Thank you for your reply sir. I tried setting qualify=yes my CPU spiked to 113%
i continuously see this on my CLI> Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669 handle_response_peerpoke: Peer '118555' is now Reachable. (388ms / 2000ms) [Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669 handle_response_peerpoke: Peer '110100' is now Lagged. (2354ms / 2000ms) [Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669 handle_response_peerpoke: Peer '118777' is now Reachable. (432ms / 2000ms) [Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669 handle_response_peerpoke: Peer '118777' is now Reachable. (436ms / 2000ms) [Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669 handle_response_peerpoke: Peer '118555' is now Reachable. (440ms / 2000ms) [Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669 handle_response_peerpoke: Peer '118555' is now Reachable. (444ms / 2000ms) could that be the cause of high cpu? i'm logged in on the cli asterisk -vr, my verbosity is only set to 1. how come i keep on seeing the NOTICE? thanks again in advanced regards, nhadie Rob Hillis wrote: > If a phone is unplugged, it's not likely to have time to send > notification of this to Asterisk before it powers off. There's nothing > you can add to your dialplan to overcome this, however you *can* set the > "qualify" parameter within sip.conf (or it's equivalent realtime table) > to overcome this. > > See http://www.voip-info.org/wiki/view/Asterisk+sip+qualify for more > information. Short version is that configuring a qualify interval is > the equivalent of setting up a heartbeat between Asterisk and registered > devices configured with a qualify interval. If the heartbeat fails, the > phone's registration is suspended. > > Nhadie wrote: >> Hi, >> >> I'm running asterisk realtime, i had prob when a user does not >> unregister properly. >> >> I tested with SPA942 and a PAP2, when phone is registered, i call using >> the SPA using x-lite no problem, but when i unplugged the power, it does >> not unregister properly, so asterisk think SPA942 is still registered, >> when i call using x-lite, asterisk tries to call it.so it gets stuck at >> >> [Aug 11 21:37:31] -- Called 102104 >> >> until it reached the timed out i set in the dialplan which is 30 secs >> >> Dial(SIP/${EXTEN}|30|t|M(setmusiconhold,moh-${EXTEN})) >> >> is there something i can add on my dialplan to first detect that the >> user is not available, or maybe force unregister, anything that would >> not make my dialplan to wait for 30 secs. >> >> also i'm not using rtcachefriends, how would i know in the CLI which >> user is registered? i tried sip prune but it shows me nothing >> >> sip prune realtime peer all >> No peers found to prune. >> >> anyone experienced this? >> >> thank you >> >> regards. >> nhadie >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> !DSPAM:48a0464541521298081403! >> >> >> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
