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 mailbox=701@default call-limit=9 allowsubscribe=yes Thanks for any help, Ken -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070509/3a33ad3a/attachment.htm
Ken, I have similar problems every now and then on one of my asterisk boxes. I'm also running CentOS4 on that box. I've found that doing a sip reload when in that state results in something along : Last reload not yet finished (can't remember the exact wording) We're using cisco 7960's here. The ONLY time I've seen this happening is when I reload everything VIA freepbx. It used to do it every time I reloaded. I read somewhere that this was a result of DNS queries not being done in a timely fashion - So I went and replaced all the host statement in my trunk with IP addresses and now it doesn't do it very often at all. I don't know if this is your problem at all but it might be worth a shot. Replace any host names with IP addresses in sip.conf and anywhere else. Failing that and if you're still pulling your hair out at the end of the week ( I know how it is), I would really consider re-installing the box (I'm using centos5 now on this server I'm configuring currently) and starting from scratch. I know it sounds like a cop out but that's what I would do.
That was in my list of things I've done, but failed to mention :). I never have used DNS on this box, but for verification I removed DNS servers and verified all addresses were IP's (which they were). There is no DNS active on this box at all. There's also no freepbx, just straight Asterisk. As for your comment on starting fresh, if I get no further help by this weekend that'll be my fun little weekend project. Thanks for the info. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of mail-lists Sent: Wednesday, May 09, 2007 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... Ken, I have similar problems every now and then on one of my asterisk boxes. I'm also running CentOS4 on that box. I've found that doing a sip reload when in that state results in something along : Last reload not yet finished (can't remember the exact wording) We're using cisco 7960's here. The ONLY time I've seen this happening is when I reload everything VIA freepbx. It used to do it every time I reloaded. I read somewhere that this was a result of DNS queries not being done in a timely fashion - So I went and replaced all the host statement in my trunk with IP addresses and now it doesn't do it very often at all. I don't know if this is your problem at all but it might be worth a shot. Replace any host names with IP addresses in sip.conf and anywhere else. Failing that and if you're still pulling your hair out at the end of the week ( I know how it is), I would really consider re-installing the box (I'm using centos5 now on this server I'm configuring currently) and starting from scratch. I know it sounds like a cop out but that's what I would do. _______________________________________________ --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
whats the asterisk version your using? On 5/10/07, Ken Williams <ken@intermountainelectronics.com> 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 > mailbox=701@default > 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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070509/2dc5e3c4/attachment.htm
Started with 1.4.1, then 1.4.2, then 1.4.4, now the latest SVN (63478). ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of franco escalona Sent: Wednesday, May 09, 2007 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... whats the asterisk version your using? On 5/10/07, Ken Williams <ken@intermountainelectronics.com > 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 mailbox=701@default 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070509/c4557180/attachment-0001.htm
I also get the mysterious SIP INVITE channels. 10.101.2.204 xxx 748e8b0a625 00102/00000 unkn No Init: INVITE And I also am running 1.4.4 on CentOS4. Is that a pattern or just coincidence? The other symptom you mention is this "...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." Do you mean no calls in or out until you reboot? I don't have that thankfully, but I do have a guy telling me that incoming audio just goes away for a few seconds at a time. He says also that it sometimes goes away for long enough time that he was mistaking it for a dropped call. But if he waits long enough it pretty generally always comes back. I have consistent solid network performance from the asterisk server to the ATA (and believe me, I've looked very hard for a network problem), and I don't know what to look at next. Incidentally, the guy hasn't called me since I rebooted last week. Is this similar to how your situation started? ********************************* Adam Moffett Plexicomm, LLC adam@plexicomm.net ph: 866-759-4678x104 *********************************
I mean that SIP phones cannot answer incoming calls or make outgoing calls. When a call comes in on ZAP, it actually rings all the phones like normal, but when you try to answer no one is there. In addition, when you try to dial out you eventually get a message on the phones saying unable to communicate with the server. So there is some traffic still traveling on the SIP channel (the server's dialing extensions from an incoming ZAP call) but no further communication...almost as if it's a one way street of communication. The server can send data out on SIP but isn't receiving any. As for your issue, we haven't really had that (thankfully), so I don't think you're heading down the horrible spot we're in right now. Tonight I'm going to remove all aspects of Asterisk and reinstall fresh, if that fails I'll format & reinstall the entire box. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Adam Moffett Sent: Wednesday, May 09, 2007 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... I also get the mysterious SIP INVITE channels. 10.101.2.204 xxx 748e8b0a625 00102/00000 unkn No Init: INVITE And I also am running 1.4.4 on CentOS4. Is that a pattern or just coincidence? The other symptom you mention is this "...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." Do you mean no calls in or out until you reboot? I don't have that thankfully, but I do have a guy telling me that incoming audio just goes away for a few seconds at a time. He says also that it sometimes goes away for long enough time that he was mistaking it for a dropped call. But if he waits long enough it pretty generally always comes back. I have consistent solid network performance from the asterisk server to the ATA (and believe me, I've looked very hard for a network problem), and I don't know what to look at next. Incidentally, the guy hasn't called me since I rebooted last week. Is this similar to how your situation started? ********************************* Adam Moffett Plexicomm, LLC adam@plexicomm.net ph: 866-759-4678x104 ********************************* _______________________________________________ --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
My configs that I've reworked in the process of trying to fix this SIP problem actually started from Freepbx. I removed and reinstalled Asterisk last night, things seem to be working smoother, I'll no by noon if the problem is fixed or not. Thanks for the help from everyone, Ken ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Deepak Naidu Sent: Wednesday, May 09, 2007 11:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SIP Problems continue... A small way to make little easy, I dont know it people are ok to that, try integrating freepbx & asterisk so you know what the sip configs should look like when things are all well. Things might stop working if there is a bug or change in configs. -- Deepak Ken Williams <ken@intermountainelectronics.com> wrote: I mean that SIP phones cannot answer incoming calls or make outgoing calls. When a call comes in on ZAP, it actually rings all the phones like normal, but when you try to answer no one is there. In addition, when you try to dial out you eventually get a message on the phones saying unable to communicate with the server. So there is some traffic still traveling on the SIP channel (the server's dialing extensions from an incoming ZAP call) but no further communication...almost as if it's a one way street of communication. The server can send data out on SIP but isn't receiving any. As for your issue, we haven't really had that (thankfully), so I don't think you're heading down the horrible spot we're in right now. Tonight I'm going to remove all aspects of Asterisk and reinstall fresh, if that fails I'll format & reinstall the entire box. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Adam Moffett Sent: Wednesday, May 09, 2007 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... I also get the mysterious SIP INVITE channels. 10.101.2.204 xxx 748e8b0a625 00102/00000 unkn No Init: INVITE And I also am running 1.4.4 on CentOS4. Is that a pattern or just coincidence? The other symptom you mention is this "...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." Do you mean no calls in or out until you reboot? I don't have that thankfully, but I do have a guy telling me that incoming audio just goes away for a few seconds at a time. He says also that it sometimes goes away for long enough time that he was mistaking it for a dropped call. But if he waits long enough it pretty generally always comes back. I have consistent solid network performance from the asterisk server to the ATA (and believe me, I've looked very hard for a network problem), and I don't know what to look at next. Incidentally, the guy hasn't called me since I rebooted last week. Is this similar to how your situation started? ********************************* Adam Moffett Plexicomm, LLC adam@plexicomm.net ph: 866-759-4678x104 ********************************* _______________________________________________ --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 _______________________________________________ --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 ________________________________ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now <http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc 2VjA21haWwEc2xrA3RhZ2xpbmU> . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070510/8d1ae7f6/attachment.htm
9 maj 2007 kl. 18.14 skrev Ken Williams:> 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: INVITEThere is an open bug report on this in the bug tracker already. I need your help to find what's causing this issue and provided I can get proper information from you, will spend time locating the bug. First, enable SIP history and catch history for these calls that hang with "sip show history" Secondly, check the dialplan and tell me more. Where are you calling, why doesn't the other end respond? It's usually calls where we retransmit a number of times and then forget to destroy the calls. If I can get a better description so I can repeat this, I'm sure the bug can be killed. In the bug tracker, there's a patch that will help you. However, until I find more exact information about the nature of these calls, I'm unwilling to commit it. To commit a fix to a poorly defined issue is usually causing more issues, something I can do in trunk but don't want to do in release code. Please send the required information directly to my e-mail address and I'll take a look. Thank you for your assistance with this bug. /Olle
Well, I removed and reinstalled Asterisk & Zaptel last night. We haven't had one lock up and we've had zero ghost channels kicking around. I copied my config files straight over, so I'm certain it's not a dialplan issue (I was thinking the same thing you were, and I started throwing hangup statements all over the place). My best guess, I had something conflicting with an older version/SVN that was causing grief. We haven't had a day with zero crashes in 2 weeks, and it was progressively getting worse where we were to the point of 4-5 crashes a day. Going an entire day with no crashes is extremely promissing. I do have a lot of data I captured that I could contribute, but I'm not sure we had the exact same problem. ken -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Olle E Johansson Sent: Thursday, May 10, 2007 12:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... 9 maj 2007 kl. 18.14 skrev Ken Williams:> 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: INVITEThere is an open bug report on this in the bug tracker already. I need your help to find what's causing this issue and provided I can get proper information from you, will spend time locating the bug. First, enable SIP history and catch history for these calls that hang with "sip show history" Secondly, check the dialplan and tell me more. Where are you calling, why doesn't the other end respond? It's usually calls where we retransmit a number of times and then forget to destroy the calls. If I can get a better description so I can repeat this, I'm sure the bug can be killed. In the bug tracker, there's a patch that will help you. However, until I find more exact information about the nature of these calls, I'm unwilling to commit it. To commit a fix to a poorly defined issue is usually causing more issues, something I can do in trunk but don't want to do in release code. Please send the required information directly to my e-mail address and I'll take a look. Thank you for your assistance with this bug. /Olle _______________________________________________ --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
I spoke to soon. Not an hour into the day this morning and we locked up. I'm back to sip debug enable & have turned sip history on, get me the bug number and I'll contribute there. Thanks, Ken -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ken Williams Sent: Thursday, May 10, 2007 5:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SIP Problems continue... Well, I removed and reinstalled Asterisk & Zaptel last night. We haven't had one lock up and we've had zero ghost channels kicking around. I copied my config files straight over, so I'm certain it's not a dialplan issue (I was thinking the same thing you were, and I started throwing hangup statements all over the place). My best guess, I had something conflicting with an older version/SVN that was causing grief. We haven't had a day with zero crashes in 2 weeks, and it was progressively getting worse where we were to the point of 4-5 crashes a day. Going an entire day with no crashes is extremely promissing. I do have a lot of data I captured that I could contribute, but I'm not sure we had the exact same problem. ken -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Olle E Johansson Sent: Thursday, May 10, 2007 12:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... 9 maj 2007 kl. 18.14 skrev Ken Williams:> 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: INVITEThere is an open bug report on this in the bug tracker already. I need your help to find what's causing this issue and provided I can get proper information from you, will spend time locating the bug. First, enable SIP history and catch history for these calls that hang with "sip show history" Secondly, check the dialplan and tell me more. Where are you calling, why doesn't the other end respond? It's usually calls where we retransmit a number of times and then forget to destroy the calls. If I can get a better description so I can repeat this, I'm sure the bug can be killed. In the bug tracker, there's a patch that will help you. However, until I find more exact information about the nature of these calls, I'm unwilling to commit it. To commit a fix to a poorly defined issue is usually causing more issues, something I can do in trunk but don't want to do in release code. Please send the required information directly to my e-mail address and I'll take a look. Thank you for your assistance with this bug. /Olle _______________________________________________ --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 _______________________________________________ --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