Re: [asterisk-users] Asterisk in Production ?
Thank you for your very kind offer. After repeatedly re-opening the ticket I finally got a clear specific answer. Strangely, in the 30 mins it took for me to take the answer, try it, and report back the results they had closed the ticket again so I couldn't report whether their solution fixed the problem or not. In fact it did, but I would have liked to have been able to document that so that others running into the same problem and scanning the bug report would know definitively if their answer was indeed correct. But - THANK YOU - and I will Certainly take you up on your most kind offer in the future! Tilghman Lesher wrote: > On Thursday 08 May 2008 23:38:14 Al Baker wrote: > >> Take a big shot of Valium before dealing with the bug tracker folks. >> There idea of "help" is to post "You have an extra space in your line" >> then CLOSE the ticket. >> That kind of clear, specific help is just what my doctor ordered to keep >> my BP nice and low >> > > If you have a problem with one of the explanations, please post the bug > number here, and I'll be happy to explain it in more detail. > > ___ -- 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 in Production ?
On Thursday 08 May 2008 23:38:14 Al Baker wrote: > Take a big shot of Valium before dealing with the bug tracker folks. > There idea of "help" is to post "You have an extra space in your line" > then CLOSE the ticket. > That kind of clear, specific help is just what my doctor ordered to keep > my BP nice and low If you have a problem with one of the explanations, please post the bug number here, and I'll be happy to explain it in more detail. -- Tilghman ___ -- 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 in Production ?
Take a big shot of Valium before dealing with the bug tracker folks. There idea of "help" is to post "You have an extra space in your line" then CLOSE the ticket. That kind of clear, specific help is just what my doctor ordered to keep my BP nice and low Benoit Plessis wrote: > Tilghman Lesher a écrit : > >> On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote: >> >> >>> lordfuknowsyou a écrit : >>> >>> Vinícius Fontes wrote: I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. >>> Any IAX2 phone or mostly SIP ? >>> Do you use Call Queues ? >>> >>> We have less user than that, less concurrent call but quite a few >>> crash/deadlocks >>> >>> >> Have you reported these issues on the bugtracker? >> >> >> > Well, the problem is finding usefull data to report. > > I've 4 core dumps thats show differents things: > > two seems to be related to ControlPlayback: > #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 > #1 0x0809c579 in ast_readframe () > #2 0x0809defc in ast_streamfile () > #3 0x0805e786 in ast_control_streamfile () > #4 0xb698be5c in ?? () from > /usr/lib/asterisk/modules/app_controlplayback.so > #5 0x08298700 in ?? () > #6 0xb470aec0 in ?? () > #7 0xb698c1fc in ?? () from > /usr/lib/asterisk/modules/app_controlplayback.so > #8 0xb698c1fa in ?? () from > /usr/lib/asterisk/modules/app_controlplayback.so > #9 0x in ?? () > > > One is pretty generic: > #0 0x0809c9bc in ast_closestream () > #1 0x08085d91 in ast_hangup () > #2 0x080cd3d8 in pbx_builtin_setvar_helper () > #3 0x080cf08e in ast_pbx_outgoing_exten () > #4 0x080fde65 in ast_inet_ntoa () > #5 0xb7eec560 in start_thread () from /lib/tls/libpthread.so.0 > #6 0xb703667e in clone () from /lib/tls/libc.so.6 > > > and the latest is thread/iax2 related: > #0 0xb7ee71c7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0 > #1 0xb562a969 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so > #2 0x0079 in ?? () > #3 0x in ?? () > #4 0xb547a148 in ?? () > #5 0x080f0508 in ast_sched_add_variable () > #6 0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so > #7 0x0012 in ?? () > > > > But my main problem is when the system just froze, > it start mostly by the Queue not working anymore, with member stuck in > 'in use' stack (should not happen > with IAX2 agent IIRC, given that we had to build macros using GROUP() to > detect in use IAX2 agent) > Then the console (asterisk -rcTvvv) start to freeze (completion doesn't > work, message stop from being displayed > and even command output is lost). > > And i'm reading http://www.asterisk.org/developers/bug-guidelines which > speak of using SVN trunk version of asterisk, > thing i'm not really eager to try on a live system... > > > > > ___ > -- 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 in Production ?
Perhaps this should be tagged under "Is * Ready For Prime Time ?" Thread Isn't an 'appliance' supposed to be a 'plug-it-in-and-runs' sort of thing ? Julian Yap wrote: > On Tue, May 6, 2008 at 1:38 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote: > >> We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 >> and it's quite unstable. >> We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX destroy >> deadlock" >> and now that we have added a Queue, it's worse than ever. The queue goes >> stuck quite often >> (agent are stuck in 'In use' state and if they logoff they can't log-in >> till an asterisk restart). >> > > There's an IAX issue with the security patch for 1.4.18.1... and 1.4.19.1. > > There's another thread on this. > > - Julian > > ___ > -- 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 in Production ?
Tzafrir Cohen a écrit : > On Tue, May 06, 2008 at 09:42:17PM +0200, Benoit Plessis wrote: > >> Tzafrir Cohen a écrit : >> >>> On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote: >>> >>> >>> Here it is, but since the AsteriskNow release has stripped the binary i fear it won't be of much use: >>> Is there any "-debug" package for asterisknow's asterisk package? >>> >>> On RedHat they are generated automatically. On Debian they require some >>> extra settings, and has been present in recent Asterisk packages (the >>> asterisk-dbg package) but not in all of the smaller modules packages. >>> >>> >>> >> Nope, already tried this before posting >> but nothing like that appears on conary >> > > I looked again at http://rbuilder.rpath.com/ and searched for the > package "asterisk". > > It does seem to have a subpackage called "asterisk:debuginfo". > I'm not able to install it but i'll look further, conary is a tricky software to say the least > >> anyway, i'll be migrating on a debian asap, since i now this >> much better and the advantages of AsteriskNow keep reducing >> > > Off topic: > That is not to say you should not try Debian ASAP ;-) > Well i tried a debian/lenny with an mISDN patched for 2.6.24 but it lead to kernel panic / server reboot after 4/5 calls on the B410p. No problem on the T220b but i need both cards ... I think i'll have to reinstall an debian/etch and either try the packaged asterisk 1.2 or manually build an 1.4 + zaptel + misdn. Everything i was looking away from when i initially choosed asteriskNow > To help you with that, here's a live CD: > http://updates.xorcom.com/iso/ > > ___ -- 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 in Production ?
On Tue, May 06, 2008 at 09:42:17PM +0200, Benoit Plessis wrote: > Tzafrir Cohen a écrit : > > On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote: > > > > > >> Here it is, but since the AsteriskNow release has stripped the binary > >> i fear it won't be of much use: > >> > > > > Is there any "-debug" package for asterisknow's asterisk package? > > > > On RedHat they are generated automatically. On Debian they require some > > extra settings, and has been present in recent Asterisk packages (the > > asterisk-dbg package) but not in all of the smaller modules packages. > > > > > Nope, already tried this before posting > but nothing like that appears on conary I looked again at http://rbuilder.rpath.com/ and searched for the package "asterisk". It does seem to have a subpackage called "asterisk:debuginfo". > > anyway, i'll be migrating on a debian asap, since i now this > much better and the advantages of AsteriskNow keep reducing Off topic: That is not to say you should not try Debian ASAP ;-) To help you with that, here's a live CD: http://updates.xorcom.com/iso/ -- 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 in Production ?
On Tue, May 6, 2008 at 1:38 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote: > We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 > and it's quite unstable. > We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX destroy > deadlock" > and now that we have added a Queue, it's worse than ever. The queue goes > stuck quite often > (agent are stuck in 'In use' state and if they logoff they can't log-in > till an asterisk restart). There's an IAX issue with the security patch for 1.4.18.1... and 1.4.19.1. There's another thread on this. - Julian ___ -- 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 in Production ?
Tzafrir Cohen a écrit : > On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote: > > >> Here it is, but since the AsteriskNow release has stripped the binary >> i fear it won't be of much use: >> > > Is there any "-debug" package for asterisknow's asterisk package? > > On RedHat they are generated automatically. On Debian they require some > extra settings, and has been present in recent Asterisk packages (the > asterisk-dbg package) but not in all of the smaller modules packages. > > Nope, already tried this before posting but nothing like that appears on conary anyway, i'll be migrating on a debian asap, since i now this much better and the advantages of AsteriskNow keep reducing as a matter of fact i already now that some thing that doesn't work under AstNow (my siemens sip hardphones, and my SIP provider (Keyyo) at least) work with the debian packaged asterisk. Well for the sip provider it's not that it doesn't work, more than the only way to have some sound is to use the 'm' flag of the Dial() command to have the moh played during the ringing. Given that, i got some sound when the call is established ... -- Benoit ___ -- 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 in Production ?
On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote: > Here it is, but since the AsteriskNow release has stripped the binary > i fear it won't be of much use: Is there any "-debug" package for asterisknow's asterisk package? On RedHat they are generated automatically. On Debian they require some extra settings, and has been present in recent Asterisk packages (the asterisk-dbg package) but not in all of the smaller modules packages. -- 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 in Production ?
Steve Totaro a écrit : > On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote: > >> lordfuknowsyou a écrit : >> >> >>> Vinícius Fontes wrote: >>> >> > >> > I use 1.4.18 with no problems. We have quite a few users(125 total >> > between branches), but the call volume at the most has been around 15 >> > active calls at a time. >> > >> Any IAX2 phone or mostly SIP ? >> Do you use Call Queues ? >> >> We have less user than that, less concurrent call but quite a few >> crash/deadlocks >> >> > > Try SIP only if you can and report back. I think you will confirm > what is pretty much a silent consensus (even among Digium Devs). > > 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 > I've tried SIP only but i already got one 'stuck' Queue member: Members: Local/[EMAIL PROTECTED] with penalty 10 (dynamic) (In use) has taken 1 calls (last was 45 secs ago) Local/[EMAIL PROTECTED] with penalty 20 (dynamic) (Not in use) has taken no calls yet Callers: 1. Zap/10-1 (wait: 0:18, prio: 0) [May 6 17:48:35] NOTICE[2047]: app_queue.c:2152 wait_for_answer: No one is answering queue 'support' (1/0/0) asterix*CLI> core show channels Channel Location State Application(Data) SIP/rtournier-081ef2 (None) Up Bridged Call(Local/[EMAIL PROTECTED] but the other end of the bridged call is long gone ___ -- 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 in Production ?
Tilghman Lesher a écrit : > On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote: > >> Tilghman Lesher a écrit : >> >>> On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote: >>> lordfuknowsyou a écrit : > Vinícius Fontes wrote: > > I use 1.4.18 with no problems. We have quite a few users(125 total > between branches), but the call volume at the most has been around 15 > active calls at a time. > Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlocks >>> Have you reported these issues on the bugtracker? >>> >> Well, the problem is finding usefull data to report. >> >> I've 4 core dumps thats show differents things: >> >> two seems to be related to ControlPlayback: >> #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 >> #1 0x0809c579 in ast_readframe () >> #2 0x0809defc in ast_streamfile () >> #3 0x0805e786 in ast_control_streamfile () >> #4 0xb698be5c in ?? () from >> /usr/lib/asterisk/modules/app_controlplayback.so >> #5 0x08298700 in ?? () >> #6 0xb470aec0 in ?? () >> #7 0xb698c1fc in ?? () from >> /usr/lib/asterisk/modules/app_controlplayback.so >> #8 0xb698c1fa in ?? () from >> /usr/lib/asterisk/modules/app_controlplayback.so >> #9 0x in ?? () >> >> > > I'd love to see a 'bt full' on this one. > Here it is, but since the AsteriskNow release has stripped the binary i fear it won't be of much use: #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 No symbol table info available. #1 0x0809c579 in ast_readframe () No symbol table info available. #2 0x0809defc in ast_streamfile () No symbol table info available. #3 0x0805e786 in ast_control_streamfile () No symbol table info available. #4 0xb698be5c in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #5 0x08298700 in ?? () No symbol table info available. #6 0xb470aec0 in ?? () No symbol table info available. #7 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #8 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #9 0x in ?? () No symbol table info available. #10 0x in ?? () No symbol table info available. #11 0x in ?? () No symbol table info available. #12 0x0bb8 in ?? () No symbol table info available. #13 0x2f727669 in ?? () No symbol table info available. #14 0x65696c63 in ?? () No symbol table info available. #15 0x2f73746e in ?? () No symbol table info available. #16 0x6a6e6f62 in ?? () No symbol table info available. #17 0x2d72756f in ?? () No symbol table info available. #18 0x6e656962 in ?? () No symbol table info available. #19 0x756e6576 in ?? () No symbol table info available. #20 0x6568632d in ?? () No symbol table info available. #21 0x6f702d7a in ?? () No symbol table info available. #22 0x62726577 in ?? () No symbol table info available. #23 0x6974756f in ?? () No symbol table info available. #24 0x2d657571 in ?? () No symbol table info available. #25 0x76726573 in ?? () No symbol table info available. #26 0x73656369 in ?? () No symbol table info available. #27 0x696c632d in ?? () No symbol table info available. #28 0x00746e65 in ?? () No symbol table info available. #29 0x0001 in ?? () No symbol table info available. #30 0xb470af20 in ?? () No symbol table info available. #31 0x081aa084 in ?? () No symbol table info available. #32 0x001b in ?? () No symbol table info available. #33 0x0025 in ?? () No symbol table info available. #34 0x0028 in ?? () No symbol table info available. #35 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #36 0x in ?? () No symbol table info available. #37 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #38 0x0829c4a8 in ?? () No symbol table info available. #39 0x0bb8 in ?? () No symbol table info available. #40 0x in ?? () No symbol table info available. #41 0xb470aec0 in ?? () No symbol table info available. #42 0x in ?? () No symbol table info available. #43 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #44 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #45 0x in ?? () No symbol table info available. #46 0x in ?? () No symbol table info available. #47 0x in ?? () No symbol table info available. #48 0x in ?? () No symbol table info available. #49 0x08298700 in ?? () No symbol table info available. #50 0xb705b631 in strcasecmp () from /lib/tls/libc.so.6 No symbol table info available. #51 0x080c8740 in pbx_substitute_variables_helper () No symbol table info available. #52 0x080cd
Re: [asterisk-users] Asterisk in Production
On May 6, 2008, at 10:20 AM, [EMAIL PROTECTED] wrote: I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. I'm running 1.4.19 and it has been pretty stable. Anything before 1.4.19, however, I found was embarrassingly unstable. I'd often get several crashes within an hour. However, since moving to 19 things have been better. I don't run Queues, though, but I do run a custom derivative of Queues that fixed some bugs and greatly enhanced its usability for us. We do tens of thousands of calls per day (mostly inbound) running on under Debian, although I had to upgrade the kernel to 2.6.23.11 in order to get ztdummy to work on my HP DL380. CPU load remains rather low. We are all SIP, no zaptel. I used to run IAX2 between my three servers (one's a backup and for testing, the other handles desk phones and ATAs), but found IAX2 very, very unreliable. It would hang Asterisk, crash, etc. I just replaced it with SIP (and turned off the module) and those problems went away. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- 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 in Production ?
On Tuesday 06 May 2008 09:02:47 Steve Totaro wrote: > On Tue, May 6, 2008 at 9:35 AM, Tilghman Lesher > > <[EMAIL PROTECTED]> wrote: > > On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote: > > > While these may not be popular opinions, I still ask, what does > > > SwitchVox use? What do some of the guys around here that setup large > > > systems use? Is ABE even using 1.4 yet? > > > > Yes, ABE version C (in release for several months) is using the 1.4 > > codebase. > > Does "In Release" equate to "In the Wild" or "In Many Production > Installations" ? I sense that there are quite a few people who are running version C and a few holdouts still running B, but that's based on a wet-finger-in-the-wind estimation, not on any industry surveys. -- Tilghman ___ -- 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 in Production ?
On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote: > Tilghman Lesher a écrit : > > On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote: > >> lordfuknowsyou a écrit : > >>> Vinícius Fontes wrote: > >>> > >>> I use 1.4.18 with no problems. We have quite a few users(125 total > >>> between branches), but the call volume at the most has been around 15 > >>> active calls at a time. > >> > >> Any IAX2 phone or mostly SIP ? > >> Do you use Call Queues ? > >> > >> We have less user than that, less concurrent call but quite a few > >> crash/deadlocks > > > > Have you reported these issues on the bugtracker? > > Well, the problem is finding usefull data to report. > > I've 4 core dumps thats show differents things: > > two seems to be related to ControlPlayback: > #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 > #1 0x0809c579 in ast_readframe () > #2 0x0809defc in ast_streamfile () > #3 0x0805e786 in ast_control_streamfile () > #4 0xb698be5c in ?? () from > /usr/lib/asterisk/modules/app_controlplayback.so > #5 0x08298700 in ?? () > #6 0xb470aec0 in ?? () > #7 0xb698c1fc in ?? () from > /usr/lib/asterisk/modules/app_controlplayback.so > #8 0xb698c1fa in ?? () from > /usr/lib/asterisk/modules/app_controlplayback.so > #9 0x in ?? () > I'd love to see a 'bt full' on this one. > One is pretty generic: > #0 0x0809c9bc in ast_closestream () > #1 0x08085d91 in ast_hangup () > #2 0x080cd3d8 in pbx_builtin_setvar_helper () > #3 0x080cf08e in ast_pbx_outgoing_exten () > #4 0x080fde65 in ast_inet_ntoa () > #5 0xb7eec560 in start_thread () from /lib/tls/libpthread.so.0 > #6 0xb703667e in clone () from /lib/tls/libc.so.6 Ditto, bt full. > and the latest is thread/iax2 related: > #0 0xb7ee71c7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0 > #1 0xb562a969 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so > #2 0x0079 in ?? () > #3 0x in ?? () > #4 0xb547a148 in ?? () > #5 0x080f0508 in ast_sched_add_variable () > #6 0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so > #7 0x0012 in ?? () > This one may need valgrind to track down the problem, but please be sure to run 1.4.18 or later, as we've already fixed a problem that produced backtraces similar to this. > But my main problem is when the system just froze, > it start mostly by the Queue not working anymore, with member stuck in > 'in use' stack (should not happen > with IAX2 agent IIRC, given that we had to build macros using GROUP() to > detect in use IAX2 agent) > Then the console (asterisk -rcTvvv) start to freeze (completion doesn't > work, message stop from being displayed > and even command output is lost). > > And i'm reading http://www.asterisk.org/developers/bug-guidelines which > speak of using SVN trunk version of asterisk, > thing i'm not really eager to try on a live system... I don't see anywhere on that page that recommends that you try SVN trunk, only the latest SVN (which is probably confusing, but what is meant is to try the latest SVN in the 1.4 branch, which is the release branch. Release candidates and releases are tagged directly off that branch). -- Tilghman ___ -- 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 in Production ?
I'm using 1.4.18 in production on 2 boxes... one of which being a custom built desktop basically, the other being a Dell 1950 III We are in a migration phase to the Dell box, right now the 1st box is doing nothing more than being a PSTN gateway to some FXO lines... basically waiting for numbers to be ported off the analog lines and onto the new T1 which is connected to the Dell box. We have the 2 boxes connected by IAX2 trunk. I had 1.4.19 and 1.4.19.1 running on the Dell box, but it started giving me a lot of trouble with the IAX2 trunk, the trunk would (seemingly) go into UNREACHABLE status and never come back without restarting asterisk (reload, or iax2 reload wouldn’t cut it). Also, occasionally people trying to make outbound calls (and this probably happened on inbound as well), would get a "all circuits are busy" message because of the IAX2 channel driver reporting congestion on the trunk even though it was up (and not congested) Unfortunately as this is a production box I didn’t really have time to try and debug it so I simply downgraded to .18 since it has proven itself well on the 1st box. So far since I;ve downgraded to .18 I haven’t had any problems. Both installs I have running ontop of Gentoo (wouldn’t recommend it if you are new to Linux or don’t like tweak-ability). That all being said, I'll probably give .20 a try when its released, as I see there have been some IAX2 bug fixes in it... but also by the time .20 is released I probably will have retired the box being used as a PSTN gateway and won’t need the IAX2 trunk anymore. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vinícius Fontes Sent: Tuesday, May 06, 2008 8:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk in Production ? There were some really unstable Asterisk releases in the 1.4 branch. I personally use 1.4.13 or 1.4.15 in production. Every single time I tried 1.4.16 or higher I had problems. Att Vinícius Fontes Desenvolvimento Canall Tecnologia em Comunicações Ltda. - "Steve Totaro" <[EMAIL PROTECTED]> escreveu: > On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis <[EMAIL PROTECTED]> > wrote: > > > > Hi, > > > > I'm wondering what version of asterisk people use in production > > environnement ? > > on which distribution ? > > > > And what is your setup like ? > > > > We are actually running an AsteriskNow appliance with asterisk > 1.4.18.1 > > and it's quite unstable. > > We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX > destroy > > deadlock" > > and now that we have added a Queue, it's worse than ever. The queue > goes > > stuck quite often > > (agent are stuck in 'In use' state and if they logoff they can't > log-in > > till an asterisk restart). > > > > > > regards > > > > I am personally a proponent of Asterisk 1.2.X as I see more and more > fatal bugs in the 1.4.X code come up on the lists as well as IAX2 > bugs. I constantly hear "Asterisk 1.4.whatever is much better, but > the bugs coming out are not just unexpected behavior that one could > live with, they are segfaults, system crashes, modules not getting > installed (Zaptel). > > I use SIP since I have seen quite a few issues with IAX2 that were > solved by simply switching to SIP. > > The above two yield solid systems under heavy load for me. OS is not > so important I do not believe. I have some running FC8 and more > running CentOS, both rock solid. I think the general consensus on OS > is use what you are most familiar with. > > While these may not be popular opinions, I still ask, what does > SwitchVox use? What do some of the guys around here that setup large > systems use? Is ABE even using 1.4 yet? All I see in the ABE > release > notes is 1.2 although I have heard that ABE should be running 1.4 > "Very Soon" many many moons ago > http://www.digium.com/en/docs/ABE/README . So either Digium doesn't > trust 1.4 enough to use it for ABE or the README is out of date. > > 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 ___ -- 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 in Production ?
Hello, Our company did 200+ installations around the globe and had no issues with stability with correct Asterisk version. We used most of 1.4. As far as I remember 1.4.16 has some nasty bugs along with 1.4.19.x (SIP + realtime). So current stable is 1.4.18.1 (for us). For load check: http://wiki.kolmisoft.com/index.php/How_fast_MOR_can_perform It shows how our billing application performs on top of Asterisk (2049 channels) and we can push it even further with some improvements. We DO NOT RESTART our Asterisk installations daily or weekly. They work for months. Regards, Mindaugas Kezys http://www.kolmisoft.com > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Benoit Plessis > Sent: Tuesday, May 06, 2008 2:39 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Asterisk in Production ? > > > Hi, > > I'm wondering what version of asterisk people use in production > environnement ? > on which distribution ? > > And what is your setup like ? > > We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 > and it's quite unstable. > We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX destroy > deadlock" > and now that we have added a Queue, it's worse than ever. The queue > goes > stuck quite often > (agent are stuck in 'In use' state and if they logoff they can't log-in > till an asterisk restart). > > > regards > > ___ > -- 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 in Production ?
Tilghman Lesher a écrit : > On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote: > >> lordfuknowsyou a écrit : >> >>> Vinícius Fontes wrote: >>> >>> I use 1.4.18 with no problems. We have quite a few users(125 total >>> between branches), but the call volume at the most has been around 15 >>> active calls at a time. >>> >> Any IAX2 phone or mostly SIP ? >> Do you use Call Queues ? >> >> We have less user than that, less concurrent call but quite a few >> crash/deadlocks >> > > Have you reported these issues on the bugtracker? > > Well, the problem is finding usefull data to report. I've 4 core dumps thats show differents things: two seems to be related to ControlPlayback: #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 #1 0x0809c579 in ast_readframe () #2 0x0809defc in ast_streamfile () #3 0x0805e786 in ast_control_streamfile () #4 0xb698be5c in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #5 0x08298700 in ?? () #6 0xb470aec0 in ?? () #7 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #8 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #9 0x in ?? () One is pretty generic: #0 0x0809c9bc in ast_closestream () #1 0x08085d91 in ast_hangup () #2 0x080cd3d8 in pbx_builtin_setvar_helper () #3 0x080cf08e in ast_pbx_outgoing_exten () #4 0x080fde65 in ast_inet_ntoa () #5 0xb7eec560 in start_thread () from /lib/tls/libpthread.so.0 #6 0xb703667e in clone () from /lib/tls/libc.so.6 and the latest is thread/iax2 related: #0 0xb7ee71c7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0 #1 0xb562a969 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #2 0x0079 in ?? () #3 0x in ?? () #4 0xb547a148 in ?? () #5 0x080f0508 in ast_sched_add_variable () #6 0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #7 0x0012 in ?? () But my main problem is when the system just froze, it start mostly by the Queue not working anymore, with member stuck in 'in use' stack (should not happen with IAX2 agent IIRC, given that we had to build macros using GROUP() to detect in use IAX2 agent) Then the console (asterisk -rcTvvv) start to freeze (completion doesn't work, message stop from being displayed and even command output is lost). And i'm reading http://www.asterisk.org/developers/bug-guidelines which speak of using SVN trunk version of asterisk, thing i'm not really eager to try on a live system... ___ -- 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 in Production ?
We are using Asterisk 1.4.13 in production, were we have almost 30 SIP users on a Asterisk box, we are also using IAX to communicate between main Asterisk server and the other. we use Queues, Conference too. Regards, Sanjay Rajdev - Original Message - From: "Benoit Plessis" <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com Sent: Tuesday, May 6, 2008 5:08:37 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: [asterisk-users] Asterisk in Production ? Hi, I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX destroy deadlock" and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). regards ___ -- 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 in Production ?
Steve Totaro a écrit : > On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote: > >> Any IAX2 phone or mostly SIP ? >> Do you use Call Queues ? >> >> We have less user than that, less concurrent call but quite a few >> crash/deadlock > > Try SIP only if you can and report back. I think you will confirm > what is pretty much a silent consensus (even among Digium Devs). > Hi, that's what i was planning seeing all thoses answers. We initialy choosed IAX2 for the sendurl() support but i'll set-up a test periode in SIP-only to compare. > Thanks, > Steve Totaro > Thanks to you ___ -- 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 in Production ?
Steve Totaro wrote: >> I use 1.4.18 with no problems. We have quite a few users(125 total >> between branches), but the call volume at the most has been around 15 >> active calls at a time. >> >> >> > > I would classify that as "Light to Medium Call Volume" or "SMB". > > Let me clarify what I consider "High Call Volume". ~400 simultaneous > calls, all SIP or 95 on a box doing quad PRI to SIP gateway duty. > 15k+ calls a day lasting an average of fifteen minutes. > > 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 > > I agree that it is SMB, never said I was a telco ;] just in production on 1.4.18. we do use sip,iax2 and pri. Our calls do last extended periods of time, especially when there are conferences. No call ques, and we do realtime voicemail,sip and iax to allow tennants web interfaces into the system through the standard 3 tiers. ___ -- 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 in Production ?
Benoit Plessis wrote: > lordfuknowsyou a écrit : > >> Vinícius Fontes wrote: >> >> I use 1.4.18 with no problems. We have quite a few users(125 total >> between branches), but the call volume at the most has been around 15 >> active calls at a time. >> >> > Any IAX2 phone or mostly SIP ? > Do you use Call Queues ? > > We have less user than that, less concurrent call but quite a few > crash/deadlocks > > > ___ > -- 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 We use SIP and IAX2, we also do fax 2 email using spandsp and rx/txfax. I did have a problem with libpri during the upgrade and had to roll back to the one I was using prior. hth ___ -- 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 in Production ?
On Tue, May 06, 2008 at 10:01:54AM -0400, Jay R. Ashworth wrote: > On Tue, May 06, 2008 at 01:38:37PM +0200, Benoit Plessis wrote: > > I'm wondering what version of asterisk people use in production > > environnement ? on which distribution ? > > > > And what is your setup like ? > > Well, we're running a cluster of about 15 boxes or so with Slack 10 or > 12 and 1.2.17(?, either 14 or 17) and VICIdial. Yeah, you'd call it > production. :-) Sorry: we're running zap channels at all the edges (Digium and Sangoma quad-T cards, primarily), and IAX2 in the middle. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth & Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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 in Production ?
Tilghman Lesher wrote: > On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote: > >> All I see in the ABE release >> notes is 1.2 although I have heard that ABE should be running 1.4 >> "Very Soon" many many moons ago >> http://www.digium.com/en/docs/ABE/README . So either Digium doesn't >> trust 1.4 enough to use it for ABE or the README is out of date. >> > > The first clue should be that the copyright listed in that file is from 2006. > Yes, it's very much out of date. > > Fix it! Beat some of those tech writers into submission! N. ___ -- 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 in Production ?
On Tue, May 6, 2008 at 9:35 AM, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote: > > While these may not be popular opinions, I still ask, what does > > SwitchVox use? What do some of the guys around here that setup large > > systems use? Is ABE even using 1.4 yet? > > Yes, ABE version C (in release for several months) is using the 1.4 codebase. > > > > All I see in the ABE release > > notes is 1.2 although I have heard that ABE should be running 1.4 > > "Very Soon" many many moons ago > > http://www.digium.com/en/docs/ABE/README . So either Digium doesn't > > trust 1.4 enough to use it for ABE or the README is out of date. > > The first clue should be that the copyright listed in that file is from 2006. > Yes, it's very much out of date. > > -- > Tilghman > Does "In Release" equate to "In the Wild" or "In Many Production Installations" ? 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 in Production ?
On Tue, May 06, 2008 at 01:38:37PM +0200, Benoit Plessis wrote: > I'm wondering what version of asterisk people use in production > environnement ? on which distribution ? > > And what is your setup like ? Well, we're running a cluster of about 15 boxes or so with Slack 10 or 12 and 1.2.17(?, either 14 or 17) and VICIdial. Yeah, you'd call it production. :-) It runs, knock on Formica-laminated particle board, pretty well. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth & Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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 in Production ?
On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote: > lordfuknowsyou a écrit : > > > Vinícius Fontes wrote: > > > > I use 1.4.18 with no problems. We have quite a few users(125 total > > between branches), but the call volume at the most has been around 15 > > active calls at a time. > > > Any IAX2 phone or mostly SIP ? > Do you use Call Queues ? > > We have less user than that, less concurrent call but quite a few > crash/deadlocks > Try SIP only if you can and report back. I think you will confirm what is pretty much a silent consensus (even among Digium Devs). 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 in Production ?
Mostly SIP, some of my clients have queues and everything is working fine by now. Att Vinícius Fontes Desenvolvimento Canall Tecnologia em Comunicações Ltda. - "Benoit Plessis" <[EMAIL PROTECTED]> escreveu: > lordfuknowsyou a écrit : > > Vinícius Fontes wrote: > > > > I use 1.4.18 with no problems. We have quite a few users(125 total > > between branches), but the call volume at the most has been around > 15 > > active calls at a time. > > > Any IAX2 phone or mostly SIP ? > Do you use Call Queues ? > > We have less user than that, less concurrent call but quite a few > crash/deadlocks > > > ___ > -- 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 in Production ?
On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote: > While these may not be popular opinions, I still ask, what does > SwitchVox use? What do some of the guys around here that setup large > systems use? Is ABE even using 1.4 yet? Yes, ABE version C (in release for several months) is using the 1.4 codebase. > All I see in the ABE release > notes is 1.2 although I have heard that ABE should be running 1.4 > "Very Soon" many many moons ago > http://www.digium.com/en/docs/ABE/README . So either Digium doesn't > trust 1.4 enough to use it for ABE or the README is out of date. The first clue should be that the copyright listed in that file is from 2006. Yes, it's very much out of date. -- Tilghman ___ -- 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 in Production ?
On Tue, 2008-05-06 at 07:58 -0400, Steve Totaro wrote: [snip] > While these may not be popular opinions, I still ask, what does > SwitchVox use? Not sure what Asterisk version they use but I saw (iirc) a presentation on their website that they run switchvox on top of Fedora Core 6. FC6 has been end-of-line for a long, long time... 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 in Production ?
On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote: > lordfuknowsyou a écrit : > > Vinícius Fontes wrote: > > > > I use 1.4.18 with no problems. We have quite a few users(125 total > > between branches), but the call volume at the most has been around 15 > > active calls at a time. > > Any IAX2 phone or mostly SIP ? > Do you use Call Queues ? > > We have less user than that, less concurrent call but quite a few > crash/deadlocks Have you reported these issues on the bugtracker? -- Tilghman ___ -- 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 in Production ?
> I use 1.4.18 with no problems. We have quite a few users(125 total > between branches), but the call volume at the most has been around 15 > active calls at a time. > > I would classify that as "Light to Medium Call Volume" or "SMB". Let me clarify what I consider "High Call Volume". ~400 simultaneous calls, all SIP or 95 on a box doing quad PRI to SIP gateway duty. 15k+ calls a day lasting an average of fifteen minutes. 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 in Production ?
lordfuknowsyou a écrit : > Vinícius Fontes wrote: > > I use 1.4.18 with no problems. We have quite a few users(125 total > between branches), but the call volume at the most has been around 15 > active calls at a time. > Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlocks ___ -- 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 in Production ?
Vinícius Fontes wrote: > There were some really unstable Asterisk releases in the 1.4 branch. I > personally use 1.4.13 or 1.4.15 in production. Every single time I tried > 1.4.16 or higher I had problems. > > > > Att > Vinícius Fontes > Desenvolvimento > Canall Tecnologia em Comunicações Ltda. > > - "Steve Totaro" <[EMAIL PROTECTED]> escreveu: > > >> On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis <[EMAIL PROTECTED]> >> wrote: >> >>> Hi, >>> >>> I'm wondering what version of asterisk people use in production >>> environnement ? >>> on which distribution ? >>> >>> And what is your setup like ? >>> >>> We are actually running an AsteriskNow appliance with asterisk >>> >> 1.4.18.1 >> >>> and it's quite unstable. >>> We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX >>> >> destroy >> >>> deadlock" >>> and now that we have added a Queue, it's worse than ever. The queue >>> >> goes >> >>> stuck quite often >>> (agent are stuck in 'In use' state and if they logoff they can't >>> >> log-in >> >>> till an asterisk restart). >>> >>> >>> regards >>> >>> >> I am personally a proponent of Asterisk 1.2.X as I see more and more >> fatal bugs in the 1.4.X code come up on the lists as well as IAX2 >> bugs. I constantly hear "Asterisk 1.4.whatever is much better, but >> the bugs coming out are not just unexpected behavior that one could >> live with, they are segfaults, system crashes, modules not getting >> installed (Zaptel). >> >> I use SIP since I have seen quite a few issues with IAX2 that were >> solved by simply switching to SIP. >> >> The above two yield solid systems under heavy load for me. OS is not >> so important I do not believe. I have some running FC8 and more >> running CentOS, both rock solid. I think the general consensus on OS >> is use what you are most familiar with. >> >> While these may not be popular opinions, I still ask, what does >> SwitchVox use? What do some of the guys around here that setup large >> systems use? Is ABE even using 1.4 yet? All I see in the ABE >> release >> notes is 1.2 although I have heard that ABE should be running 1.4 >> "Very Soon" many many moons ago >> http://www.digium.com/en/docs/ABE/README . So either Digium doesn't >> trust 1.4 enough to use it for ABE or the README is out of date. >> >> 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 >> > > ___ > -- 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 I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. ___ -- 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 in Production ?
There were some really unstable Asterisk releases in the 1.4 branch. I personally use 1.4.13 or 1.4.15 in production. Every single time I tried 1.4.16 or higher I had problems. Att Vinícius Fontes Desenvolvimento Canall Tecnologia em Comunicações Ltda. - "Steve Totaro" <[EMAIL PROTECTED]> escreveu: > On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis <[EMAIL PROTECTED]> > wrote: > > > > Hi, > > > > I'm wondering what version of asterisk people use in production > > environnement ? > > on which distribution ? > > > > And what is your setup like ? > > > > We are actually running an AsteriskNow appliance with asterisk > 1.4.18.1 > > and it's quite unstable. > > We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX > destroy > > deadlock" > > and now that we have added a Queue, it's worse than ever. The queue > goes > > stuck quite often > > (agent are stuck in 'In use' state and if they logoff they can't > log-in > > till an asterisk restart). > > > > > > regards > > > > I am personally a proponent of Asterisk 1.2.X as I see more and more > fatal bugs in the 1.4.X code come up on the lists as well as IAX2 > bugs. I constantly hear "Asterisk 1.4.whatever is much better, but > the bugs coming out are not just unexpected behavior that one could > live with, they are segfaults, system crashes, modules not getting > installed (Zaptel). > > I use SIP since I have seen quite a few issues with IAX2 that were > solved by simply switching to SIP. > > The above two yield solid systems under heavy load for me. OS is not > so important I do not believe. I have some running FC8 and more > running CentOS, both rock solid. I think the general consensus on OS > is use what you are most familiar with. > > While these may not be popular opinions, I still ask, what does > SwitchVox use? What do some of the guys around here that setup large > systems use? Is ABE even using 1.4 yet? All I see in the ABE > release > notes is 1.2 although I have heard that ABE should be running 1.4 > "Very Soon" many many moons ago > http://www.digium.com/en/docs/ABE/README . So either Digium doesn't > trust 1.4 enough to use it for ABE or the README is out of date. > > 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 ___ -- 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 in Production ?
On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote: > > Hi, > > I'm wondering what version of asterisk people use in production > environnement ? > on which distribution ? > > And what is your setup like ? > > We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 > and it's quite unstable. > We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX destroy > deadlock" > and now that we have added a Queue, it's worse than ever. The queue goes > stuck quite often > (agent are stuck in 'In use' state and if they logoff they can't log-in > till an asterisk restart). > > > regards > I am personally a proponent of Asterisk 1.2.X as I see more and more fatal bugs in the 1.4.X code come up on the lists as well as IAX2 bugs. I constantly hear "Asterisk 1.4.whatever is much better, but the bugs coming out are not just unexpected behavior that one could live with, they are segfaults, system crashes, modules not getting installed (Zaptel). I use SIP since I have seen quite a few issues with IAX2 that were solved by simply switching to SIP. The above two yield solid systems under heavy load for me. OS is not so important I do not believe. I have some running FC8 and more running CentOS, both rock solid. I think the general consensus on OS is use what you are most familiar with. While these may not be popular opinions, I still ask, what does SwitchVox use? What do some of the guys around here that setup large systems use? Is ABE even using 1.4 yet? All I see in the ABE release notes is 1.2 although I have heard that ABE should be running 1.4 "Very Soon" many many moons ago http://www.digium.com/en/docs/ABE/README . So either Digium doesn't trust 1.4 enough to use it for ABE or the README is out of date. 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
[asterisk-users] Asterisk in Production ?
Hi, I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX destroy deadlock" and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). regards ___ -- 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 in production as a fax server, anyone?
Paulo Scardine wrote: I have a worst issue for you... If your fax solution is ever going to receive fax in Brazil, how would you block collect calls? I have made a fax server solution with cheap Digium hardware that works in Brazil (2 E1s). -- Paulo, He is mentioning E1/PRI, so I assume the well known "collect call on E1/R2 thingie" doesn't apply to him. Adolfo R. Brandes escreveu: Greetings, All-Knowing Asterisk Users List, My company needs to build a reliable fax server that can handle at least 30 simultaneous incoming faxes from the PSTN, using PRI. We realize that this can be solved in any number of ways using a Linux box, but since IVR is also a must, Asterisk popped up as the most promising solution. After combing these lists for clues, we began experimenting extensively with Asterisk and its software DSP and fax capabilities in most of their incarnations, such as Rxfax or Iaxmodem/Hylafax, together with Digium's E1 cards in server-grade Intel motherboards, all in a dedicated test environment. Unfortunately, though, we have yet to achieve reliable and satisfactory results, even with only 1 fax call at a time. I won't go into the details because we don't need technical support, given that this is, as of yet, a very loosely defined test. What we want is is merely a pointer in the right direction. So here it comes: Has anybody ever achieved, or know of someone who has, reliable 30 simultaneous PRI fax calls using Asterisk and Asterisk-compatible hardware and software? We are hardware agnostic, so if you say Sangoma's cards do it better than Digium's, or that Eicon Diva cards' hardware DSP and chan_capi are the only solution, we have no problem going there. I would be most thankful, however, for detailed explanations of successful scenarios, including such things as motherboard make and model, processor speed, Linux distribution and version, and anything else you decide to be even marginally pertinent. ___ --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
Re: [Asterisk-Users] Asterisk in production as a fax server, anyone?
I have a worst issue for you... If your fax solution is ever going to receive fax in Brazil, how would you block collect calls? I have made a fax server solution with cheap Digium hardware that works in Brazil (2 E1s). -- Paulo Adolfo R. Brandes escreveu: Greetings, All-Knowing Asterisk Users List, My company needs to build a reliable fax server that can handle at least 30 simultaneous incoming faxes from the PSTN, using PRI. We realize that this can be solved in any number of ways using a Linux box, but since IVR is also a must, Asterisk popped up as the most promising solution. After combing these lists for clues, we began experimenting extensively with Asterisk and its software DSP and fax capabilities in most of their incarnations, such as Rxfax or Iaxmodem/Hylafax, together with Digium's E1 cards in server-grade Intel motherboards, all in a dedicated test environment. Unfortunately, though, we have yet to achieve reliable and satisfactory results, even with only 1 fax call at a time. I won't go into the details because we don't need technical support, given that this is, as of yet, a very loosely defined test. What we want is is merely a pointer in the right direction. So here it comes: Has anybody ever achieved, or know of someone who has, reliable 30 simultaneous PRI fax calls using Asterisk and Asterisk-compatible hardware and software? We are hardware agnostic, so if you say Sangoma's cards do it better than Digium's, or that Eicon Diva cards' hardware DSP and chan_capi are the only solution, we have no problem going there. I would be most thankful, however, for detailed explanations of successful scenarios, including such things as motherboard make and model, processor speed, Linux distribution and version, and anything else you decide to be even marginally pertinent. Thank you very much, Adolfo R. Brandes ___ --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
Re: [Asterisk-Users] Asterisk in production as a fax server, anyone?
Julio Arruda escreveu: Paulo, He is mentioning E1/PRI, so I assume the well known "collect call on E1/R2 thingie" doesn't apply to him. Julio, I have 1 E1 from telefonica and 1 from Embratel. Telefonica has a better deal for incoming calls (gave us more DIDs) but Embratel has better rates. I've had a real hard time trying to make E1/PRI signaling work with Embratel, with no success. In the end, I had to use MFC/5C. Telefonica and Embratel will not block collect calls for you, they dont care, its easy money. May be he is linking to a smaller and more flexible telco, or may be he will put the * box behind another PBX that has better support for MFC/5C than libmfcr2. I'm just curious anyway. The automated collect call system in Brazil is really dumb and unfair, and is abused so many ways... I want to beat the crap out of the genius who invented this system where the callee does not have to explicitly accept a collect call. Anatel (the telco government agency in Brazil) dont even acknowledge this as problem, because they will not accept complaints against Anatel regulations, just against the telcos, and the telcos are following this dumb rules to the letter. Its because regulatory agencies in Brazil are here not to protect the citizens, just to extort money from private companies to burn in our corrupt political engine. Sorry for the rant, but I would like to hear from other people running * in Brazil, how they address this trouble. -- Paulo ___ --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
Re: [Asterisk-Users] Asterisk in production as a fax server, anyone?
Olivier Krief wrote: What would be an acceptable zttest result to avoid fax frame slippings ? A result of 100% would be acceptable. :-) I believe that 99.8% seems to be also acceptable... at least according to popular concensus. However, 92% is very bad, and at that result you would have a very hard time getting faxes through. Lee. ___ --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
Re: [Asterisk-Users] Asterisk in production as a fax server, anyone?
2006/3/30, Lee Howard <[EMAIL PROTECTED]>: However, based on the comments you give I'd suspect that you're havingwhat people seem to be calling "frame slipping". There seem to be somemotherboards that react poorly with Zap cards (or their respective drivers) and cause that. Your zttest results should be revealing here.I don't know that anyone has yet proposed a conclusive solution to thatmatter other than to keep trying different motherboards until you find one that works.What would be an acceptable zttest result to avoid fax frame slippings ? ___ --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
Re: [Asterisk-Users] Asterisk in production as a fax server, anyone?
Adolfo R. Brandes wrote: After combing these lists for clues, we began experimenting extensively with Asterisk and its software DSP and fax capabilities in most of their incarnations, such as Rxfax or Iaxmodem/Hylafax, together with Digium's E1 cards in server-grade Intel motherboards, all in a dedicated test environment. Unfortunately, though, we have yet to achieve reliable and satisfactory results, even with only 1 fax call at a time. Well, if you weren't just trying to test you could send some logs to an appropriate mailing list (like iaxmodem-users or whatever would be most appropriate), and you'd probably get a very quick indication as to why. However, based on the comments you give I'd suspect that you're having what people seem to be calling "frame slipping". There seem to be some motherboards that react poorly with Zap cards (or their respective drivers) and cause that. Your zttest results should be revealing here. I don't know that anyone has yet proposed a conclusive solution to that matter other than to keep trying different motherboards until you find one that works. I won't go into the details because we don't need technical support, given that this is, as of yet, a very loosely defined test. What we want is is merely a pointer in the right direction. So here it comes: Has anybody ever achieved, or know of someone who has, reliable 30 simultaneous PRI fax calls using Asterisk and Asterisk-compatible hardware and software? Yes. The concurrent calls really isn't that big of a deal, either, if those are your thoughts. The bigger issue seems to be the quality of the audio as it is delivered to the fax application/modem. We are hardware agnostic, so if you say Sangoma's cards do it better than Digium's, or that Eicon Diva cards' hardware DSP and chan_capi are the only solution, we have no problem going there. In my experience Sangoma offers better (or maybe just more caring) post-sale support than does Digium, however, the "frame slipping" issue seems to happen with both sets of hardware. Maybe that's because Sangomas partially use the zaptel driver as well as does the Digium hardware. As for Eicon Divas being used for IVR, I've never used them that way. I would be most thankful, however, for detailed explanations of successful scenarios, including such things as motherboard make and model, processor speed, Linux distribution and version, and anything else you decide to be even marginally pertinent. The most success I've seen has been to bridge the call through Asterisk to a T1 fax modem such as a Patton 2977 or an Eicon Diva Server with HylaFAX running the modems. (So you put a crossover cable between the T1/E1 fax modem and one of the ports on your TE405P card and bridge the fax call, after IVR, to the fax port and let HylaFAX take it from there.) That said, I see no reason why you couldn't have just as much success with iaxmodem-hylafax or even txfax/rxfax... I just haven't personally used either of those as much as I have the T1-faxmodem solution. Lee. ___ --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
Re: [Asterisk-Users] Asterisk in production as a fax server, anyone?
Hi Adolfo, I have done this and it works. I have maxed out an E1 with 30 concurrent calls of which at least 25 would have been fax. Hardware is nothing special, Dell Poweredge 750, 512mb ram, single SATA drive with either of a TE410p or TE110p card. OS is FC2 with kernel 2.6.9 I expect the server would handle 60 concurrent calls. Asterisk is 1.0.10 with spandsp 0.0.2pre25 and libtiff 3.5.7 Email me privately if you want more details. Craig - Original Message - From: "Adolfo R. Brandes" <[EMAIL PROTECTED]> To: Sent: Thursday, March 30, 2006 10:20 PM Subject: [Asterisk-Users] Asterisk in production as a fax server, anyone? Greetings, All-Knowing Asterisk Users List, My company needs to build a reliable fax server that can handle at least 30 simultaneous incoming faxes from the PSTN, using PRI. We realize that this can be solved in any number of ways using a Linux box, but since IVR is also a must, Asterisk popped up as the most promising solution. After combing these lists for clues, we began experimenting extensively with Asterisk and its software DSP and fax capabilities in most of their incarnations, such as Rxfax or Iaxmodem/Hylafax, together with Digium's E1 cards in server-grade Intel motherboards, all in a dedicated test environment. Unfortunately, though, we have yet to achieve reliable and satisfactory results, even with only 1 fax call at a time. I won't go into the details because we don't need technical support, given that this is, as of yet, a very loosely defined test. What we want is is merely a pointer in the right direction. So here it comes: Has anybody ever achieved, or know of someone who has, reliable 30 simultaneous PRI fax calls using Asterisk and Asterisk-compatible hardware and software? We are hardware agnostic, so if you say Sangoma's cards do it better than Digium's, or that Eicon Diva cards' hardware DSP and chan_capi are the only solution, we have no problem going there. I would be most thankful, however, for detailed explanations of successful scenarios, including such things as motherboard make and model, processor speed, Linux distribution and version, and anything else you decide to be even marginally pertinent. Thank you very much, Adolfo R. Brandes ___ --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
[Asterisk-Users] Asterisk in production as a fax server, anyone?
Greetings, All-Knowing Asterisk Users List, My company needs to build a reliable fax server that can handle at least 30 simultaneous incoming faxes from the PSTN, using PRI. We realize that this can be solved in any number of ways using a Linux box, but since IVR is also a must, Asterisk popped up as the most promising solution. After combing these lists for clues, we began experimenting extensively with Asterisk and its software DSP and fax capabilities in most of their incarnations, such as Rxfax or Iaxmodem/Hylafax, together with Digium's E1 cards in server-grade Intel motherboards, all in a dedicated test environment. Unfortunately, though, we have yet to achieve reliable and satisfactory results, even with only 1 fax call at a time. I won't go into the details because we don't need technical support, given that this is, as of yet, a very loosely defined test. What we want is is merely a pointer in the right direction. So here it comes: Has anybody ever achieved, or know of someone who has, reliable 30 simultaneous PRI fax calls using Asterisk and Asterisk-compatible hardware and software? We are hardware agnostic, so if you say Sangoma's cards do it better than Digium's, or that Eicon Diva cards' hardware DSP and chan_capi are the only solution, we have no problem going there. I would be most thankful, however, for detailed explanations of successful scenarios, including such things as motherboard make and model, processor speed, Linux distribution and version, and anything else you decide to be even marginally pertinent. Thank you very much, Adolfo R. Brandes ___ --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
Re: [Asterisk-Users] Asterisk in Production
On 9/28/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: The machine has 1 GB RAM. When I boot themachine fresh with asterisk, it uses approximately 80 MB of RAM whenI run top. After 10 days, top shows that it's using 730 MB of RAM. Are you sure it is asterisk that is using all of that memory? Are you reading a line that says something like: Mem: 971236k total, 321576k used, 649660k free, 81552k buffers If yes, you may not be interpreting it right.-- "We hold [...] that all men are created equal; that they areendowed [...] with certain inalienable rights; that amongthese are life, liberty, and the pursuit of happiness" -- Thomas Jefferson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk in Production
I was reading on the wiki different possibilities of automatically restarting asterisk every so often. In some places, people mention they restart it once a day other on shorter or longer intervals. I believe the main reason people are doing this is because of possible memory leaks. I'm running a system for IVR services. It's not a heavily loaded box, but there is almost always someone using the system. I issued a restart when convenient but it's been 10 days and it hasn't restarted. I wonder if it's really necessary to restart asterisk? Are there really memory leaks? The machine has 1 GB RAM. When I boot the machine fresh with asterisk, it uses approximately 80 MB of RAM when I run top. After 10 days, top shows that it's using 730 MB of RAM. Then I wonder, what would happen if this was a busy system? Would I be forced into having to restart asterisk and potentially dropping all active calls? What are other people doing? I asked a question about whether there are people out there offering vonage-like services running on top of asterisk and I received several responses. I assume that if there is some sort of traffic on those boxes, they may be suffering from similar symptoms. What are the recommendations for maintaining a production asterisk system and these "potential bugs" that cause memory leaks? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
