<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Arial, Helvetica, sans-serif; font-size: 12pt; color: #000000'>Hi Carsten,<br><br>Thanks for the tip. All audio is going through RTPProxy on the Kamailio server, not directly to Asterisk. <br><br>I will look into that patch.<br><br>Thanks!<br><br><div><span style="font-family: arial,helvetica,sans-serif;">Brett</span><br></div><br>----- Original Message -----<br>From: "Carsten Bock" <carsten@ng-voice.com><br>To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users@lists.sip-router.org><br>Sent: Thursday, June 23, 2011 12:46:11 AM GMT -08:00 US/Canada Pacific<br>Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint<br><br>Hi,<br><br>another solution might be, to either configure an RTP-Timeout on the<br>Asterisk (if you send your calls through the asterisk anyway).<br>You might also consider using the RTPProxy with the patch in the<br>sip-router-repository. With the patch, the RTPProxy will trigger a<br>teardown of calls (via XML-RPC) if the RTP-Session has a timeout.<br><br>Carsten<br><br>2011/6/23 Brett Woollum <brett@woollum.com>:<br>> Hi Alex,<br>><br>> Thanks for this information. I've started researching the session-timer<br>> capabilities in Asterisk, and I think that's my solution. I've already<br>> implemented it on a test system and it works well, except that it's using<br>> reINVITES to update as opposed to UPDATE messages, resulting in chops in the<br>> audio every so often. I'll research this further though.<br>><br>> Thanks again!<br>> Brett<br>><br>> ----- Original Message -----<br>> From: "Alex Balashov" <abalashov@evaristesys.com><br>> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users<br>> Mailing List" <sr-users@lists.sip-router.org><br>> Cc: sr-users@lists.sip-router.org<br>> Sent: Wednesday, June 22, 2011 10:22:18 PM GMT -08:00 US/Canada Pacific<br>> Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to<br>> SIP endpoint<br>><br>> This is a complex topic. There is no way for a proxy like Kamailio to<br>> detect this scenario per se. Kamailio reacts to and forwards signaling<br>> events. If an endpoint disappears, it won't send any of those to indicate<br>> that it has gone away. How would Kamailio know? Media stream timeout?<br>> Kamailio doesn't relay media.<br>> Your Kamailio-side solution is a dialog timeout, requiring use of<br>> dialog-stateful tracking using the dialog module. But that will time out<br>> calls indiscriminately, so you need to make it long enough to not anger your<br>> users but short enough to be useful.<br>> Your endpoint solution is SIP Session Timers.<br>><br>> --<br>> Alex Balashov - Principal<br>> Evariste Systems LLC<br>> 260 Peachtree Street NW<br>> Suite 2200<br>> Atlanta, GA 30303<br>> Tel: +1-678-954-0670<br>> Fax: +1-404-961-1892<br>> Web: http://www.evaristesys.com/<br>> On Jun 23, 2011, at 1:10 AM, Brett Woollum <brett@woollum.com> wrote:<br>><br>> Hello,<br>><br>> We are running Kamailio as a registration point for our SIP phones, which<br>> then interacts with Asterisk. SIP registrations are processed by Kamailio,<br>> but everything else is passed to Asterisk. The Kamailio configuration is<br>> close to the article at:<br>> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.<br>> Everything seems to be working well, until today.<br>><br>> I found several calls today that were still connected to our provider, even<br>> though our SIP phones were not active. There were three calls with timers at<br>> 9 hours and counting. We had some IP connectivity issues earlier today, and<br>> I'm wonder if it's related.<br>><br>> If a SIP phone was connected and on a call (through kamailio), and the<br>> kamailio/asterisk servers became unreachable, the SIP phones will drop the<br>> call. But, it appears that kamailio/asterisk never drop the call in this<br>> case, and the call stays live with the carrier. I had to manually kill the<br>> calls by command prompt.<br>><br>> What's the best way to handle this? Is there a way to have kamailio or<br>> asterisk poll the phone to see if it's still on the call or something? How<br>> can I give visibility to asterisk or kamailio so the calls are always<br>> dropped properly? I don't want to run up a large bill because of calls that<br>> didn't terminate when they should have.<br>><br>> Thanks!<br>> Brett<br>><br>> _______________________________________________<br>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>> sr-users@lists.sip-router.org<br>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users<br>><br>> _______________________________________________ SIP Express Router (SER) and<br>> Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org<br>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users<br>> _______________________________________________<br>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>> sr-users@lists.sip-router.org<br>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users<br>><br>><br><br><br><br>-- <br>Carsten Bock<br>http://www.ng-voice.com<br>mailto:carsten@ng-voice.com<br><br>Schomburgstr. 80<br>22767 Hamburg<br>Germany<br><br>Mobile +49 179 2021244<br>Office +49 40 34927219<br>Fax +49 40 34927220<br><br>_______________________________________________<br>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>sr-users@lists.sip-router.org<br>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users<br></div></body></html>