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lordfuknowsyou at gmai...
Guest

Posted: Tue May 06, 2008 9:16 am Post subject: [asterisk-users] Asterisk in Production ?

Benoit Plessis wrote:

Quote:

lordfuknowsyou a ?crit :

Quote:

Vin?cius 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

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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

lordfuknowsyou at gmai...
Guest

Posted: Tue May 06, 2008 9:19 am Post subject: [asterisk-users] Asterisk in Production ?

Steve Totaro wrote:

Quote:

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

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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.

benoit at plessis.info
Guest

Posted: Tue May 06, 2008 9:19 am Post subject: [asterisk-users] Asterisk in Production ?

Steve Totaro a ?crit :

Quote:

On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <benoit at plessis.info> wrote:

Quote:

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.

Quote:

Thanks,
Steve Totaro

Thanks to you

sanjay.rajdev at feath...
Guest

Posted: Tue May 06, 2008 9:29 am Post subject: [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" <benoit at plessis.info> To: asterisk-users at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080506/08a42b89/attachment.htm

benoit at plessis.info
Guest

Posted: Tue May 06, 2008 9:30 am Post subject: [asterisk-users] Asterisk in Production ?

Tilghman Lesher a ?crit :

Quote:

On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:

Quote:

lordfuknowsyou a ?crit :

Quote:

Vin?cius 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 0x00000000 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 0x00000079 in ?? ()
#3 0x00000000 in ?? ()
#4 0xb547a148 in ?? ()
#5 0x080f0508 in ast_sched_add_variable ()
#6 0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so
#7 0x00000012 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...

mkezys at gmail.com
Guest

Posted: Tue May 06, 2008 9:33 am Post subject: [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

Quote:

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
bounces at lists.digium.com] On Behalf Of Benoit Plessis
Sent: Tuesday, May 06, 2008 2:39 PM
To: asterisk-users at 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

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mwatson at becon.org
Guest

Posted: Tue May 06, 2008 9:44 am Post subject: [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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Vin?cius 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
Vin?cius Fontes
Desenvolvimento
Canall Tecnologia em Comunica??es Ltda.

----- "Steve Totaro" <stotaro at totarotechnologies.com> escreveu:

Quote:

On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis <benoit at plessis.info>
wrote:

Quote:

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

Quote:

and it's quite unstable.
We have ~30 IAX2 SoftPhones and encounter some "Avoiding IAX

destroy

Quote:

deadlock"
and now that we have added a Queue, it's worse than ever. The queue

goes

Quote:

stuck quite often
(agent are stuck in 'In use' state and if they logoff they can't

log-in

Quote:

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

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tilghman at mail.jeffa...
Guest

Posted: Tue May 06, 2008 9:54 am Post subject: [asterisk-users] Asterisk in Production ?

On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote:

Quote:

Tilghman Lesher a ?crit :

Quote:

On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:

Quote:

lordfuknowsyou a ?crit :

Quote:

Vin?cius 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?

I'd love to see a 'bt full' on this one.

Quote:

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.

Quote:

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 0x00000079 in ?? ()
#3 0x00000000 in ?? ()
#4 0xb547a148 in ?? ()
#5 0x080f0508 in ast_sched_add_variable ()
#6 0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so
#7 0x00000012 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.

Quote:

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

tilghman at mail.jeffa...
Guest

Posted: Tue May 06, 2008 10:01 am Post subject: [asterisk-users] Asterisk in Production ?

On Tuesday 06 May 2008 09:02:47 Steve Totaro wrote:

Quote:

On Tue, May 6, 2008 at 9:35 AM, Tilghman Lesher

<tilghman at mail.jeffandtilghman.com> wrote:

Quote:

On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote:

Quote:

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

benoit at plessis.info
Guest

Posted: Tue May 06, 2008 10:37 am Post subject: [asterisk-users] Asterisk in Production ?

Tilghman Lesher a ?crit :

Quote:

On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote:

Quote:

Tilghman Lesher a ?crit :

Quote:

On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:

Quote:

lordfuknowsyou a ?crit :

Quote:

Vin?cius 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?

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 0x00000000 in ?? ()
No symbol table info available.
#10 0x00000000 in ?? ()
No symbol table info available.
#11 0x00000000 in ?? ()
No symbol table info available.
#12 0x00000bb8 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 0x00000001 in ?? ()
No symbol table info available.
#30 0xb470af20 in ?? ()
No symbol table info available.
#31 0x081aa084 in ?? ()
No symbol table info available.
#32 0x0000001b in ?? ()
No symbol table info available.
#33 0x00000025 in ?? ()
No symbol table info available.
#34 0x00000028 in ?? ()
No symbol table info available.
#35 0xb698c1fc in ?? () from
/usr/lib/asterisk/modules/app_controlplayback.so
No symbol table info available.
#36 0x00000000 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 0x00000bb8 in ?? ()
No symbol table info available.
#40 0x00000000 in ?? ()
No symbol table info available.
#41 0xb470aec0 in ?? ()
No symbol table info available.
#42 0x00000000 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 0x00000000 in ?? ()
No symbol table info available.
#46 0x00000000 in ?? ()
No symbol table info available.
#47 0x00000000 in ?? ()
No symbol table info available.
#48 0x00000000 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 0x080cd170 in pbx_builtin_setvar_helper ()
No symbol table info available.
#53 0x080cf08e in ast_pbx_outgoing_exten ()
No symbol table info available.
#54 0x080fde65 in ast_inet_ntoa ()
No symbol table info available.
#55 0xb7f61560 in start_thread () from /lib/tls/libpthread.so.0
No symbol table info available.
#56 0xb70ab67e in clone () from /lib/tls/libc.so.6
No symbol table info available.

Quote:

Ditto, bt full.

#0 0x0809c9bc in ast_closestream ()
No symbol table info available.
#1 0x08085d91 in ast_hangup ()
No symbol table info available.
#2 0x080cd3d8 in pbx_builtin_setvar_helper ()
No symbol table info available.
#3 0x080cf08e in ast_pbx_outgoing_exten ()
No symbol table info available.
#4 0x080fde65 in ast_inet_ntoa ()
No symbol table info available.
#5 0xb7eec560 in start_thread () from /lib/tls/libpthread.so.0
No symbol table info available.
#6 0xb703667e in clone () from /lib/tls/libc.so.6
No symbol table info available.

Quote:

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.

Quote:

Ok, yes it has confused me

benoit at plessis.info
Guest

Posted: Tue May 06, 2008 10:51 am Post subject: [asterisk-users] Asterisk in Production ?

Steve Totaro a ?crit :

Quote:

On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <benoit at plessis.info> wrote:

Quote:

lordfuknowsyou a ?crit :

Quote:

Vin?cius 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

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I've tried SIP only but i already got one 'stuck' Queue member:
Members:
Local/136 at queues with penalty 10 (dynamic) (In use) has taken 1
calls (last was 45 secs ago)
Local/888 at queues 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/136 at queues-

but the other end of the bridged call is long gone

tzafrir.cohen at xorco...
Guest

Posted: Tue May 06, 2008 2:06 pm Post subject: [asterisk-users] Asterisk in Production ?

On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:

Quote:

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:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir

benoit at plessis.info
Guest

Posted: Tue May 06, 2008 2:42 pm Post subject: [asterisk-users] Asterisk in Production ?

Tzafrir Cohen a ?crit :

Quote:

On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:

Quote:

Here it is, but since the AsteriskNow release has stripped the binary
i fear it won't be of much use:

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

julianokyap at gmail.com
Guest

Posted: Tue May 06, 2008 3:06 pm Post subject: [asterisk-users] Asterisk in Production ?

On Tue, May 6, 2008 at 1:38 AM, Benoit Plessis <benoit at plessis.info> wrote:

Quote:

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

tzafrir.cohen at xorco...
Guest

Posted: Wed May 07, 2008 5:35 am Post subject: [asterisk-users] Asterisk in Production ?

On Tue, May 06, 2008 at 09:42:17PM +0200, Benoit Plessis wrote:

Quote:

Tzafrir Cohen a ?crit :

Quote:

On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:

Quote:

Here it is, but since the AsteriskNow release has stripped the binary
i fear it won't be of much use:

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".

Quote:

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 Wink

To help you with that, here's a live CD:
http://updates.xorcom.com/iso/

--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir

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