif; ">From:</span></b><span style="font-size: 10pt; font-family: Tahoma, sans-serif; "><span class="Apple-converted-space"> </span>Mark Rosedale [mailto:mrosedale@oreilly.com]<span class="Apple-converted-space"> </span><br><b>Sent:</b><span class="Apple-converted-space"> </span>Friday, July 01, 2011 2:17 PM<br><b>To:</b><span class="Apple-converted-space"> </span><a href="mailto:jonathan.thomas@us.patersons.net" style="color: blue; text-decoration: underline; ">jonathan.thomas@us.patersons.net</a><br><b>Cc:</b><span class="Apple-converted-space"> </span>'Asterisk Users Mailing List - Non-Commercial Discussion'<br><b>Subject:</b><span class="Apple-converted-space"> </span>Re: [asterisk-users] Dropping Conference calls<o:p></o:p></span></div></div></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; ">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></div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; ">I would be interested in the debut/logs if you have them.<o:p></o:p></div></div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; ">I do have <span class="apple-style-span"><span style="font-size: 9pt; font-family: Verdana, sans-serif; ">Spawn extension...exited non-zero on 'SIP/'</span></span><o:p></o:p></div></div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-family: Verdana, sans-serif; ">Here is the specifics </span><o:p></o:p></div></div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><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></div></div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><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></div><div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; ">On Jul 1, 2011, at 11:21 AM, Jonathan Thomas wrote:<o:p></o:p></div></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><br><br><o:p></o:p></div><div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; ">The key item in my logs, which would preface the call dropping, was:<span class="Apple-converted-space"> </span><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 class="Apple-converted-space"> </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]<span class="Apple-converted-space"> </span><br>Sent: Friday, July 01, 2011 10:45 AM<br>To:<span class="Apple-converted-space"> </span><a href="mailto:jonathan.thomas@us.patersons.net" style="color: blue; text-decoration: underline; ">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?<span class="Apple-converted-space"> </span><br><br>I'm looking into the sip session timers. I believe the issue lies there, but<br>haven't confirmed that just yet.<span class="Apple-converted-space"> </span><br>On Jul 1, 2011, at 10:31 AM, Jonathan Thomas wrote:<br><br><br><o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; ">900ms?<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><br><br><br><br>Email has been scanned for viruses<o:p></o:p></div></div></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', serif; "><br>Email has been scanned for viruses<o:p></o:p></div></div><br>Email has been scanned for viruses<br></div></blockquote></div><br></div></body></html>