Re: [asterisk-users] Asterisk 1.4 reliability problems
Matt Florell [EMAIL PROTECTED] writes: But seriously, several of my clients use SIP exclusively, passing tens of thousand of calls a day on Asterisk 1.2.X with no issues. I have noticed that the load is slightly lower for SIP-only in 1.4, but I have not noticed any stability issues revolving around SIP on 1.2.X. No hung calls? Our 1.2.x customer PBX's are drowning in channel.c: Avoided deadlock for '0x91dbee8', 9 retries!. Of course you can just ignore the hung calls if you want, but they mess up hint state and prevent graceful restarts. 1.4.x fixes it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On 3/19/08, Benny Amorsen [EMAIL PROTECTED] wrote: Matt Florell [EMAIL PROTECTED] writes: But seriously, several of my clients use SIP exclusively, passing tens of thousand of calls a day on Asterisk 1.2.X with no issues. I have noticed that the load is slightly lower for SIP-only in 1.4, but I have not noticed any stability issues revolving around SIP on 1.2.X. No hung calls? Our 1.2.x customer PBX's are drowning in channel.c: Avoided deadlock for '0x91dbee8', 9 retries!. Of course you can just ignore the hung calls if you want, but they mess up hint state and prevent graceful restarts. 1.4.x fixes it. I will say that we did notice some SIP issues with older 1.2 releases, but on the current 1.2.24+ releases we really haven't had many problems, and we do not have hung channels. I should mention that most of these installations have all phones on a LAN and almost none of the calls are native SIP-bridged since they go through meetme rooms which might account for why we do not see problems like this. As for 1.4.X we are moving closer to putting a live production machine on it, just a few more weeks of testing like we have had for the last month, and I should be convinced of it's stability. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
You hit the nail on the head when you said , just a few more weeks of testing like we have had for the last month Some of the code you see makes you think the testing was slightly more than a clean compile :) Matt Florell wrote: On 3/19/08, Benny Amorsen [EMAIL PROTECTED] wrote: Matt Florell [EMAIL PROTECTED] writes: But seriously, several of my clients use SIP exclusively, passing tens of thousand of calls a day on Asterisk 1.2.X with no issues. I have noticed that the load is slightly lower for SIP-only in 1.4, but I have not noticed any stability issues revolving around SIP on 1.2.X. No hung calls? Our 1.2.x customer PBX's are drowning in channel.c: Avoided deadlock for '0x91dbee8', 9 retries!. Of course you can just ignore the hung calls if you want, but they mess up hint state and prevent graceful restarts. 1.4.x fixes it. I will say that we did notice some SIP issues with older 1.2 releases, but on the current 1.2.24+ releases we really haven't had many problems, and we do not have hung channels. I should mention that most of these installations have all phones on a LAN and almost none of the calls are native SIP-bridged since they go through meetme rooms which might account for why we do not see problems like this. As for 1.4.X we are moving closer to putting a live production machine on it, just a few more weeks of testing like we have had for the last month, and I should be convinced of it's stability. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Curious, you mention a number of problems that have gone on for months Question: Have you reported ANY or ALL of them to DIGIUM and if so what has been their response on each of these problems ? Ben Willcox wrote: Hello All, We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. We are running asterisk 1.4.17~dfsg-2+b1 on Debian Lenny, with a Digium PRI card, and have approximately 120 sip peers, mostly Snom 360s, with a few Grandstream GXP2000 and a handful of Handytone 486 units. The symptoms, when they occur, are as follows: -The inability to receive incoming calls to our ISDN PRI (callers get a busy tone), this starts off becoming intermittent but becomes permanent. -Asterisk cli commands work once, but then no longer return any data until disconnecting and reconnecting to the cli, i.e. sip show peers, show channels etc. -Internal SIP calls stop working -Calls remain stuck in queues, the queue members do not ring, and show as Busy when issuing a 'queue show' command. We've actually had these sort of problems for many months now, which originally started when we were running Asterisk 1.2 on Gentoo. We have done a large amount of fault finding and testing, which has involved a replacement ISDN card, reinstall on complete different server hardware, and changing to Asterisk 1.4 on Debian Lenny. I believe there may be two separate issues here - we did track down one problem to our cacti and nagios monitoring scripts, which were connecting and disconnecting to the manager interface several times per minute, which eventually caused asterisk to give the above symptoms, although in addition to the above, asterisk would consume 100% cpu on the box, and eventually need a hard-reboot of the server. I posted about this to the list a few weeks ago, and it was confirmed that this could cause such a problem. After stopping these services the problems were much reduced. However, we have now completely disabled the manager interface (enabled=no in manager.conf), and yesterday the problem occurred again - a restart of asterisk got everything going again. So really I'm at a loss as to where to go from here. A colleague of mine also has the same problem at his site running Asterisk 1.4 on Debian Lenny, he has never used the manager interface, and has completely different server hardware and ISDN card, so I wonder if it's a Debian specific problem? One option is to try reverting back to Asterisk 1.2, but that isn't really a long-term solution. We also had major problems with 1.2 with our Snom 360 phones, as with any Snom firmware 6.2.2 there was a serious problem whereby on hangup the channels were not cleared down, meaning we had many outgoing ISDN calls held open for many hours until we realised the problem. This problem does not occur in Asterisk 1.4, although we have many log messages such as: chan_sip.c: Remote host can't match request BYE to call callid so I don't know if this is anything to worry about? Any help would be gratefully received! Thanks, Ben ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Ben Willcox wrote: Hello All, One option is to try reverting back to Asterisk 1.2, but that isn't really a long-term solution. We also had major problems with 1.2 with Two things, 1.) On your queue setup, avoid using AgenCallbackLogin, it's known to cause deadlocked channels. 2.) Restart the Asterisk service once a week. I do this via a CRON job at 3am on Sundays. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Could you clarify what you mean by a Dead Locked Channel ? That is not a term I am familiar with used in context to channels, databases yes, channels ??? Thx Doug Lytle wrote: Ben Willcox wrote: Hello All, One option is to try reverting back to Asterisk 1.2, but that isn't really a long-term solution. We also had major problems with 1.2 with Two things, 1.) On your queue setup, avoid using AgenCallbackLogin, it's known to cause deadlocked channels. 2.) Restart the Asterisk service once a week. I do this via a CRON job at 3am on Sundays. Doug ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Al Baker wrote: Could you clarify what you mean by a Dead Locked Channel ? That is not a term I am familiar with used in context to channels, databases yes, channels ??? Non functional, but showing up within the console and not being released. core show channels, sip show channels, etc. Channels within Asterisk link technology types. IAX,SIP,ZAP, Whatever. I may have it incorrect; if so, someone will correct me. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On Tue, Mar 18, 2008 at 5:40 AM, Ben Willcox [EMAIL PROTECTED] wrote: Hello All, We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. We are running asterisk 1.4.17~dfsg-2+b1 on Debian Lenny, with a Digium PRI card, and have approximately 120 sip peers, mostly Snom 360s, with a few Grandstream GXP2000 and a handful of Handytone 486 units. The symptoms, when they occur, are as follows: -The inability to receive incoming calls to our ISDN PRI (callers get a busy tone), this starts off becoming intermittent but becomes permanent. -Asterisk cli commands work once, but then no longer return any data until disconnecting and reconnecting to the cli, i.e. sip show peers, show channels etc. -Internal SIP calls stop working -Calls remain stuck in queues, the queue members do not ring, and show as Busy when issuing a 'queue show' command. We've actually had these sort of problems for many months now, which originally started when we were running Asterisk 1.2 on Gentoo. We have done a large amount of fault finding and testing, which has involved a replacement ISDN card, reinstall on complete different server hardware, and changing to Asterisk 1.4 on Debian Lenny. I believe there may be two separate issues here - we did track down one problem to our cacti and nagios monitoring scripts, which were connecting and disconnecting to the manager interface several times per minute, which eventually caused asterisk to give the above symptoms, although in addition to the above, asterisk would consume 100% cpu on the box, and eventually need a hard-reboot of the server. I posted about this to the list a few weeks ago, and it was confirmed that this could cause such a problem. After stopping these services the problems were much reduced. However, we have now completely disabled the manager interface (enabled=no in manager.conf), and yesterday the problem occurred again - a restart of asterisk got everything going again. So really I'm at a loss as to where to go from here. A colleague of mine also has the same problem at his site running Asterisk 1.4 on Debian Lenny, he has never used the manager interface, and has completely different server hardware and ISDN card, so I wonder if it's a Debian specific problem? One option is to try reverting back to Asterisk 1.2, but that isn't really a long-term solution. We also had major problems with 1.2 with our Snom 360 phones, as with any Snom firmware 6.2.2 there was a serious problem whereby on hangup the channels were not cleared down, meaning we had many outgoing ISDN calls held open for many hours until we realised the problem. This problem does not occur in Asterisk 1.4, although we have many log messages such as: chan_sip.c: Remote host can't match request BYE to call callid so I don't know if this is anything to worry about? Any help would be gratefully received! Thanks, Ben I have seen this when banging on the AMI but you eliminated that. Why not try a different OS such as CentOS for now? That would be my next step. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On Tue, 2008-03-18 at 07:04 -0400, Al Baker wrote: Could you clarify what you mean by a Dead Locked Channel ? That is not a term I am familiar with used in context to channels, databases yes, channels ??? A channel got locked but never unlocked causing all sorts of funky behavior. It's a bug. The developers have fixed a ton of these deadlocks in 1.4 so it's usually a good plan to try the latest and greatest version to see if the problem goes away. I'm not very familiar with queue setups but Doug Lytle's advice sounds like a plan. And try 1.4.19-rc2 to see if the deadlock problem persists. If it does then please file a bug so it can be looked at. Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
I would suggest upgrading to at least 1.4.18. I was able to run it for about 2 weeks and almost one million calls before I could get it to crash, and the 1.4.19RC2 seems to fix even more of the locking issues as well. I know a lot of these problems still existed under 1.4.17. MATT--- On 3/18/08, Patrick [EMAIL PROTECTED] wrote: On Tue, 2008-03-18 at 07:04 -0400, Al Baker wrote: Could you clarify what you mean by a Dead Locked Channel ? That is not a term I am familiar with used in context to channels, databases yes, channels ??? A channel got locked but never unlocked causing all sorts of funky behavior. It's a bug. The developers have fixed a ton of these deadlocks in 1.4 so it's usually a good plan to try the latest and greatest version to see if the problem goes away. I'm not very familiar with queue setups but Doug Lytle's advice sounds like a plan. And try 1.4.19-rc2 to see if the deadlock problem persists. If it does then please file a bug so it can be looked at. Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On Tue, 18 Mar 2008, Steve Totaro wrote: Why not try a different OS such as CentOS for now? That would be my next step. I wouldn't suggest chasing distros is the way to solve issues, especially if you're happy with the hardware. Personally, I'd go back to Debian, but stick to stable (Etch) and then compile and install a custom kernel tailored exactly to your hardware, then compile and install your own asterisk from source. But only because that's what I do, and it works for me ... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Check around on bugs.digium.com. You'll find a number of issues reported that sound similar. I'm hoping that 1.4.19 will fix a lot of stuff, since the release candidates seem much more stable to me. I couldn't keep Asterisk up for more than a few days before on 1.4.18. I've also applied a few SIP-related patches from various bug reports and things are much, much more stable. 1.4.17, which you mentioned, is also very buggy. 1.4.18 fixed many issues. Norman Franke Answering Service for Directors, Inc. www.myasd.com On Mar 18, 2008, at 7:40 AM, [EMAIL PROTECTED] wrote: We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Hi All, Thanks for all the replies. Here are my responses to the responses: On Tue, 2008-03-18 at 06:13 -0400, Al Baker wrote: Curious, you mention a number of problems that have gone on for months Question: Have you reported ANY or ALL of them to DIGIUM and if so what has been their response on each of these problems ? We have been working very closely with the reseller that supplied us with the system, and although we have made progress over this time and they have given us a lot of technical support, I now feel that it will be quicker to progress the current issues independently. I don't know if the issues were escalated as far as Digium though. Tzafrir Cohen wrote: The symptoms you mention suggest some sort of deadlock. Please enable debug and the full log. Maybe this will provide some hints. But please check that the full log is rotated in /etc/logrotate.d/asterisk . Can you reproduce this situation? e.g.: by extensive usage of the manager interface? If so, it might help for testing. I will enable full debug logging. I suspect that we could reproduce the original problem with the manager interface by stress testing it with multiple connections, but I'm not sure if this is the same problem that we are currently experiencing. I also want to avoid causing problems on our production system at the moment, as it is rather 'delicate' as far as the users are concerned at the moment. Steve Totaro wrote: Why not try a different OS such as CentOS for now? That would be my next step. I have considered this, to at least to establish whether it is a Debian specific problem, either with the asterisk packages themselves, or some other configuration or package issue. I am umming and ahhing between this and Gordon's suggestion below: Gordon Henderson wrote: Personally, I'd go back to Debian, but stick to stable (Etch) and then compile and install a custom kernel tailored exactly to your hardware, then compile and install your own asterisk from source. I'm thinking that this may be the way I should go, then I will have the freedom to install any version of asterisk that I need, whilst also keeping my favourite distro. Doug Lytle wrote: Two things, 1.) On your queue setup, avoid using AgenCallbackLogin, it's known to cause deadlocked channels. 2.) Restart the Asterisk service once a week. I do this via a CRON job at 3am on Sundays. We're actually not using Agents on our queues, just SIP channels, so hopefully this is not the problem. We simulate 'agents' logging in and out by pausing and unpausing queue members. I am now going to add a cron job to restart asterisk daily, in the hope that until the problem is resolved properly, at least it will help relieve some of the pain by making it stable for a full 24hrs at a time. Matt Florell wrote: I would suggest upgrading to at least 1.4.18. I was able to run it for about 2 weeks and almost one million calls before I could get it to crash, and the 1.4.19RC2 seems to fix even more of the locking issues as well. I know a lot of these problems still existed under 1.4.17. A million calls sounds good, but 2 weeks, not so good. It's a bit disappointing to me that crashing /ever/ is acceptable, I had always had the understanding that asterisk was supposed to be rock-solid. I suppose it's some consolation that its not just me that has problems! Thanks for all the input. I think short term I will restart asterisk daily, then the action plan is to revert back to Debian Etch, and then install asterisk 1.4.18 from source, and hopefully this will improve things. Thanks, Ben ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
I would suggest taking latest 1.4 branch from SVN (or 1.4.19-rc3 when it's out). There has been few deadlocks fixed since rc2. Recompile asterisk with DEBUG_THREADS enabled (in make menuselect), If you're not using safe_asterisk script to start it, you should execute also ulimit -c unlimited before launching asterisk.. When your asterisk is deadlocked, open CLI and execute core show locks. Copy that output, and submit to bugs.digium.com - it will tell developers where exactly is problem. Then, do killall -11 asterisk. It will dump asterisk to core file, and that might provide helpful information later. If your have been requested backtraces, look in /tmp (or in directory you launched asterisk from) for core file. Open that core file with gdb /usr/sbin/asterisk core. and take a dump of thread apply all bt full (make sure you set set pagination off in gdb before this) Regards, Atis On 3/18/08, Norman Franke [EMAIL PROTECTED] wrote: Check around on bugs.digium.com. You'll find a number of issues reported that sound similar. I'm hoping that 1.4.19 will fix a lot of stuff, since the release candidates seem much more stable to me. I couldn't keep Asterisk up for more than a few days before on 1.4.18. I've also applied a few SIP-related patches from various bug reports and things are much, much more stable. 1.4.17, which you mentioned, is also very buggy. 1.4.18 fixed many issues. Norman Franke Answering Service for Directors, Inc. www.myasd.com On Mar 18, 2008, at 7:40 AM, [EMAIL PROTECTED] wrote: We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On Tue, 2008-03-18 at 11:05 -0400, Norman Franke wrote: I've also applied a few SIP-related patches from various bug reports and things are much, much more stable. Mind sharing which patches you have applied? Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On Tue, Mar 18, 2008 at 8:05 AM, Gordon Henderson [EMAIL PROTECTED] wrote: On Tue, 18 Mar 2008, Steve Totaro wrote: Why not try a different OS such as CentOS for now? That would be my next step. I wouldn't suggest chasing distros is the way to solve issues, especially if you're happy with the hardware. Personally, I'd go back to Debian, but stick to stable (Etch) and then compile and install a custom kernel tailored exactly to your hardware, then compile and install your own asterisk from source. But only because that's what I do, and it works for me ... Gordon Well personally, I would go to 1.2.x unless there was some feature in 1.4 that is absolutely needed but the OP said that was not a long term option. I have deployed ONE 1.4 system and that is because I had to, no work arounds due to hardware (unless zaptel 1.4 plays nice with Asterisk 1.2). I will probably continue this train of thought (1.2.X is more production ready) until these threads stop popping up on the list. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On 3/18/08, Ben Willcox [EMAIL PROTECTED] wrote: A million calls sounds good, but 2 weeks, not so good. It's a bit disappointing to me that crashing /ever/ is acceptable, I had always had the understanding that asterisk was supposed to be rock-solid. I suppose it's some consolation that its not just me that has problems! Thanks for all the input. I think short term I will restart asterisk daily, then the action plan is to revert back to Debian Etch, and then install asterisk 1.4.18 from source, and hopefully this will improve things. Keep in mind that my tests go from 0 to 400 calls in about 1 minute then they keep that volume for several hours, and I kept running them for two weeks, and about 6 hours into the last test is when it crashed. I should mention that 1.2.26.2 is what I still use on all of my production servers and they will go for months without a crash. As for rebooting nightly or weekly, that is something we do on a lot of our high-volume servers just to be safe. When pushing Asterisk to high concurrent call volumes it is a good idea to give it a fresh start every day if you can. If Asterisk is being used as a standard office PBX it should be able to run for months with no crashes. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Off-topic note: On Tue, Mar 18, 2008 at 05:45:04PM +0200, Atis Lezdins wrote: If you're not using safe_asterisk script to start it, you should execute also ulimit -c unlimited before launching asterisk.. Without -g (at least on Linux) Asterisk will refuse to generate core dumps. With -g it will generate core files but will also set the ulimit to unlimited. With safe_asterisk you have -g enabled by default, and hence ulimit -c unlimited on by default. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
I believe most of them will be in 1.4.19-rc3 (and in SVN), but I applied patches to 1.4.19-rc2 from: Patches from 11712 and 12098. Plus another one I reported as 12162. Norman Franke Answering Service for Directors, Inc. www.myasd.com On Mar 18, 2008, at 12:11 PM, [EMAIL PROTECTED] wrote: On Tue, 2008-03-18 at 11:05 -0400, Norman Franke wrote: I've also applied a few SIP-related patches from various bug reports and things are much, much more stable. Mind sharing which patches you have applied? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Steve Totaro [EMAIL PROTECTED] writes: I will probably continue this train of thought (1.2.X is more production ready) until these threads stop popping up on the list. I think you're being too kind to 1.2.x. It has numerous problems, most especially with locking in chan_sip. 1.4.x is a HUGE improvement. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On 3/18/08, Benny Amorsen [EMAIL PROTECTED] wrote: Steve Totaro [EMAIL PROTECTED] writes: I will probably continue this train of thought (1.2.X is more production ready) until these threads stop popping up on the list. I think you're being too kind to 1.2.x. It has numerous problems, most especially with locking in chan_sip. 1.4.x is a HUGE improvement. Who uses chan_sip? Long live IAX! :) But seriously, several of my clients use SIP exclusively, passing tens of thousand of calls a day on Asterisk 1.2.X with no issues. I have noticed that the load is slightly lower for SIP-only in 1.4, but I have not noticed any stability issues revolving around SIP on 1.2.X. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Have you tried disabling highpriority=yes in asterisk.conf? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH4G9SDQNt8rg0Kp4RAjIoAKCQEP/e8pR27gbz9p1ilGw8AvWA+wCgs7qX mIrPzDRPWsGt9goKwljsT0Q= =W2og -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users