<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><meta http-equiv=Content-Type content="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 12 (filtered medium)"><base href="x-msg://86/"><style><!--
/* Font Definitions */
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
{font-family:Tahoma;
panose-1:2 11 6 4 3 5 4 4 2 4;}
@font-face
{font-family:Verdana;
panose-1:2 11 6 4 3 5 4 4 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
span.apple-converted-space
{mso-style-name:apple-converted-space;}
span.apple-style-span
{mso-style-name:apple-style-span;}
span.apple-tab-span
{mso-style-name:apple-tab-span;}
span.EmailStyle20
{mso-style-type:personal-reply;
font-family:"Calibri","sans-serif";
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;
font-size:10.0pt;}
@page WordSection1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.WordSection1
{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</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'>Talk about getting lucky with a <snip>! The first line you provided in your log was enough to look-up related errors and find a similar one. Although I have not encountered a Frame_Control(8), it’s indicated as ‘Congestion’ by frame.h<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */<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'>The one issue I found that appears similar (at least for the error/behavior) was in relation to the admin’s chan_dahdi.conf – the callprogress=yes option to be specific. Perhaps posting your chan_dahdi.conf would be helpful in at least verifying your settings are correct.<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><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"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Mark Rosedale<br><b>Sent:</b> Wednesday, July 06, 2011 5:37 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 made the change you suggested. That still hasn't worked, but I did manage to grab some logging from a dropped call.<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI/i1/18883203585-7e<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] res_rtp_asterisk.c: Setting the marker bit due to a source update<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Requested indication 20 on channel DAHDI/i1/18883203585-7e<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] channel.c: Bridge stops bridging channels SIP/7531-00000077 and DAHDI/i1/18883203585-7e<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] cdr_mysql.c: Inserting a CDR record.<o:p></o:p></p></div><div><p class=MsoNormal>[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')<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] channel.c: Hanging up channel 'DAHDI/i1/18883203585-7e'<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: dahdi_hangup(DAHDI/i1/18883203585-7e)<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on DAHDI/i1/18883203585-7e<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] sig_pri.c: sig_pri_hangup 1<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] sig_pri.c: Not yet hungup... Calling hangup once with icause, and clearing call<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Disabled echo cancellation on channel 1<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Set option TDD MODE, value: OFF(0) on DAHDI/i1/18883203585-7e<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Updated conferencing on 1, with 0 conference users<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on DAHDI/i1/18883203585-7e<o:p></o:p></p></div><div><p class=MsoNormal>[Jul 6 16:19:37] VERBOSE[25950] chan_dahdi.c: -- Hungup 'DAHDI/i1/18883203585-7e'<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><div><p class=MsoNormal>On Jul 1, 2011, at 2:38 PM, Jonathan Thomas wrote:<o:p></o:p></p></div><p class=MsoNormal><br><br><o:p></o:p></p><div><div><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</span><o:p></o:p></p></div><div><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</span><o:p></o:p></p></div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div><div><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)</span><o:p></o:p></p></div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div><div><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. </span><o:p></o:p></p></div><div><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.</span><o:p></o:p></p></div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>JT</span><o:p></o:p></p></div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in;border-width:initial;border-color:initial'><div><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span class=apple-converted-space><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> </span></span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>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">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</span><o:p></o:p></p></div></div></div><div><p class=MsoNormal> <o:p></o:p></p></div><div><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><div><div><p class=MsoNormal> <o:p></o:p></p></div></div><div><div><p class=MsoNormal>I would be interested in the debut/logs if you have them.<o:p></o:p></p></div></div><div><div><p class=MsoNormal> <o:p></o:p></p></div></div><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><div><div><p class=MsoNormal> <o:p></o:p></p></div></div><div><div><p class=MsoNormal><span style='font-family:"Verdana","sans-serif"'>Here is the specifics </span><o:p></o:p></p></div></div><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><div><div><p class=MsoNormal> <o:p></o:p></p></div></div><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><div><div><p class=MsoNormal>On Jul 1, 2011, at 11:21 AM, Jonathan Thomas wrote:<o:p></o:p></p></div></div><div><p class=MsoNormal><br><br><br><o:p></o:p></p></div><div><div><p class=MsoNormal>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><span class=apple-converted-space> </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">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><br><o:p></o:p></p></div><div><p class=MsoNormal>900ms?<o:p></o:p></p></div><div><p class=MsoNormal><br><br><br><br>Email has been scanned for viruses<o:p></o:p></p></div></div></div><div><p class=MsoNormal> <o:p></o:p></p></div></div><div><p class=MsoNormal><br>Email has been scanned for viruses<o:p></o:p></p></div><p class=MsoNormal><br>Email has been scanned for viruses<o:p></o:p></p></div></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>