whats the asterisk version your using? On 5/10/07, Ken Williams <[EMAIL PROTECTED]> wrote:
SIP channel hang ups are progressively getting worse and I'm really grasping at straws here trying to find out what the cause is. The problem start, once a week or so the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. It's now doing it multiple times a day and I fear having to go back to our old phone system if I can't find a fix in the near future. When the SIP channel locks up the only fix is to restart Asterisk. SIP RELOAD & RELOAD CHAN_SIP do no good. Here's a few things I've noticed and changes I've made in hopes of making it better. First, I've currently got 71 active SIP channels when only 2 people are on the phone. This doesn't happen every time, but could be part of the cause. The 'ghost' channels are all INVITES, how do I clear these without rebooting the system? 10.200.26.116 716 0a2a959d3d3 00102/00000 unkn No Init: INVITE 10.200.26.115 715 1dee947d485 00102/00000 unkn No Init: INVITE 10.200.26.104 704 28808764699 00102/00000 unkn No Init: INVITE 10.200.26.104 704 36d3e88f59c 00102/00000 unkn No Init: INVITE 10.200.26.104 704 0e00060800d 00102/00000 unkn No Init: INVITE Second, I've gone through and basically redone my extensions.conf to have it flow much smoother and clearer. I thought for sure my problem was coming from a loop somewhere in extensions.conf, but I'm now certain my extensions.conf is fine (but I'm glad I redid it, much easier to follow now). Third, I removed 'qualify=yes' from my sip.conf. I had read where people were having SIP channel lockups with this enabled, I again thought I had found the problem...but alas...In addition I had seen someone suggest setting REINVITE=NO, in addition to CANREINVITE=NO...no good. Fourth, I downgraded all my GXP-2000's to the latest released version of the software (1.1.1.14), some were on a newer version that I'm not sure where it came from (1.1.2.x). I also removed the 2 phones that were on 1.1.3.x (they can't be downgraded), as those apparently had lock up issues as well...again thought I had found the problem... Fifth, I installed the latest SVN of 1.4 last night in hopes it was a known issue that had been fixed....nope.... We don't have a very complicated setup at all. The server is running CentOS 4, it has two TDM-400 cards with 6 FXS & 2 FXO. We have about 25 GXP-2000 phones. My dialplan is nice and clean now. If no one has any further suggestions I'm to the point of opening a bug report with digium. I've read a ton on other people who have had this problem and followed the fixes for those people, but I can't seem to get to the bottom of it. I have multiple SIP DEBUG console logs and DEBUG/VERBOSE set to 4 logs around the time SIP stops responding. SIP.CONF: [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=gsm context=from-internal allowsubscribe=yes notifyhold=no limitonpeers=yes [701] type=friend secret=blahblah port=5060 host=dynamic dtmfmode=rfc2833 dial=SIP/701 context=from-internal canreinvite=no reinvite=no [EMAIL PROTECTED] call-limit=9 allowsubscribe=yes Thanks for any help, Ken _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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