[asterisk-users] Asterisk crashes when storing voicemail via odbc

2017-06-20 Thread Mike Diehl
Hi all,

I'm working on migrating all of my servers to store voicemail in a mysql 
database via 
odbc.

I've got a development server that I can reconfigure and test at will.  When 
it's 
configured to store vm on the file system, it seems to be rock solid.

However, when I ONLY change it to store vm in the database, it becomes very 
unstable.  

Here's what it's doing.  When I attempt leave a voicemail, I am prompted to 
leave a 
message.  Once I have left a message, the console locks up and I have to 
killall -9 to get 
it to restart and become responsive again.

I'm running Asterisk 13.14.0 built by root @ server on a x86_64 running Linux 
on 
2017-06-20 14:27:06 UTC

For odbc, I've got unixODBC 2.3.2-r2.

Are these the versions I should be using?  If so, any recommendations as to how 
to 
troubleshoot this would be most welcome.

TIA,

-- 
Mike Diehl


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Re: [asterisk-users] Asterisk Crashes Randomly with Cepstral Swift TTS

2014-10-19 Thread Patrick Laimbock
Hi Bryan,

On 10/18/2014 11:47 PM, Bryan Burroughs wrote:
 All,
 
 Has anyone seen this before? This appears to be a Swift or app_swift
 bug. I'm having a difficult time finding any information or support on this.

I haven't used app_Swift with Cepstral but iirc it wasn't deemed very
stable.

 Asterisk version:
 Asterisk 11.6-cert4 built by asterisk @ ivrd02 on a x86_64 running Linux
 on 2014-08-11 13:55:25 UTC
 OS:
 Linux livrp03 2.6.32-431.11.2.el6.x86_64 #1 SMP Mon Mar 3 13:32:45 EST
 2014 x86_64 x86_64 x86_64 GNU/Linux

If you are not tied to the certified Asterisk version then perhaps try
using the latest Asterisk version (currently 11.13.0).

 When Asterisk crashes, the backtrace always looks something like the
 following:

[snip]

 The out of bounds line looks like it may be pointing to the issue.
 
 *argv = {0x0, 0xb9b0 Address 0xb9b0 out of
 bounds, 0x0}*

Have you tried contacting the app_swift developer and/or filed a bug at
https://issues.asterisk.org/jira/secure/Dashboard.jspa ?

 Should I look into using another TTS engine?

You could try UniMRCP which sits between Asterisk and Cepstral replacing
app_swift: http://unimrcp.org

HTH,
Patrick

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Re: [asterisk-users] Asterisk Crashes Randomly with Cepstral Swift TTS

2014-10-19 Thread Bryan Burroughs
Do you or anyone have any experience using Swift with UniMRCP? I tried a 
few months ago but gave up. The documentation is not very clear.


Yes, I am reaching out to the developer of app_swift.

thanks,

Bryan Burroughs

On 10/19/2014 08:02 AM, Patrick Laimbock wrote:

Hi Bryan,

On 10/18/2014 11:47 PM, Bryan Burroughs wrote:

All,

Has anyone seen this before? This appears to be a Swift or app_swift
bug. I'm having a difficult time finding any information or support on this.

I haven't used app_Swift with Cepstral but iirc it wasn't deemed very
stable.


Asterisk version:
Asterisk 11.6-cert4 built by asterisk @ ivrd02 on a x86_64 running Linux
on 2014-08-11 13:55:25 UTC
OS:
Linux livrp03 2.6.32-431.11.2.el6.x86_64 #1 SMP Mon Mar 3 13:32:45 EST
2014 x86_64 x86_64 x86_64 GNU/Linux

If you are not tied to the certified Asterisk version then perhaps try
using the latest Asterisk version (currently 11.13.0).


When Asterisk crashes, the backtrace always looks something like the
following:

[snip]


The out of bounds line looks like it may be pointing to the issue.

*argv = {0x0, 0xb9b0 Address 0xb9b0 out of
bounds, 0x0}*

Have you tried contacting the app_swift developer and/or filed a bug at
https://issues.asterisk.org/jira/secure/Dashboard.jspa ?


Should I look into using another TTS engine?

You could try UniMRCP which sits between Asterisk and Cepstral replacing
app_swift: http://unimrcp.org

HTH,
Patrick




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[asterisk-users] Asterisk Crashes Randomly with Cepstral Swift TTS

2014-10-18 Thread Bryan Burroughs

All,

Has anyone seen this before? This appears to be a Swift or app_swift 
bug. I'm having a difficult time finding any information or support on this.


Asterisk version:
Asterisk 11.6-cert4 built by asterisk @ ivrd02 on a x86_64 running Linux 
on 2014-08-11 13:55:25 UTC

OS:
Linux livrp03 2.6.32-431.11.2.el6.x86_64 #1 SMP Mon Mar 3 13:32:45 EST 
2014 x86_64 x86_64 x86_64 GNU/Linux


When Asterisk crashes, the backtrace always looks something like the 
following:



[Thread debugging using libthread_db enabled]
Core was generated by `/opt/asterisk/sbin/asterisk -f -C 
/opt/asterisk/etc/asterisk/asterisk.c'.

Program terminated with signal 11, Segmentation fault.
#0  0x7f79439ba061 in LM_set_port_number () from 
/opt/swift/lib/libswift.so.6
#0  0x7f79439ba061 in LM_set_port_number () from 
/opt/swift/lib/libswift.so.6

No symbol table info available.
#1  0x7f79439bb7c6 in LM_init () from /opt/swift/lib/libswift.so.6
No symbol table info available.
#2  0x7f79439c5f0d in swift_engine_open () from 
/opt/swift/lib/libswift.so.6

No symbol table info available.
#3  0x7f7943bf9b6e in app_exec (chan=0x7f7914009e08, 
data=0x7f7928f9c680 413 E 3RD ST) at app_swift.c:370

res = 0
max_digits = 0
timeout = 0
alreadyran = 0
ms = 0
len = 6043056
availatend = 0
argv = {0x0, 0xb9b0 Address 0xb9b0 out 
of bounds, 0x0}

text = 0x7f7928f9a230 413 E 3RD ST
rc = 0x0
tmp_exten = \000
results = '\000' repeats 19 times
u = 0x7f7974001dc0
f = 0x7f7928f9a4c0
next = {tv_sec = 6014649, tv_usec = 21020457123253}
ps = 0x7f79740165e0
parse = 0x7f7928f9a230 413 E 3RD ST
old_writeformat = {id = 687449040, fattr = {format_attr = 
{32633, 5184330, 0, 687448768, 32633, 5710841, 0, 48, 48, 687448992, 
32633, 687448800, 32633, 1946162544, 0, 1946162544, 32633, 8448136, 0, 
84, 0, 4294967292, 0, 687449104, 32633, 5827993, 0, 84, 0, 1717986919, 
1717986918, 8448135, 0, 5623124, 0, 687449040, 32633, 5630801, 0, 9, 0, 
687449040, 0, 687449104, 32633, 40, 0, 1413666990, 0, 5222777, 0, 
1946239199, 32633, 687449192, 32633, 80, 0, 0, 0, 2, 0, 5187840, 0, 
1946207160}, rtp_marker_bit = 121 'y'}}
args = {argc = 1, argv = 0x7f7928f9a280, text = 0x7f7928f9a230 
413 E 3RD ST, timeout = 0x0, max_digits = 0x0}
myf = {f = {frametype = 0, subclass = {integer = 0, format = 
{id = 0, fattr = {format_attr = {0, 0, 0, 0, 257, 0, 2, 0, 1839600224, 
32767, 645521943, 1069709714, 4, 0, 0, 0, 1839600224, 32767, 687448144, 
32633, 127525399, 3225199326, 3012944407, 1069709582, 1, 0 repeats 33 
times, 1839600016, 32767, 2, 0, 34150224, 0}, rtp_marker_bit = 0 
'\000'}}}, datalen = 0, samples = 97009376, mallocd = 0, mallocd_hdr_len 
= 177, offset = 1, src = 0x7f7928f9a050 \207, incomplete sequence 
\350\200, data = {ptr = 0x0, uint32 = 0, pad = 
\000\000\000\000\000\000\000}, delivery = {tv_sec = 0, tv_usec = 0}, 
frame_list = {next = 0x0}, flags = 0, ts = 0, len = 0, seqno = 0}, 
offset = 
\000\000\000\000\000\000\000\000\310\314\335\005\000\000\000\000\310\314\335\005, 
'\000' repeats 20 times, 
`\177\316\005\000\000\000\000\377\377\000\000\001\000\000\000\000\000\000\000\000\000\000


The out of bounds line looks like it may be pointing to the issue.

*argv = {0x0, 0xb9b0 Address 0xb9b0 out of 
bounds, 0x0}*


Should I look into using another TTS engine?

thanks,

--
Bryan Burroughs

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[asterisk-users] Asterisk crashes when reloading configs...

2014-07-02 Thread Carlos Chavez
I am having a very strange problem.  We use Asterisk 11.X (have 
tried several versions, including certified) which reads its config 
files in realtime from a SQLITE3 database.  Everything runs fine but 
lately asterisk has been crashing when we issue a reload command via 
Manager.  Our web interface uses AMI to reload the dialplan and right 
after it does that ( I can see the results on the CLI) asterisk 
crashes.  This does not seem to happen every time but some days it 
crashes often.  Any ideas where to start looking for the problem?


--
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Carlos Chávez
+52 (55)9116-91161


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Re: [asterisk-users] Asterisk crashes when reloading configs...

2014-07-02 Thread Matthew Jordan
On Wed, Jul 2, 2014 at 12:38 PM, Carlos Chavez cur...@telecomabmex.com wrote:
 I am having a very strange problem.  We use Asterisk 11.X (have tried
 several versions, including certified) which reads its config files in
 realtime from a SQLITE3 database.  Everything runs fine but lately asterisk
 has been crashing when we issue a reload command via Manager.  Our web
 interface uses AMI to reload the dialplan and right after it does that ( I
 can see the results on the CLI) asterisk crashes.  This does not seem to
 happen every time but some days it crashes often.  Any ideas where to start
 looking for the problem?


Please get a backtrace illustrating the problem:

https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

Once you have a properly generated backtrace, open an issue on
issues.asterisk.org.

Thanks -

Matt

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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[asterisk-users] Asterisk crashes suddenly

2014-05-28 Thread Daniel - Asterisk
Hello friends,

I have been experienced suddenly stops for my Asterisk server, I do not why
is it happening. Asterisk's debug messages only tell me I have lacked g729
codec for translation to one peer minutes before the crashes occur

[2014-05-27 09:48:30] WARNING[15384][C-017c] channel.c: Unable to find
a codec translation path from (ulaw) to (g729)
[2014-05-27 09:48:30] WARNING[15384][C-017c] chan_sip.c: Asked to
transmit frame type g729, while native formats is (ulaw) read/write =
ulaw/ulaw
[2014-05-27 09:48:30] WARNING[15384][C-017c] chan_sip.c: Asked to
transmit frame type ulaw, while native formats is (g729) read/write =
ulaw/slin
[2014-05-27 09:48:30] WARNING[15384][C-017c] channel.c: Codec mismatch
on channel SIP/20108-0051 setting write format to g729 from ulaw native
formats (ulaw)

And it stops after a failed attended transfer between two of my SIP peers.

[2014-05-27 09:48:32] WARNING[15384][C-017c] channel.c: No path to
translate from SIP/20108-0051 to SIP/30201-0052
[2014-05-27 09:48:32] WARNING[15384][C-017c] channel.c: Can't make
SIP/20108-0051 and SIP/30201-0052 compatible
[2014-05-27 09:48:32] WARNING[15384][C-017c] features.c: Bridge failed
on channels SIP/20108-0051 and SIP/30201-0052
There are no more messages from Asterisk but I have found this message on
kern.log, messages  and syslog.1 on /var/log/:
May 27 09:48:32 pbx-thor-PE kernel: [334427.888524] asterisk[15384] general
protection ip:482a13 sp:7f335b87c898 error:0 in asterisk[40+221000]

I am using Debian 7.5 64 bits with Asterisk 11.9.0

Thank you!

Elder D. Arohuanca
Lima - Peru
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Re: [asterisk-users] Asterisk crashes suddenly

2014-05-28 Thread Sander Smeenk
Quoting Daniel - Asterisk (earohua...@gmail.com):

 And it stops after a failed attended transfer between two of my SIP peers.
 May 27 09:48:32 pbx-thor-PE kernel: [334427.888524] asterisk[15384] general
 protection ip:482a13 sp:7f335b87c898 error:0 in asterisk[40+221000]
 I am using Debian 7.5 64 bits with Asterisk 11.9.0

Have you checked for core dumps?  AFAIK, Debian packages create core
dumps on segfaults in /tmp(?). You could use that to further pin point
the issue, perhaps.

$ gdb /usr/sbin/asterisk /tmp/something.core
gdb bt full

If it is reproducable, try recompiling Asterisk with debug symbols to
get a clearer backtrace of what happened.

-Sndr.
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Re: [asterisk-users] Asterisk crashes at meetme kick all

2014-02-18 Thread Deka, Rajib IN MAA SL
Thanks a lot Patrick.

Regards
Rajib Deka
Siemens Ltd.

--

Message: 7
Date: Mon, 17 Feb 2014 10:22:02 +0100
From: Patrick Laimbock patr...@laimbock.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk crashes at meetme kick all
Message-ID: 5301d4ba.3040...@laimbock.com
Content-Type: text/plain; charset=windows-1252; format=flowed

On 17-02-14 09:26, Deka, Rajib IN MAA SL wrote:
 Dear Forum,
 I have encountered a similar issue as below in Asterisk 10.0.0. 
 Asterisk crashed while executing ?meetme kick all? CLI command from 
 manager interface. The link says the issue has been closed however I 
 am not able to identify in which release of asterisk this issue has been 
 fixed.
 Please help.
 _https://issues.asterisk.org/jira/browse/ASTERISK-15741_

AFAICT this issue has not been fixed due to inactivity. Note the Suspended due 
to lack of activity remark. Also the 1.6 version mentioned in the bugreport is 
EOL. Version 10.0.0 you mentioned is also EOL so any bugreport you file against 
version 10.0.0 will not be acted upon unless you can reproduce it with the 
latest Asterisk version 11.x.x or 12.x.x.

I recommend you upgrade to an Asterisk LTS (long term support) version like the 
latest 11.x.x (currently 11.7.0). For more information about Asterisk LTS 
versions go to:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

If you still see a bug when running Asterisk 11.x.x (or 12.x.x) you can report 
it at the Asterisk issue tracker at:

https://issues.asterisk.org/jira/secure/Dashboard.jspa

Before filing a bug please read the information at:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Howtoreportabug

--
Patrick




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[asterisk-users] Asterisk crashes at meetme kick all

2014-02-17 Thread Deka, Rajib IN MAA SL
Dear Forum,

I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk 
crashed while executing meetme kick all CLI command from manager interface. 
The link says the issue has been closed however I am not able to identify in 
which release of asterisk this issue has been fixed. Please help.

https://issues.asterisk.org/jira/browse/ASTERISK-15741


With best regards,
Rajib Deka
Siemens Ltd.


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Re: [asterisk-users] Asterisk crashes at meetme kick all

2014-02-17 Thread Patrick Laimbock

On 17-02-14 09:26, Deka, Rajib IN MAA SL wrote:

Dear Forum,
I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk
crashed while executing “meetme kick all” CLI command from manager
interface. The link says the issue has been closed however I am not able
to identify in which release of asterisk this issue has been fixed.
Please help.
_https://issues.asterisk.org/jira/browse/ASTERISK-15741_


AFAICT this issue has not been fixed due to inactivity. Note the 
Suspended due to lack of activity remark. Also the 1.6 version 
mentioned in the bugreport is EOL. Version 10.0.0 you mentioned is also 
EOL so any bugreport you file against version 10.0.0 will not be acted 
upon unless you can reproduce it with the latest Asterisk version 11.x.x 
or 12.x.x.


I recommend you upgrade to an Asterisk LTS (long term support) version 
like the latest 11.x.x (currently 11.7.0). For more information about 
Asterisk LTS versions go to:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

If you still see a bug when running Asterisk 11.x.x (or 12.x.x) you can 
report it at the Asterisk issue tracker at:


https://issues.asterisk.org/jira/secure/Dashboard.jspa

Before filing a bug please read the information at:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Howtoreportabug

--
Patrick


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Re: [asterisk-users] Asterisk crashes on high IO load

2011-04-05 Thread Thorsten Göllner

Did you take a look at
/var/log/syslog
/var/log/asterisk/messages
?

Using Debian? Take a look at iotop (apt-get install iotop). There you 
can see information about which process consumes high io load.


Am 04.04.2011 17:23, schrieb Maximilian Grobecker:

Hello Thorsten,

the system has 4 GB RAM and about 2,5 GB free so swap space is not used
or exhausted.
Maybe the high load is not cause of this crashes but it's the only thing
the crashes can be reproduced with.


Thank you!

Maximilian Grobecker


Am 04.04.2011 16:03, schrieb Thorsten Göllner:

Take a look with top at your system when high io load is seen. Maybe
the machine is running out of ram and starts swapping?

Am 04.04.2011 15:04, schrieb Maximilian Grobecker:

Hi!

I'm writing to this list because I've got a very confusing issue with
our Asterisk 1.8.3.2 installation.

On high IO load on the hard drives Asterisk becomes instable and crashes
after a few minutes.
I tried to reproduce this by running bonnie++ on the hardware while
making calls.
The calls didn't get disturbed (no noises or crackles) but after about
five minutes Asterisk suddenly crashed without any further error
messages.


Are you experiencing the same problem?
I'm really confused now why Asterisk crashes...


Thank you!
Maximilian Grobecker

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OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf

Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54


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Re: [asterisk-users] Asterisk crashes on high IO load

2011-04-05 Thread Maximilian Grobecker
Hi,

the log files contained (sometimes) lines about refcount -1 in astobj.c.
I also generated core dumps and analyzed them - but there were always
errors in another module.

Mabye I found the solution:
Asterisk seems to crash when a required module cannot be loaded fast
enough due to heavy disk usage.
When I move the modules directory to another hard disk Asterisk runs fine.

I'm using autoload=yes in modules.conf and have several noload lines
in it. Is there a possibility to say asterisk to load all modules to RAM
at start time and not on demand?


Thanks and greetings
Max



Am 05.04.2011 09:45, schrieb Thorsten Göllner:
 Did you take a look at
 /var/log/syslog
 /var/log/asterisk/messages
 ?
 
 Using Debian? Take a look at iotop (apt-get install iotop). There you
 can see information about which process consumes high io load.
 
 Am 04.04.2011 17:23, schrieb Maximilian Grobecker:
 Hello Thorsten,

 the system has 4 GB RAM and about 2,5 GB free so swap space is not used
 or exhausted.
 Maybe the high load is not cause of this crashes but it's the only thing
 the crashes can be reproduced with.


 Thank you!

 Maximilian Grobecker


 Am 04.04.2011 16:03, schrieb Thorsten Göllner:
 Take a look with top at your system when high io load is seen. Maybe
 the machine is running out of ram and starts swapping?

 Am 04.04.2011 15:04, schrieb Maximilian Grobecker:
 Hi!

 I'm writing to this list because I've got a very confusing issue with
 our Asterisk 1.8.3.2 installation.

 On high IO load on the hard drives Asterisk becomes instable and
 crashes
 after a few minutes.
 I tried to reproduce this by running bonnie++ on the hardware while
 making calls.
 The calls didn't get disturbed (no noises or crackles) but after about
 five minutes Asterisk suddenly crashed without any further error
 messages.


 Are you experiencing the same problem?
 I'm really confused now why Asterisk crashes...


 Thank you!
 Maximilian Grobecker

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Re: [asterisk-users] Asterisk crashes on high IO load

2011-04-05 Thread Matt Riddell

On 6/04/11 12:39 AM, Maximilian Grobecker wrote:

Hi,

the log files contained (sometimes) lines about refcount -1 in astobj.c.
I also generated core dumps and analyzed them - but there were always
errors in another module.

Mabye I found the solution:
Asterisk seems to crash when a required module cannot be loaded fast
enough due to heavy disk usage.
When I move the modules directory to another hard disk Asterisk runs fine.

I'm using autoload=yes in modules.conf and have several noload lines
in it. Is there a possibility to say asterisk to load all modules to RAM
at start time and not on demand?


You could compile Asterisk with embedded modules?

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[asterisk-users] Asterisk crashes on high IO load

2011-04-04 Thread Maximilian Grobecker
Hi!

I'm writing to this list because I've got a very confusing issue with
our Asterisk 1.8.3.2 installation.

On high IO load on the hard drives Asterisk becomes instable and crashes
after a few minutes.
I tried to reproduce this by running bonnie++ on the hardware while
making calls.
The calls didn't get disturbed (no noises or crackles) but after about
five minutes Asterisk suddenly crashed without any further error messages.


Are you experiencing the same problem?
I'm really confused now why Asterisk crashes...


Thank you!
Maximilian Grobecker

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Re: [asterisk-users] Asterisk crashes on high IO load

2011-04-04 Thread Thorsten Göllner
Take a look with top at your system when high io load is seen. Maybe 
the machine is running out of ram and starts swapping?


Am 04.04.2011 15:04, schrieb Maximilian Grobecker:

Hi!

I'm writing to this list because I've got a very confusing issue with
our Asterisk 1.8.3.2 installation.

On high IO load on the hard drives Asterisk becomes instable and crashes
after a few minutes.
I tried to reproduce this by running bonnie++ on the hardware while
making calls.
The calls didn't get disturbed (no noises or crackles) but after about
five minutes Asterisk suddenly crashed without any further error messages.


Are you experiencing the same problem?
I'm really confused now why Asterisk crashes...


Thank you!
Maximilian Grobecker

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Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54


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Re: [asterisk-users] Asterisk crashes on high IO load

2011-04-04 Thread Maximilian Grobecker
Hello Thorsten,

the system has 4 GB RAM and about 2,5 GB free so swap space is not used
or exhausted.
Maybe the high load is not cause of this crashes but it's the only thing
the crashes can be reproduced with.


Thank you!

Maximilian Grobecker


Am 04.04.2011 16:03, schrieb Thorsten Göllner:
 Take a look with top at your system when high io load is seen. Maybe
 the machine is running out of ram and starts swapping?
 
 Am 04.04.2011 15:04, schrieb Maximilian Grobecker:
 Hi!

 I'm writing to this list because I've got a very confusing issue with
 our Asterisk 1.8.3.2 installation.

 On high IO load on the hard drives Asterisk becomes instable and crashes
 after a few minutes.
 I tried to reproduce this by running bonnie++ on the hardware while
 making calls.
 The calls didn't get disturbed (no noises or crackles) but after about
 five minutes Asterisk suddenly crashed without any further error
 messages.


 Are you experiencing the same problem?
 I'm really confused now why Asterisk crashes...


 Thank you!
 Maximilian Grobecker

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Re: [asterisk-users] Asterisk crashes on high IO load

2011-04-04 Thread Nic Colledge
Are you using IAX? There are some problems causing crashes for us related to 
laggyness on IAX channels with 1.8 versions. 

There are a bunch of problems with IAX related to 
https://issues.asterisk.org/view.php?id=17521

Nic.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Maximilian 
Grobecker
Sent: 04 April 2011 16:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk crashes on high IO load

Hello Thorsten,

the system has 4 GB RAM and about 2,5 GB free so swap space is not used
or exhausted.
Maybe the high load is not cause of this crashes but it's the only thing
the crashes can be reproduced with.


Thank you!

Maximilian Grobecker


Am 04.04.2011 16:03, schrieb Thorsten Göllner:
 Take a look with top at your system when high io load is seen. Maybe
 the machine is running out of ram and starts swapping?
 
 Am 04.04.2011 15:04, schrieb Maximilian Grobecker:
 Hi!

 I'm writing to this list because I've got a very confusing issue with
 our Asterisk 1.8.3.2 installation.

 On high IO load on the hard drives Asterisk becomes instable and crashes
 after a few minutes.
 I tried to reproduce this by running bonnie++ on the hardware while
 making calls.
 The calls didn't get disturbed (no noises or crackles) but after about
 five minutes Asterisk suddenly crashed without any further error
 messages.


 Are you experiencing the same problem?
 I'm really confused now why Asterisk crashes...


 Thank you!
 Maximilian Grobecker

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-08-08 Thread Manmohan Singh Jandu
Hi Dan,

I was trying to make the Invite working. I am getting following error
when i try to make a call.

[Aug  8 16:55:22] NOTICE[15082]: chan_local.c:534 local_call: No such
extension/context 73...@default while calling Local channel
[Aug  8 16:55:22] NOTICE[15082]: channel.c:4042
__ast_request_and_dial: Unable to call channel Local/73281
[Aug  8 16:55:22] ERROR[12166]: pbx.c:9301 device_state_cb: Received
invalid event that had no device IE
[Aug  8 16:55:22] ERROR[12166]: app_queue.c:1099 device_state_cb:
Received invalid event that had no device IE

Following is my dialplan in /etc/asterisk/extensions.conf

[outgoing]
exten = _73...,1,Dial(SIP/callman02SIP/callman01/${EXTEN:2})
exten = _73...,n,Congestion

following is in lib/defines.php

//Outcall defaults
define (CHAN_TYPE, Local); //Use Local to let dialplan decide which chan
define (OUT_CONTEXT, outgoing); //Select a context to place the call from
define (OUT_PEER, ); // Use this if not using CHAN_TYPE Local
define (OUT_CALL_CID, Parlez 1996); // Caller ID for Invites

--Manmohan Singh
On Fri, Aug 6, 2010 at 12:46 AM, Dan Austin dan_aus...@phoenix.com wrote:
 Manmohan wrote:
 I commented locale.php in defines.php and it perfectly worked well.

 Now i am wondering what is this invite participants for, while adding
 conference. wherein it asks for first name, lastname, emailaddress 
 telephone number..
 The 'Invite Others' option is mostly for installs that do not have
 a consistent e-mail environment, and are using the SERVER mailer.
 This feature lets the server send invite emails to multiple parties.
 In my environments we have Exchange and Outlook, so I prefer the CLIENT
 mailer, and I can manage the invitations in my mail client

 Let me brief you how i had done this setup. I had created a SIP trunk
 between Cisco Call manager and Asterisk server. And i am using webmeetme
 for Audio conferencing.
 Sounds familiar.  I put this package together after wasting too much
 money and time trying to make an expensive Cisco conferencing solution
 work.

 Other than the invite participants, while the conf call is going on we
 get couple of more options, when we click to the current ongoing conference
 number.

 End call -- To end the conference call
 Yes

 Extend -- I am sure this is to extend the time of the call for which its
 scheduled for, but not sure on how much time does it extends by default
 OR is there any way to define the customized time on whatever required.
 10 minutes is the default.  I thought I had made it configurable in 
 lib/defines.php,
 but no I have it hard coded in conf_add (to be fixed in the next release now).
 You can search for +600 and change it to any value you like.

 Invite-- When i click this button it asks me telephone number. I assume this
 is any number which asterisk server can reach as per the dialplan configured
 in extension.conf in /etc/asterisk.. Though this invite button looks pretty
 much interesting to use but whenever i enter any phone number it says
 System error not sure if am understand this wrongly.
 You understand it correctly, but the default settings are likely not working.
 Check out the section 'Outcall defaults' in lib/defines.php.  It is likely you
 need to change the OUT_CONTEXT at a minimum.

 Dan

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-08-05 Thread Manmohan Singh Jandu
Hi Dan,

I commented locale.php in defines.php and it perfectly worked well.

Now i am wondering what is this invite participants for, while adding
conference. wherein it asks for first name, lastname, emailaddress 
telephone number..

Let me brief you how i had done this setup. I had created a SIP trunk
between Cisco Call manager and Asterisk server. And i am using webmeetme for
Audio conferencing.

Other than the invite participants, while the conf call is going on we get
couple of more options, when we click to the current ongoing conference
number.

End call -- To end the conference call
Extend -- I am sure this is to extend the time of the call for which its
scheduled for, but not sure on how much time does it extends by default OR
is there any way to define the customized time on whatever required.
Invite-- When i click this button it asks me telephone number. I assume this
is any number which asterisk server can reach as per the dialplan configured
in extension.conf in /etc/asterisk.. Though this invite button looks pretty
much interesting to use but whenever i enter any phone number it says
System error not sure if am understand this wrongly.

Please correct me if i am wrong.

--Manmohan Singh

On Wed, Aug 4, 2010 at 9:14 PM, Dan Austin dan_aus...@phoenix.com wrote:

 Manmohan wrote:
  I had tried the new version of webmeetme i.e., 4.0.2
  The recording works very well.
 Great!

  I see following php errors whenever i try to add in conference.

  [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice:
  Undefined variable: order in /var/www/html/web-meetme/meetme_control.php
  on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4
 You can ignore the Notices.  They are fairly harmless, and only mean that
 variable is not set by the code or being passed in on the URL.  You can
 turn off notices in /etc/php.ini if they bother you.

  Also the Reports link doesnt display anything and in httpd error logs it
 gives me following php errors:
  [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning:
  include(locale.php) [a href='function.include'function.include/a]:
  failed to open stream: No such file or directory in
 /var/www/html/web-meetme/lib/defines.php
  on line 3, referer: http://10.1.1.30/web-meetme/daily.php?

 In lib/defines.php, either comment out the 3rd line or add ../ before
 locale.php-
include(../locale.php);

 But that is not likely why you do not get the reports.  The most likely
 cause is
 A PHP notice is being thrown while the GD code is rendering the graph,
 resulting in
 a corrupt image which your browser cannot display.

 Check these settings /etc/php.ini-
 error_reporting  =  E_ALL
 display_errors = Off

 Dan

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Manmohan Singh Jandu
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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-08-05 Thread Dan Austin
Manmohan wrote:
 I commented locale.php in defines.php and it perfectly worked well.

 Now i am wondering what is this invite participants for, while adding 
 conference. wherein it asks for first name, lastname, emailaddress  
 telephone number..
The 'Invite Others' option is mostly for installs that do not have
a consistent e-mail environment, and are using the SERVER mailer.
This feature lets the server send invite emails to multiple parties.
In my environments we have Exchange and Outlook, so I prefer the CLIENT
mailer, and I can manage the invitations in my mail client

 Let me brief you how i had done this setup. I had created a SIP trunk
 between Cisco Call manager and Asterisk server. And i am using webmeetme
 for Audio conferencing.
Sounds familiar.  I put this package together after wasting too much
money and time trying to make an expensive Cisco conferencing solution
work.

 Other than the invite participants, while the conf call is going on we
 get couple of more options, when we click to the current ongoing conference
 number.

 End call -- To end the conference call
Yes

 Extend -- I am sure this is to extend the time of the call for which its
 scheduled for, but not sure on how much time does it extends by default 
 OR is there any way to define the customized time on whatever required.
10 minutes is the default.  I thought I had made it configurable in 
lib/defines.php,
but no I have it hard coded in conf_add (to be fixed in the next release now).
You can search for +600 and change it to any value you like.

 Invite-- When i click this button it asks me telephone number. I assume this 
 is any number which asterisk server can reach as per the dialplan configured
 in extension.conf in /etc/asterisk.. Though this invite button looks pretty 
 much interesting to use but whenever i enter any phone number it says 
 System error not sure if am understand this wrongly.
You understand it correctly, but the default settings are likely not working.
Check out the section 'Outcall defaults' in lib/defines.php.  It is likely you
need to change the OUT_CONTEXT at a minimum.

Dan

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-08-04 Thread Manmohan Singh Jandu
Hi Dan,

I had tried the new version of webmeetme i.e., 4.0.2
The recording works very well.

I see following php errors whenever i try to add in conference.

[Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice:
Undefined variable: order in /var/www/html/web-meetme/meetme_control.php on
line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4
[Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice:
Undefined variable: sens in /var/www/html/web-meetme/meetme_control.php on
line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4
[Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice:
Undefined variable: current_page in
/var/www/html/web-meetme/meetme_control.php on line 278, referer:
http://10.1.1.30/web-meetme/meetme_control.php?s=4
[Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice:
Undefined variable: dateReq in /var/www/html/web-meetme/meetme_control.php
on line 573, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4


Also the Reports link doesnt display anything and in httpd error logs it
gives me following php errors:
[Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning:
include(locale.php) [a href='function.include'function.include/a]:
failed to open stream: No such file or directory in
/var/www/html/web-meetme/lib/defines.php on line 3, referer:
http://10.1.1.30/web-meetme/daily.php?
[Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning:
include() [a href='function.include'function.include/a]: Failed opening
'locale.php' for inclusion (include_path='.:/usr/share/pear:/usr/share/php')
in /var/www/html/web-meetme/lib/defines.php on line 3, referer:
http://10.1.1.30/web-meetme/daily.php?
[Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning:
include(locale.php) [a href='function.include'function.include/a]:
failed to open stream: No such file or directory in
/var/www/html/web-meetme/lib/email_body.php on line 3, referer:
http://10.1.1.30/web-meetme/daily.php?
[Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning:
include() [a href='function.include'function.include/a]: Failed opening
'locale.php' for inclusion (include_path='.:/usr/share/pear:/usr/share/php')
in /var/www/html/web-meetme/lib/email_body.php on line 3, referer:
http://10.1.1.30/web-meetme/daily.php?


Otherwise i am able to record and play the recorded file from the speaker
button.

--Manmohan Singh

On Fri, Jul 30, 2010 at 9:10 PM, Dan Austin dan_aus...@phoenix.com wrote:

 Manmohan wrote:
  There was on very silly mistake and i missed to check that properly.
 Really apologize for that.
  Following change was done to get the conf-recording into the proper path:

  chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings

  following is the output:

  [r...@linuxtest sounds]# ll
  total 6416
  drwxrwxr-x  2 asterisk asterisk4096 Jul 30 08:29 conf-recordings
  [r...@linuxtest sounds]# ll conf-recordings/
  total 4060
  -rw-r--r-- 1 asterisk asterisk 4150124 Jul 30 08:27
 meetme-conf-rec-74438-1280463795.8.wav

  The only thing now is no speaker icon onto the webpage when i click to
 past conference link.
 The web interface cannot find the recording.  The reason it cannot is that
 the name is wrong.  By wrong, I mean it contains information that the
 database
 and program is not aware of (1280463795.8).  To make this clear, if this
 conference was
 the 3rd one you ever scheduled on this system the correct file name would
 be-
 meetme-conf-rec-74438-3.wav using the format
 meetme-conf-rec-%PIN%-%BOOKID%.wav
 The database knows the pin and bookid, so it can construct the file name
 and test if it
 exists.



  Do you say that if i shift from 4.0.1 to 4.0.2 will fix the issue (of
 getting speaker icon in past conference)?
 I was not able to get the change into app_meetme to use the bookid in the
 filename,
 even though it has access to bookid.  I gave up and now store the filename
 in
 the database, which app_meetme will use if it exists.

 Other that a handful of bug-fixes, this is the major difference between
 4.0.1 and 4.0.2

 Dan

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Manmohan Singh Jandu
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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-08-04 Thread Dan Austin
Manmohan wrote:
 I had tried the new version of webmeetme i.e., 4.0.2
 The recording works very well.
Great!

 I see following php errors whenever i try to add in conference.

 [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice:  
 Undefined variable: order in /var/www/html/web-meetme/meetme_control.php 
 on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4
You can ignore the Notices.  They are fairly harmless, and only mean that
variable is not set by the code or being passed in on the URL.  You can
turn off notices in /etc/php.ini if they bother you.

 Also the Reports link doesnt display anything and in httpd error logs it 
 gives me following php errors:
 [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning:  
 include(locale.php) [a href='function.include'function.include/a]: 
 failed to open stream: No such file or directory in 
 /var/www/html/web-meetme/lib/defines.php
 on line 3, referer: http://10.1.1.30/web-meetme/daily.php?

In lib/defines.php, either comment out the 3rd line or add ../ before 
locale.php-
include(../locale.php);

But that is not likely why you do not get the reports.  The most likely cause is
A PHP notice is being thrown while the GD code is rendering the graph, 
resulting in
a corrupt image which your browser cannot display.

Check these settings /etc/php.ini-
error_reporting  =  E_ALL
display_errors = Off

Dan

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-30 Thread Dan Austin
Manmohan wrote:
 I did added the record option in user options as well. 

 $Mod_Options = array(array(_(Announce), I), array(_(Record), r));
 $User_Options = array(array(_(Announce), I), array(_(Listen Only), 
 m), array(_(Wait for Leader), w), 
 array(_(Record), r));

 And the gre8 news is, it did worked this time. But it saved the recorded file 
 in the following path:
That is good to hear.

 /var/lib/asterisk/sounds/    with the name as 
 meetme-conf-rec-74438-1280463795.8.wav

 Than i tried to move the file to /var/lib/asterisk/sounds/conf-recordings/ 
 just to see that it gives me a 
 speaker icon when i click to past conferences.

 Unfortunately i couldnt see this speaker icon to hear this recorded 
 conference .wav file.
I am not surprised.  By default MeetMe creates unique file names by appending
pin-uniqueid, but uniqueid is not know until the conference starts, so the web 
interface
does not know what to look for.  Part of the changes to app_meetme included 
setting the
realtime filename to use.

 I tried to download the .wav file into my windows machine and the filed 
 played well..

 like i mentioned in my earlier mail that following line i had added in 
 lib/define.php, please correct me if i am wrong:


 define (RECORDING_PATH, /var/lib/asterisk/sounds/conf-recordings/);

 Do you think This recording path is taking the effect here?

That setting effect where the WMM interface looks for recordings and not where 
Asterisk puts
them.  Looking back at your email history, I see you are on 4.0.1.  After all 
of the suggestions,
I remembered that I too found problems with recordings and addressed them in 
4.0.2

That version adds a field to the database and stores the recording names in the 
database.  I
recommend using that version instead of 4.0.1.  You can move your copy of 
lib/defines.php to 
the 4.0.2 install and keep your changes.

Dan


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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-30 Thread Manmohan Singh Jandu
Hi Dan,

There was on very silly mistake and i missed to check that properly. Really
apologize for that.
Following change was done to get the conf-recording into the proper path:

chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings

following is the output:

[r...@linuxtest sounds]# ll
total 6416
drwxrwxr-x  2 asterisk asterisk4096 Jul 30 08:29 conf-recordings
[r...@linuxtest sounds]# ll conf-recordings/
total 4060
-rw-r--r-- 1 asterisk asterisk 4150124 Jul 30 08:27
meetme-conf-rec-74438-1280463795.8.wav

The only thing now is no speaker icon onto the webpage when i click to past
conference link.

Do you say that if i shift from 4.0.1 to 4.0.2 will fix the issue (of
getting speaker icon in past conference)?

--Manmohan Singh

On Fri, Jul 30, 2010 at 8:16 PM, Dan Austin dan_aus...@phoenix.com wrote:

 Manmohan wrote:
  I did added the record option in user options as well.

  $Mod_Options = array(array(_(Announce), I), array(_(Record), r));
  $User_Options = array(array(_(Announce), I), array(_(Listen Only),
 m), array(_(Wait for Leader), w),
  array(_(Record), r));

  And the gre8 news is, it did worked this time. But it saved the recorded
 file in the following path:
 That is good to hear.

  /var/lib/asterisk/sounds/with the name as
 meetme-conf-rec-74438-1280463795.8.wav

  Than i tried to move the file to
 /var/lib/asterisk/sounds/conf-recordings/ just to see that it gives me a
  speaker icon when i click to past conferences.

  Unfortunately i couldnt see this speaker icon to hear this recorded
 conference .wav file.
 I am not surprised.  By default MeetMe creates unique file names by
 appending
 pin-uniqueid, but uniqueid is not know until the conference starts, so the
 web interface
 does not know what to look for.  Part of the changes to app_meetme included
 setting the
 realtime filename to use.

  I tried to download the .wav file into my windows machine and the filed
 played well..

  like i mentioned in my earlier mail that following line i had added in
 lib/define.php, please correct me if i am wrong:


  define (RECORDING_PATH, /var/lib/asterisk/sounds/conf-recordings/);

  Do you think This recording path is taking the effect here?

 That setting effect where the WMM interface looks for recordings and not
 where Asterisk puts
 them.  Looking back at your email history, I see you are on 4.0.1.  After
 all of the suggestions,
 I remembered that I too found problems with recordings and addressed them
 in 4.0.2

 That version adds a field to the database and stores the recording names in
 the database.  I
 recommend using that version instead of 4.0.1.  You can move your copy of
 lib/defines.php to
 the 4.0.2 install and keep your changes.

 Dan


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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-30 Thread Dan Austin
Manmohan wrote:
 There was on very silly mistake and i missed to check that properly. Really 
 apologize for that.
 Following change was done to get the conf-recording into the proper path:

 chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings

 following is the output:

 [r...@linuxtest sounds]# ll
 total 6416
 drwxrwxr-x  2 asterisk asterisk    4096 Jul 30 08:29 conf-recordings
 [r...@linuxtest sounds]# ll conf-recordings/
 total 4060
 -rw-r--r-- 1 asterisk asterisk 4150124 Jul 30 08:27 
 meetme-conf-rec-74438-1280463795.8.wav

 The only thing now is no speaker icon onto the webpage when i click to past 
 conference link.
The web interface cannot find the recording.  The reason it cannot is that
the name is wrong.  By wrong, I mean it contains information that the database
and program is not aware of (1280463795.8).  To make this clear, if this 
conference was
the 3rd one you ever scheduled on this system the correct file name would be-
meetme-conf-rec-74438-3.wav using the format meetme-conf-rec-%PIN%-%BOOKID%.wav
The database knows the pin and bookid, so it can construct the file name and 
test if it
exists.



 Do you say that if i shift from 4.0.1 to 4.0.2 will fix the issue (of getting 
 speaker icon in past conference)?
I was not able to get the change into app_meetme to use the bookid in the 
filename,
even though it has access to bookid.  I gave up and now store the filename in
the database, which app_meetme will use if it exists.

Other that a handful of bug-fixes, this is the major difference between 4.0.1 
and 4.0.2

Dan

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-29 Thread Dan Austin

Manmohan wrote:
 Following is the output for core set verbose 5, 
 also i am really not able to get on the admin pin
 thing? Do you mean, that with admin pin configured
 we cant use recording?

You are actually running a version that has been fixed
to support recording with pin-less or user pins.  I should
point out that the default settings in WMM is only to present
the recording checkbox with the admin pin field.

It is a fairly simple edit to add the recording checkbox to the
user pin (and these options apply if no pin is set).

Look in lib/defines.php for Mod_Options and User_Options to 
See how to add or remove MeetMe options from the GUI and database.

Dan

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-29 Thread Manmohan Singh Jandu
Hi Dan,

I did added the record option in user options as well.

$Mod_Options = array(array(_(Announce), I), array(_(Record), r));
$User_Options = array(array(_(Announce), I), array(_(Listen Only),
m), array(_(Wait for Leader), w), array(_(Record), r));

And the gre8 news is, it did worked this time. But it saved the recorded
file in the following path:

/var/lib/asterisk/sounds/with the name as
meetme-conf-rec-74438-1280463795.8.wav

Than i tried to move the file to /var/lib/asterisk/sounds/conf-recordings/
just to see that it gives me a speaker icon when i click to past
conferences.

Unfortunately i couldnt see this speaker icon to hear this recorded
conference .wav file.

I tried to download the .wav file into my windows machine and the filed
played well..

like i mentioned in my earlier mail that following line i had added in
lib/define.php, please correct me if i am wrong:


define (RECORDING_PATH, /var/lib/asterisk/sounds/conf-recordings/);

Do you think This recording path is taking the effect here?

--Manmohan Singh.


On Fri, Jul 30, 2010 at 2:56 AM, Dan Austin dan_aus...@phoenix.com wrote:


 Manmohan wrote:
  Following is the output for core set verbose 5,
  also i am really not able to get on the admin pin
  thing? Do you mean, that with admin pin configured
  we cant use recording?

 You are actually running a version that has been fixed
 to support recording with pin-less or user pins.  I should
 point out that the default settings in WMM is only to present
 the recording checkbox with the admin pin field.

 It is a fairly simple edit to add the recording checkbox to the
 user pin (and these options apply if no pin is set).

 Look in lib/defines.php for Mod_Options and User_Options to
 See how to add or remove MeetMe options from the GUI and database.

 Dan

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-28 Thread Dan Austin
Manmohan wrote:
 I can see the path does exists but i cant see any recordings
 happening inn there.  There are no files in it

 Following is the output:

 /var/lib/asterisk/sounds
 drwxrwxrwx  2 asterisk apache   4096 Jun 27 20:54 conf-recordings

 I hope m understandly this correctly but m sure m missing something here ;-)

You did understand, and we have eliminated another of the possible
issues.  Are you assigning an admin pin to these conferences?
There is a patch that allows recording pinless concenferences, but is
has oddly not been merged yet.  Try setting an admin pin.

If that does not work, send the CLI output with core set verbose 5 as
you dial in to the conference.

Dan

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-28 Thread Manmohan Singh Jandu
Hi Dan,

Following is the output for core set verbose 5, also i am really not able to
get on the admin pin thing? Do you mean, that with admin pin configured we
cant use recording?

LinuxTest*CLI core set verbose 5
Verbosity was 3 and is now 5
  == Using SIP RTP CoS mark 5
-- Executing [...@callman_incoming:1] MeetMe(SIP/callman02-0002,
) in new stack
-- SIP/callman02-0002 Playing 'conf-getconfno.ulaw' (language
'en')
  == Parsing '/etc/asterisk/meetme.conf':   == Found
-- Created MeetMe conference 1023 for conference '77972'
-- SIP/callman02-0002 Playing 'conf-getpin.ulaw' (language 'en')
Starting recording of MeetMe Conference 77972 into file ..
-- SIP/callman02-0002 Playing 'vm-rec-name.ulaw' (language 'en')
[Jul 29 09:16:19] WARNING[25049]: file.c:1160 ast_writefile: No such format
''
-- SIP/callman02-0002 Playing 'beep.ulaw' (language 'en')
-- x=0, open writing:
/var/spool/asterisk/meetme/meetme-username-77972-1 format: sln, 0x9bea628
-- User ended message by pressing #
-- SIP/callman02-0002 Playing 'auth-thankyou.ulaw' (language 'en')
-- SIP/callman02-0002 Playing 'conf-onlyperson.ulaw' (language
'en')
  == Using SIP RTP CoS mark 5
-- Executing [...@callman_incoming:1] MeetMe(SIP/callman02-0003,
) in new stack
-- SIP/callman02-0003 Playing 'conf-getconfno.ulaw' (language
'en')
-- SIP/callman02-0003 Playing 'conf-getpin.ulaw' (language 'en')
Starting recording of MeetMe Conference 77972 into file ..
-- SIP/callman02-0003 Playing 'vm-rec-name.ulaw' (language 'en')
-- SIP/callman02-0003 Playing 'beep.ulaw' (language 'en')
-- x=0, open writing:
/var/spool/asterisk/meetme/meetme-username-77972-2 format: sln, 0x9bea628
-- User ended message by pressing #
-- SIP/callman02-0003 Playing 'auth-thankyou.ulaw' (language 'en')
-- DAHDI/pseudo-736798397 Playing
'/var/spool/asterisk/meetme/meetme-username-77972-2.slin' (language 'en')
-- DAHDI/pseudo-736798397 Playing 'conf-hasjoin.ulaw' (language 'en')
-- SIP/callman02-0003 Playing 'conf-placeintoconf.ulaw' (language
'en')
  == Spawn extension (callman_incoming, 493, 1) exited non-zero on
'SIP/callman02-0002'
-- Executing [...@callman_incoming:1] Set(SIP/callman02-0002,
CDR(bookId)=) in new stack
-- Executing [...@callman_incoming:2] Set(SIP/callman02-0002,
CDR(CIDnum)=281) in new stack
-- Executing [...@callman_incoming:3] Set(SIP/callman02-0002,
CDR(CIDname)=Manmohan Singh Jandu) in new stack
-- DAHDI/pseudo-736798397 Playing
'/var/spool/asterisk/meetme/meetme-username-77972-1.slin' (language 'en')
-- SIP/callman02-0003 Playing 'conf-leaderhasleft.ulaw' (language
'en')
-- DAHDI/pseudo-736798397 Playing 'conf-hasleft.ulaw' (language 'en')
-- Hungup 'DAHDI/pseudo-923268627'
-- Hungup 'DAHDI/pseudo-736798397'
  == Spawn extension (callman_incoming, 493, 1) exited non-zero on
'SIP/callman02-0003'
-- Executing [...@callman_incoming:1] Set(SIP/callman02-0003,
CDR(bookId)=) in new stack
-- Executing [...@callman_incoming:2] Set(SIP/callman02-0003,
CDR(CIDnum)=115) in new stack
-- Executing [...@callman_incoming:3] Set(SIP/callman02-0003,
CDR(CIDname)=cipc) in new stack


On Thu, Jul 29, 2010 at 2:39 AM, Dan Austin dan_aus...@phoenix.com wrote:

 Manmohan wrote:
  I can see the path does exists but i cant see any recordings
  happening inn there.  There are no files in it

  Following is the output:

  /var/lib/asterisk/sounds
  drwxrwxrwx  2 asterisk apache   4096 Jun 27 20:54 conf-recordings

  I hope m understandly this correctly but m sure m missing something here
 ;-)

 You did understand, and we have eliminated another of the possible
 issues.  Are you assigning an admin pin to these conferences?
 There is a patch that allows recording pinless concenferences, but is
 has oddly not been merged yet.  Try setting an admin pin.

 If that does not work, send the CLI output with core set verbose 5 as
 you dial in to the conference.

 Dan

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-28 Thread Manmohan Singh Jandu
Also following is what i am putting in lib/define.php

define (RECORDING_PATH, /var/lib/asterisk/sounds/conf-recordings/);

On Thu, Jul 29, 2010 at 9:20 AM, Manmohan Singh Jandu
manmoha...@gmail.comwrote:

 Hi Dan,

 Following is the output for core set verbose 5, also i am really not able
 to get on the admin pin thing? Do you mean, that with admin pin configured
 we cant use recording?

 LinuxTest*CLI core set verbose 5
 Verbosity was 3 and is now 5

   == Using SIP RTP CoS mark 5
 -- Executing [...@callman_incoming:1] MeetMe(SIP/callman02-0002,
 ) in new stack
 -- SIP/callman02-0002 Playing 'conf-getconfno.ulaw' (language
 'en')

   == Parsing '/etc/asterisk/meetme.conf':   == Found
 -- Created MeetMe conference 1023 for conference '77972'
 -- SIP/callman02-0002 Playing 'conf-getpin.ulaw' (language 'en')
 Starting recording of MeetMe Conference 77972 into file ..
 -- SIP/callman02-0002 Playing 'vm-rec-name.ulaw' (language 'en')
 [Jul 29 09:16:19] WARNING[25049]: file.c:1160 ast_writefile: No such format
 ''
 -- SIP/callman02-0002 Playing 'beep.ulaw' (language 'en')
 -- x=0, open writing:
 /var/spool/asterisk/meetme/meetme-username-77972-1 format: sln, 0x9bea628
 -- User ended message by pressing #
 -- SIP/callman02-0002 Playing 'auth-thankyou.ulaw' (language
 'en')
 -- SIP/callman02-0002 Playing 'conf-onlyperson.ulaw' (language
 'en')

   == Using SIP RTP CoS mark 5
 -- Executing [...@callman_incoming:1] MeetMe(SIP/callman02-0003,
 ) in new stack
 -- SIP/callman02-0003 Playing 'conf-getconfno.ulaw' (language
 'en')
 -- SIP/callman02-0003 Playing 'conf-getpin.ulaw' (language 'en')
 Starting recording of MeetMe Conference 77972 into file ..
 -- SIP/callman02-0003 Playing 'vm-rec-name.ulaw' (language 'en')
 -- SIP/callman02-0003 Playing 'beep.ulaw' (language 'en')
 -- x=0, open writing:
 /var/spool/asterisk/meetme/meetme-username-77972-2 format: sln, 0x9bea628
 -- User ended message by pressing #
 -- SIP/callman02-0003 Playing 'auth-thankyou.ulaw' (language
 'en')
 -- DAHDI/pseudo-736798397 Playing
 '/var/spool/asterisk/meetme/meetme-username-77972-2.slin' (language 'en')
 -- DAHDI/pseudo-736798397 Playing 'conf-hasjoin.ulaw' (language 'en')
 -- SIP/callman02-0003 Playing 'conf-placeintoconf.ulaw' (language
 'en')
   == Spawn extension (callman_incoming, 493, 1) exited non-zero on
 'SIP/callman02-0002'
 -- Executing [...@callman_incoming:1] Set(SIP/callman02-0002,
 CDR(bookId)=) in new stack
 -- Executing [...@callman_incoming:2] Set(SIP/callman02-0002,
 CDR(CIDnum)=281) in new stack
 -- Executing [...@callman_incoming:3] Set(SIP/callman02-0002,
 CDR(CIDname)=Manmohan Singh Jandu) in new stack
 -- DAHDI/pseudo-736798397 Playing
 '/var/spool/asterisk/meetme/meetme-username-77972-1.slin' (language 'en')
 -- SIP/callman02-0003 Playing 'conf-leaderhasleft.ulaw' (language
 'en')
 -- DAHDI/pseudo-736798397 Playing 'conf-hasleft.ulaw' (language 'en')
 -- Hungup 'DAHDI/pseudo-923268627'
 -- Hungup 'DAHDI/pseudo-736798397'
   == Spawn extension (callman_incoming, 493, 1) exited non-zero on
 'SIP/callman02-0003'
 -- Executing [...@callman_incoming:1] Set(SIP/callman02-0003,
 CDR(bookId)=) in new stack
 -- Executing [...@callman_incoming:2] Set(SIP/callman02-0003,
 CDR(CIDnum)=115) in new stack
 -- Executing [...@callman_incoming:3] Set(SIP/callman02-0003,
 CDR(CIDname)=cipc) in new stack



 On Thu, Jul 29, 2010 at 2:39 AM, Dan Austin dan_aus...@phoenix.comwrote:

 Manmohan wrote:
  I can see the path does exists but i cant see any recordings
  happening inn there.  There are no files in it

  Following is the output:

  /var/lib/asterisk/sounds
  drwxrwxrwx  2 asterisk apache   4096 Jun 27 20:54 conf-recordings

  I hope m understandly this correctly but m sure m missing something here
 ;-)

 You did understand, and we have eliminated another of the possible
 issues.  Are you assigning an admin pin to these conferences?
 There is a patch that allows recording pinless concenferences, but is
 has oddly not been merged yet.  Try setting an admin pin.

 If that does not work, send the CLI output with core set verbose 5 as
 you dial in to the conference.

 Dan

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   http://lists.digium.com/mailman/listinfo/asterisk-users




 --
 Thanks  Regards
 Manmohan Singh Jandu




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Thanks  Regards
Manmohan Singh Jandu
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New to 

Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-27 Thread Manmohan Singh Jandu
Hi Dan,

I can see the path does exists but i cant see any recordings happening inn
there.
There are no files in it

Following is the output:

/var/lib/asterisk/sounds
drwxrwxrwx  2 asterisk apache   4096 Jun 27 20:54 conf-recordings

I hope m understandly this correctly but m sure m missing something here ;-)

--Manmohan Singh.



On Tue, Jul 27, 2010 at 3:03 AM, Dan Austin dan_aus...@phoenix.com wrote:



 Manmohan Singh Jandu wrote:
  OK, now i added the column members in the table booking manually.
  and disabled selinux to have this working.

  Now i am struggling with recording option in webmeetme.
  Not sure on how to enable it, though m checking the checkbox
  while creating the conference. But where does this save and how to
 retrieve it?

 The location of the recordings is set in lib/defines.php as RECORDING_PATH,
 which
 defaults to /var/lib/asterisk/sounds/conf-recordings/

 You can listen to the recordings after the conferences scheduled stop time
 by looking at the Past conferences page and clicking on the speaker icon
 next to the conference number.

 A couple of items to note-
1.  You may have to check the path to ensure it exists and that
the asterisk process can write to it.
2.  Your web service accounts needs read permissions for that path
3.  The speaker icon only displays if a recording exists.

 Dan

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Manmohan Singh Jandu
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Re: [asterisk-users] Asterisk crashes to start if compiled with pbx_lua on latest updated CentOS

2010-07-27 Thread Paul Belanger
On Tue, Jul 27, 2010 at 12:45 AM, Faisal Hanif fai...@vopium.com wrote:
 Did any one got it solved? If yes how?

Yes, read doc/backtrace.txt.  It will explain how to generate an
unoptimized backtrace, then uploaded it to the mailing list.


-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-26 Thread Dan Austin
Manmohan Singh Jandu wrote:

 Excellent!
 I finally got it working, it was ODBC drivers issue 
 actually. Installed the proper compatible version and its working.
I thought that might be the case.

 There are still few errors which i see on asterisk console:
 [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:1032 require_odbc: 
 Realtime table book...@meetme requires column  'members', but that column 
 does not exist!
WMM does not use that column.  You can disable it by
Setting logmembercount=no in meetme.conf

 Also when i try to click the conference to manage it realtime it gives me 
 Error connection to the manager!

 Following are the database files which i used:

 /web-meetme/cbmysql/db-admin-user-create.txt
 /web-meetme/cbmysql/db-table-create-v6.txt
 /web-meetme/cbmysql/db-tables-v6.txt

 Am i missing something here now?
The WMM web interface used the Asterisk manager
interface to monitor and manage conferences.
The readme file documents the required changes to
manager.conf.

Sorry for the delay responding, I was on vacation
last week with no email access.

Dan




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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-26 Thread Dan Austin


Manmohan Singh Jandu wrote:
 OK, now i added the column members in the table booking manually.
 and disabled selinux to have this working.

 Now i am struggling with recording option in webmeetme.
 Not sure on how to enable it, though m checking the checkbox
 while creating the conference. But where does this save and how to retrieve 
 it?

The location of the recordings is set in lib/defines.php as RECORDING_PATH, 
which
defaults to /var/lib/asterisk/sounds/conf-recordings/

You can listen to the recordings after the conferences scheduled stop time
by looking at the Past conferences page and clicking on the speaker icon
next to the conference number.

A couple of items to note-
1.  You may have to check the path to ensure it exists and that
the asterisk process can write to it.
2.  Your web service accounts needs read permissions for that path
3.  The speaker icon only displays if a recording exists.

Dan

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[asterisk-users] Asterisk crashes to start if compiled with pbx_lua on latest updated CentOS

2010-07-26 Thread Faisal Hanif

 Hi,

I have tried number of time if we update any CentOS system (or use 
latest CentOS version) then compile asterisk 1.6.2 with pbx_lua support, 
asterisk will crash on starting and will give a core dump.


Issue is easy to produce,

Install latest CentOS on a system.
Install LUA  LUA Headers using YUM.
Download and Compile latest release of asterisk 1.6.2.
Try to start start asterisk in console mode.
It will crash on LUA and will give a core dump

Did any one got it solved? If yes how?

Regards,

Faisal Hanif

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-19 Thread Manmohan Singh Jandu
Excellent!
I finally got it working, it was ODBC drivers issue actually. Installed the
proper compatible version and its working.

There are still few errors which i see on asterisk console:
[Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:1032 require_odbc:
Realtime table book...@meetme requires column 'members', but that column
does not exist!
[Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:440 update_odbc: Key
field 'members' does not exist in table 'book...@meetme'.  Update will fail
[Jul 19 13:58:51] WARNING[30213]: res_odbc.c:616
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22:
[MySQL][ODBC 3.51 Driver][mysqld-5.0.77]Unknown column 'members' in 'field
list' (80)
[Jul 19 13:58:51] WARNING[30213]: res_odbc.c:628
ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a
reconnect...
[Jul 19 13:58:51] WARNING[30213]: res_odbc.c:723 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
[Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1427 odbc_obj_connect:
Connecting meetme
[Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1455 odbc_obj_connect: res_odbc:
Connected to meetme [meetme]
[Jul 19 13:58:51] WARNING[30213]: res_odbc.c:616
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22:
[MySQL][ODBC 3.51 Driver][mysqld-5.0.77]Unknown column 'members' in 'field
list' (80)
[Jul 19 13:58:51] WARNING[30213]: res_odbc.c:628
ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a
reconnect...
[Jul 19 13:58:51] WARNING[30213]: res_odbc.c:723 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
[Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1427 odbc_obj_connect:
Connecting meetme
[Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1455 odbc_obj_connect: res_odbc:
Connected to meetme [meetme]
-- SIP/callman02-0005 Playing 'conf-onlyperson.ulaw' (language
'en')


Also when i try to click the conference to manage it realtime it gives me
Error connection to the manager!

Following are the database files which i used:

/web-meetme/cbmysql/db-admin-user-create.txt
/web-meetme/cbmysql/db-table-create-v6.txt
/web-meetme/cbmysql/db-tables-v6.txt

Am i missing something here now?



On Tue, Jul 13, 2010 at 8:43 PM, cov...@ccs.covici.com wrote:

 cov...@ccs.covici.com wrote:

  Dan Austin dan_aus...@phoenix.com wrote:
 
   Manmohan wrote:
  
My Web-MeetMe_v4.0.1, i followed the instructions in the
README File in the same package.
   Good.  There are other instruction packages, but since I wrote
   the README it is the one I am most familiar with.
  
Are you using RealTime enabled app_meetme or app_cbmysql
from the WMM package? 
i didnt get this actually what do i need to check here? Please
dont mind but m not so good in opensource world. I try to read and
understand and on trial n error basis try  to implement things.
Though had very much interest in learning things.
   Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk
   was in a separate Asterisk application (app_cbmysql).  With version 4
 of
   WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe
   application.
  
   The README in 4.0.1 lists the steps to setup RealTime (database)
 support
   for Asterisk and MeetMe.  This narrows down the possible problems,
 since
   we do not need to consider app_cbmysql.
  
   Did you install Asterisk from a package with yum, or did you compile it
   yourself?
  
   Dan
 
  I am getting this error without webmeetme at all, after upgrading to
  svn-275706 from an earlier version 262801.  Its a certain argument of
  meetme which I have not trafcked down yet which is causing this.

 OK, if the argument to meetme is conference number,TcMsrm it does not
 crash, but if it is conference number, cMs then it dies -- asterisk
 dies.  Is this enough for someone to figure out?

 --
 Your life is like a penny.  You're going to lose it.  The question is:
 How do
 you spend it?

 John Covici
 cov...@ccs.covici.com

 --
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 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 asterisk-users mailing list
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   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
Thanks  Regards
Manmohan Singh Jandu
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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-19 Thread Manmohan Singh Jandu
OK, now i added the column members in the table booking manually.

and disabled selinux to have this working.

Now i am struggling with recording option in webmeetme.
Not sure on how to enable it, though m checking the checkbox while creating
the conference. But where does this save and how to retrieve it?

On Mon, Jul 19, 2010 at 9:57 AM, Manmohan Singh Jandu
manmoha...@gmail.comwrote:

 Excellent!
 I finally got it working, it was ODBC drivers issue actually. Installed the
 proper compatible version and its working.

 There are still few errors which i see on asterisk console:
 [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:1032 require_odbc:
 Realtime table book...@meetme requires column 'members', but that column
 does not exist!
 [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:440 update_odbc: Key
 field 'members' does not exist in table 'book...@meetme'.  Update will
 fail
 [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:616
 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22:
 [MySQL][ODBC 3.51 Driver][mysqld-5.0.77]Unknown column 'members' in 'field
 list' (80)
 [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:628
 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a
 reconnect...
 [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:723 ast_odbc_sanity_check:
 Connection is down attempting to reconnect...
 [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1427 odbc_obj_connect:
 Connecting meetme
 [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1455 odbc_obj_connect:
 res_odbc: Connected to meetme [meetme]
 [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:616
 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22:
 [MySQL][ODBC 3.51 Driver][mysqld-5.0.77]Unknown column 'members' in 'field
 list' (80)
 [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:628
 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a
 reconnect...
 [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:723 ast_odbc_sanity_check:
 Connection is down attempting to reconnect...
 [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1427 odbc_obj_connect:
 Connecting meetme
 [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1455 odbc_obj_connect:
 res_odbc: Connected to meetme [meetme]
 -- SIP/callman02-0005 Playing 'conf-onlyperson.ulaw' (language
 'en')


 Also when i try to click the conference to manage it realtime it gives me
 Error connection to the manager!

 Following are the database files which i used:

 /web-meetme/cbmysql/db-admin-user-create.txt
 /web-meetme/cbmysql/db-table-create-v6.txt
 /web-meetme/cbmysql/db-tables-v6.txt

 Am i missing something here now?




 On Tue, Jul 13, 2010 at 8:43 PM, cov...@ccs.covici.com wrote:

 cov...@ccs.covici.com wrote:

  Dan Austin dan_aus...@phoenix.com wrote:
 
   Manmohan wrote:
  
My Web-MeetMe_v4.0.1, i followed the instructions in the
README File in the same package.
   Good.  There are other instruction packages, but since I wrote
   the README it is the one I am most familiar with.
  
Are you using RealTime enabled app_meetme or app_cbmysql
from the WMM package? 
i didnt get this actually what do i need to check here? Please
dont mind but m not so good in opensource world. I try to read and
understand and on trial n error basis try  to implement things.
Though had very much interest in learning things.
   Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk
   was in a separate Asterisk application (app_cbmysql).  With version 4
 of
   WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe
   application.
  
   The README in 4.0.1 lists the steps to setup RealTime (database)
 support
   for Asterisk and MeetMe.  This narrows down the possible problems,
 since
   we do not need to consider app_cbmysql.
  
   Did you install Asterisk from a package with yum, or did you compile
 it
   yourself?
  
   Dan
 
  I am getting this error without webmeetme at all, after upgrading to
  svn-275706 from an earlier version 262801.  Its a certain argument of
  meetme which I have not trafcked down yet which is causing this.

 OK, if the argument to meetme is conference number,TcMsrm it does not
 crash, but if it is conference number, cMs then it dies -- asterisk
 dies.  Is this enough for someone to figure out?

 --
 Your life is like a penny.  You're going to lose it.  The question is:
 How do
 you spend it?

 John Covici
 cov...@ccs.covici.com

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




 --
 Thanks  Regards
 Manmohan Singh Jandu




-- 
Thanks  Regards
Manmohan Singh Jandu
-- 

Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-13 Thread covici
Dan Austin dan_aus...@phoenix.com wrote:

 Manmohan wrote:
 
  My Web-MeetMe_v4.0.1, i followed the instructions in the 
  README File in the same package.
 Good.  There are other instruction packages, but since I wrote
 the README it is the one I am most familiar with.
 
  Are you using RealTime enabled app_meetme or app_cbmysql 
  from the WMM package?  
  i didnt get this actually what do i need to check here? Please 
  dont mind but m not so good in opensource world. I try to read and
  understand and on trial n error basis try  to implement things. 
  Though had very much interest in learning things.
 Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk
 was in a separate Asterisk application (app_cbmysql).  With version 4 of
 WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe
 application.
 
 The README in 4.0.1 lists the steps to setup RealTime (database) support
 for Asterisk and MeetMe.  This narrows down the possible problems, since
 we do not need to consider app_cbmysql.
 
 Did you install Asterisk from a package with yum, or did you compile it
 yourself?  
 
 Dan

I am getting this error without webmeetme at all, after upgrading to
svn-275706 from an earlier version 262801.  Its a certain argument of
meetme which I have not trafcked down yet which is causing this.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-13 Thread covici
cov...@ccs.covici.com wrote:

 Dan Austin dan_aus...@phoenix.com wrote:
 
  Manmohan wrote:
  
   My Web-MeetMe_v4.0.1, i followed the instructions in the 
   README File in the same package.
  Good.  There are other instruction packages, but since I wrote
  the README it is the one I am most familiar with.
  
   Are you using RealTime enabled app_meetme or app_cbmysql 
   from the WMM package?  
   i didnt get this actually what do i need to check here? Please 
   dont mind but m not so good in opensource world. I try to read and
   understand and on trial n error basis try  to implement things. 
   Though had very much interest in learning things.
  Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk
  was in a separate Asterisk application (app_cbmysql).  With version 4 of
  WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe
  application.
  
  The README in 4.0.1 lists the steps to setup RealTime (database) support
  for Asterisk and MeetMe.  This narrows down the possible problems, since
  we do not need to consider app_cbmysql.
  
  Did you install Asterisk from a package with yum, or did you compile it
  yourself?  
  
  Dan
 
 I am getting this error without webmeetme at all, after upgrading to
 svn-275706 from an earlier version 262801.  Its a certain argument of
 meetme which I have not trafcked down yet which is causing this.

OK, if the argument to meetme is conference number,TcMsrm it does not
crash, but if it is conference number, cMs then it dies -- asterisk
dies.  Is this enough for someone to figure out?

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-12 Thread Dan Austin
Manmohan wrote:
 Unfortunately m not able to get rid of the below mentioned errors. not sure 
 on where i am missing now.
 On Sat, Jul 10, 2010 at 9:41 AM, Manmohan Singh Jandu manmoha...@gmail.com 
 wrote:
 Ahh here is the catch i was still using app_cbmysql for this.
 now i had removed and just followed the README of 4.0 for WMM  
 and m getting following on ,my asterisk console.

  Verbosity is at least 3
   == Using SIP RTP CoS mark 5
     -- Executing [...@phones:1] MeetMe(SIP/492-, ) in new stack
     -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en')
   == Parsing '/etc/asterisk/meetme.conf':   == Found
 [Jul 10 13:42:15] NOTICE[16906]: res_odbc.c:1427 odbc_obj_connect: Connecting 
 meetme
 [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1452 odbc_obj_connect: res_odbc: 
 Error SQLConnect=-1 errno=0 
 [unixODBC][Driver Manager]Data source name not found, and no default driver 
 specified
 [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1273 ast_odbc_request_obj2: 
 Failed to connect to meetme
 [Jul 10 13:42:15] ERROR[16906]: res_config_odbc.c:144 realtime_odbc: No 
 database handle available with the name of 
 'meetme' (check res_odbc.conf)
     -- SIP/492- Playing 'conf-invalid.ulaw' (language 'en')
     -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en')
     -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en')
  == Spawn extension (phones, 493, 1) exited non-zero on 'SIP/492-'


 (Initially i installed using yum, i was getting the same issue.
 Than i scrapped everything and installed it manually.)

The good news is that you are making progress.  Do you have the package 
unixODBC installed?
The hint to that would have been if you created a new /etc/odbc.ini instead of 
editing 
a sample that the package would have installed.

Dan

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-11 Thread Manmohan Singh Jandu
Unfortunately m not able to get rid of the below mentioned errors. not sure
on where i am missing now.

On Sat, Jul 10, 2010 at 9:41 AM, Manmohan Singh Jandu
manmoha...@gmail.comwrote:

 Ahh here is the catch i was still using app_cbmysql for this.
 now i had removed and just followed the README of 4.0 for WMM
 and m getting following on ,my asterisk console.

 Verbosity is at least 3
   == Using SIP RTP CoS mark 5
 -- Executing [...@phones:1] MeetMe(SIP/492-, ) in new
 stack
 -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en')
   == Parsing '/etc/asterisk/meetme.conf':   == Found
 [Jul 10 13:42:15] NOTICE[16906]: res_odbc.c:1427 odbc_obj_connect:
 Connecting meetme
 [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1452 odbc_obj_connect:
 res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source
 name not found, and no default driver specified
 [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1273 ast_odbc_request_obj2:
 Failed to connect to meetme
 [Jul 10 13:42:15] ERROR[16906]: res_config_odbc.c:144 realtime_odbc: No
 database handle available with the name of 'meetme' (check res_odbc.conf)
 -- SIP/492- Playing 'conf-invalid.ulaw' (language 'en')
 -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en')
 -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en')
   == Spawn extension (phones, 493, 1) exited non-zero on 'SIP/492-'


 (Initially i installed using yum, i was getting the same issue.
 Than i scrapped everything and installed it manually.)




 On Fri, Jul 9, 2010 at 8:39 PM, Dan Austin dan_aus...@phoenix.com wrote:

 Manmohan wrote:

  My Web-MeetMe_v4.0.1, i followed the instructions in the
  README File in the same package.
 Good.  There are other instruction packages, but since I wrote
 the README it is the one I am most familiar with.

  Are you using RealTime enabled app_meetme or app_cbmysql
  from the WMM package? 
  i didnt get this actually what do i need to check here? Please
  dont mind but m not so good in opensource world. I try to read and
  understand and on trial n error basis try  to implement things.
  Though had very much interest in learning things.
 Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk
 was in a separate Asterisk application (app_cbmysql).  With version 4 of
 WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe
 application.

 The README in 4.0.1 lists the steps to setup RealTime (database) support
 for Asterisk and MeetMe.  This narrows down the possible problems, since
 we do not need to consider app_cbmysql.

 Did you install Asterisk from a package with yum, or did you compile it
 yourself?

 Dan


 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




 --
 Thanks  Regards
 Manmohan Singh Jandu




-- 
Thanks  Regards
Manmohan Singh Jandu
-- 
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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-09 Thread Manmohan Singh Jandu
Hi,

My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the
same package.

Are you using RealTime enabled app_meetme or app_cbmysql from the WMM
package?  i didnt get this actually what do i need to check here?
Please dont mind but m not so good in opensource world. I try to read and
understand and on trial n error basis try  to implement things. Though had
very much interest in learning things.

The GDB output is huge on, Following are my GDB errors.

[r...@linuxtest tmp]# gdb asterisk core.LinuxTest-2010-07-07T21:13:15+0400 |
more
GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1)
Copyright (C) 2009 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html

This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type show copying
and show warranty for details.
This GDB was configured as i386-redhat-linux-gnu.
For bug reporting instructions, please see:
http://www.gnu.org/software/gdb/bugs/...
Reading symbols from /usr/sbin/asterisk...done.

warning: .dynamic section for /usr/lib/libidn.so.11 is not at the expected
address

warning: difference appears to be caused by prelink, adjusting expectations
[New Thread 3212]


SOME OF THE LINES IN the end of GDB Error:

Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done.
Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so
Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no
debugging symbols found)...done.
Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so
Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done.
Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so
Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
Program terminated with signal 11, Segmentation fault.
#0  0x01027d9d in mysql_fetch_row () from
/usr/lib/mysql/libmysqlclient.so.15


--Manmohan Singh.

On Thu, Jul 8, 2010 at 11:21 PM, Dan Austin dan_aus...@phoenix.com wrote:

 Manmohan wrote:
  I was looking for audio conferencing solution where i got Web-meetme.
  I had installed Asterisk 1.6.2.9 on Centos  5.4. Its perfecting working
  fine. I tried using Meetme even meetme app is working perfectly fine.
  I installed Webmeetme 4.0 and integrated with my asterisk. When i try
  to dial the conference number it take me to an IVR wherein it asks for
  the conference number. The time i provide the conference number, asterisk
  crashes giving segmentation fault.
  I have been trying to google up and checked lot of forums but didnt get
  any solution for this yet.

 Which instructions did you follow for the integration?  Are you using
 RealTime enabled app_meetme or app_cbmysql from the WMM package?  Which
 exact version of WMM?

 Dan

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
Thanks  Regards
Manmohan Singh Jandu
-- 
_
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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-09 Thread Chandrakant Solanki
Hi

Install mysql 'n mysql-devel which includes
/usr/lib/mysql/libmysqlclient.so.15 library.

And also insert /usr/lib/mysql into /etc/ld.so.conf and then execute
ldconfig command on terminal.


-- 
Regards,

Chandrakant Solanki

On Fri, Jul 9, 2010 at 2:32 PM, Manmohan Singh Jandu
manmoha...@gmail.comwrote:

 Hi,

 My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the
 same package.

 Are you using RealTime enabled app_meetme or app_cbmysql from the WMM
 package?  i didnt get this actually what do i need to check here?
 Please dont mind but m not so good in opensource world. I try to read and
 understand and on trial n error basis try  to implement things. Though had
 very much interest in learning things.

 The GDB output is huge on, Following are my GDB errors.

 [r...@linuxtest tmp]# gdb asterisk core.LinuxTest-2010-07-07T21:13:15+0400
 | more
 GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1)
 Copyright (C) 2009 Free Software Foundation, Inc.
 License GPLv3+: GNU GPL version 3 or later 
 http://gnu.org/licenses/gpl.html
 This is free software: you are free to change and redistribute it.
 There is NO WARRANTY, to the extent permitted by law.  Type show copying
 and show warranty for details.
 This GDB was configured as i386-redhat-linux-gnu.
 For bug reporting instructions, please see:
 http://www.gnu.org/software/gdb/bugs/...
 Reading symbols from /usr/sbin/asterisk...done.

 warning: .dynamic section for /usr/lib/libidn.so.11 is not at the
 expected address

 warning: difference appears to be caused by prelink, adjusting expectations
 [New Thread 3212]


 SOME OF THE LINES IN the end of GDB Error:

 Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done.
 Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so
 Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no
 debugging symbols found)...done.
 Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so
 Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done.
 Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so
 Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
 Program terminated with signal 11, Segmentation fault.
 #0  0x01027d9d in mysql_fetch_row () from
 /usr/lib/mysql/libmysqlclient.so.15


 --Manmohan Singh.

 On Thu, Jul 8, 2010 at 11:21 PM, Dan Austin dan_aus...@phoenix.comwrote:

 Manmohan wrote:
  I was looking for audio conferencing solution where i got Web-meetme.
  I had installed Asterisk 1.6.2.9 on Centos  5.4. Its perfecting working
  fine. I tried using Meetme even meetme app is working perfectly fine.
  I installed Webmeetme 4.0 and integrated with my asterisk. When i try
  to dial the conference number it take me to an IVR wherein it asks for
  the conference number. The time i provide the conference number,
 asterisk
  crashes giving segmentation fault.
  I have been trying to google up and checked lot of forums but didnt get
  any solution for this yet.

 Which instructions did you follow for the integration?  Are you using
 RealTime enabled app_meetme or app_cbmysql from the WMM package?  Which
 exact version of WMM?

 Dan

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




 --
 Thanks  Regards
 Manmohan Singh Jandu

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-09 Thread Manmohan Singh Jandu
It still crashes and in gdb trace following is what its showing:

--More--
warning: .dynamic section for /usr/lib/mysql/libmysqlclient.so.15 is not
at the expected address

warning: difference appears to be caused by prelink, adjusting expectations
[New Thread 13310]


LAST FEW LINES IN GDB TRACE:

Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done.
Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so
Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no
debugging symbols found)...done.
Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so
Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done.
Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so
Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
Program terminated with signal 11, Segmentation fault.
#0  0x003acd9d in mysql_fetch_row () from
/usr/lib/mysql/libmysqlclient.so.15



--Manmohan Singh



On Fri, Jul 9, 2010 at 2:36 PM, Manmohan Singh Jandu
manmoha...@gmail.comwrote:

 Hi,

 Following is what i did.
 [r...@linuxtest ~]# yum install mysql*
 Loaded plugins: fastestmirror, kmod
 Loading mirror speeds from cached hostfile
  * addons: centos.skknet.net
  * base: centos.skknet.net
  * extras: centos.skknet.net
  * updates: centos.skknet.net
 Setting up Install Process
 Package mysql-devel-5.0.77-4.el5_5.3.i386 already installed and latest
 version
 Package mysql-server-5.0.77-4.el5_5.3.i386 already installed and latest
 version
 Package mysql-5.0.77-4.el5_5.3.i386 already installed and latest version
 Package mysql-connector-odbc-3.51.26r1127-1.el5.i386 already installed and
 latest version
 Resolving Dependencies
 -- Running transaction check
 --- Package mysql-bench.i386 0:5.0.77-4.el5_5.3 set to be updated
 --- Package mysql-test.i386 0:5.0.77-4.el5_5.3 set to be updated
 -- Finished Dependency Resolution

 Dependencies Resolved


 
  PackageArchVersionRepository
 Size

 
 Installing:
  mysql-benchi3865.0.77-4.el5_5.3   updates
 507 k
  mysql-test i3865.0.77-4.el5_5.3   updates
 3.7 M

 Transaction Summary

 
 Install   2 Package(s)
 Upgrade   0 Package(s)

 Total download size: 4.2 M
 Is this ok [y/N]: y
 Downloading Packages:
 (1/2): mysql-bench-5.0.77-4.el5_5.3.i386.rpm | 507 kB 00:02
 (2/2): mysql-test-5.0.77-4.el5_5.3.i386.rpm  | 3.7 MB 00:11

 
 Total   295 kB/s | 4.2 MB 00:14
 Running rpm_check_debug
 Running Transaction Test
 Finished Transaction Test
 Transaction Test Succeeded
 Running Transaction
   Installing : mysql-bench
 1/2
   Installing : mysql-test
 2/2

 Installed:
   mysql-bench.i386 0:5.0.77-4.el5_5.3 mysql-test.i386
 0:5.0.77-4.el5_5.3

 Complete!
 [r...@linuxtest ~]# yum install mysql-devel*
 Loaded plugins: fastestmirror, kmod
 Loading mirror speeds from cached hostfile
  * addons: centos.skknet.net
  * base: centos.skknet.net
  * extras: centosr4.centos.org
  * updates: centosg4.centos.org
 Setting up Install Process
 Package mysql-devel-5.0.77-4.el5_5.3.i386 already installed and latest
 version
 Nothing to do
 [r...@linuxtest ~]# cat /etc/ld.so.conf
 include ld.so.conf.d/*.conf
 [r...@linuxtest ~]# vi /etc/ld.so.conf
 [r...@linuxtest ~]# ldconfig
 [r...@linuxtest ~]# cat /etc/ld.so.conf
 include ld.so.conf.d/*.conf
 /usr/lib/mysql
 [r...@linuxtest ~]# ldconfig
 [r...@linuxtest ~]#

 Thanks  Regards
 Manmohan Singh




 On Fri, Jul 9, 2010 at 2:19 PM, Chandrakant Solanki 
 solanki.chandrak...@gmail.com wrote:

 Hi

 Install mysql 'n mysql-devel which includes
 /usr/lib/mysql/libmysqlclient.so.15 library.

 And also insert /usr/lib/mysql into /etc/ld.so.conf and then execute
 ldconfig command on terminal.


 --
 Regards,

 Chandrakant Solanki


 On Fri, Jul 9, 2010 at 2:32 PM, Manmohan Singh Jandu 
 manmoha...@gmail.com wrote:

 Hi,

 My Web-MeetMe_v4.0.1, i followed the instructions in the README File in
 the same package.

 Are you using RealTime enabled app_meetme or app_cbmysql from the WMM
 package?  i didnt get this actually what do i need to check here?
 Please dont mind but m not so good in opensource world. I try to read and
 understand and on trial n error basis try  to implement things. Though had
 very much interest in learning things.

 The GDB output is huge on, Following are my GDB errors.

 [r...@linuxtest tmp]# gdb asterisk
 core.LinuxTest-2010-07-07T21:13:15+0400 | more
 GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1)
 Copyright (C) 2009 Free Software Foundation, Inc.
 License GPLv3+: GNU GPL version 3 or later 
 

Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-09 Thread Manmohan Singh Jandu
Hi,

Following is what i did.
[r...@linuxtest ~]# yum install mysql*
Loaded plugins: fastestmirror, kmod
Loading mirror speeds from cached hostfile
 * addons: centos.skknet.net
 * base: centos.skknet.net
 * extras: centos.skknet.net
 * updates: centos.skknet.net
Setting up Install Process
Package mysql-devel-5.0.77-4.el5_5.3.i386 already installed and latest
version
Package mysql-server-5.0.77-4.el5_5.3.i386 already installed and latest
version
Package mysql-5.0.77-4.el5_5.3.i386 already installed and latest version
Package mysql-connector-odbc-3.51.26r1127-1.el5.i386 already installed and
latest version
Resolving Dependencies
-- Running transaction check
--- Package mysql-bench.i386 0:5.0.77-4.el5_5.3 set to be updated
--- Package mysql-test.i386 0:5.0.77-4.el5_5.3 set to be updated
-- Finished Dependency Resolution

Dependencies Resolved


 PackageArchVersionRepository
Size

Installing:
 mysql-benchi3865.0.77-4.el5_5.3   updates
507 k
 mysql-test i3865.0.77-4.el5_5.3   updates
3.7 M

Transaction Summary

Install   2 Package(s)
Upgrade   0 Package(s)

Total download size: 4.2 M
Is this ok [y/N]: y
Downloading Packages:
(1/2): mysql-bench-5.0.77-4.el5_5.3.i386.rpm | 507 kB 00:02
(2/2): mysql-test-5.0.77-4.el5_5.3.i386.rpm  | 3.7 MB 00:11

Total   295 kB/s | 4.2 MB 00:14
Running rpm_check_debug
Running Transaction Test
Finished Transaction Test
Transaction Test Succeeded
Running Transaction
  Installing : mysql-bench
1/2
  Installing : mysql-test
2/2

Installed:
  mysql-bench.i386 0:5.0.77-4.el5_5.3 mysql-test.i386 0:5.0.77-4.el5_5.3

Complete!
[r...@linuxtest ~]# yum install mysql-devel*
Loaded plugins: fastestmirror, kmod
Loading mirror speeds from cached hostfile
 * addons: centos.skknet.net
 * base: centos.skknet.net
 * extras: centosr4.centos.org
 * updates: centosg4.centos.org
Setting up Install Process
Package mysql-devel-5.0.77-4.el5_5.3.i386 already installed and latest
version
Nothing to do
[r...@linuxtest ~]# cat /etc/ld.so.conf
include ld.so.conf.d/*.conf
[r...@linuxtest ~]# vi /etc/ld.so.conf
[r...@linuxtest ~]# ldconfig
[r...@linuxtest ~]# cat /etc/ld.so.conf
include ld.so.conf.d/*.conf
/usr/lib/mysql
[r...@linuxtest ~]# ldconfig
[r...@linuxtest ~]#

Thanks  Regards
Manmohan Singh



On Fri, Jul 9, 2010 at 2:19 PM, Chandrakant Solanki 
solanki.chandrak...@gmail.com wrote:

 Hi

 Install mysql 'n mysql-devel which includes
 /usr/lib/mysql/libmysqlclient.so.15 library.

 And also insert /usr/lib/mysql into /etc/ld.so.conf and then execute
 ldconfig command on terminal.


 --
 Regards,

 Chandrakant Solanki


 On Fri, Jul 9, 2010 at 2:32 PM, Manmohan Singh Jandu manmoha...@gmail.com
  wrote:

 Hi,

 My Web-MeetMe_v4.0.1, i followed the instructions in the README File in
 the same package.

 Are you using RealTime enabled app_meetme or app_cbmysql from the WMM
 package?  i didnt get this actually what do i need to check here?
 Please dont mind but m not so good in opensource world. I try to read and
 understand and on trial n error basis try  to implement things. Though had
 very much interest in learning things.

 The GDB output is huge on, Following are my GDB errors.

 [r...@linuxtest tmp]# gdb asterisk
 core.LinuxTest-2010-07-07T21:13:15+0400 | more
 GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1)
 Copyright (C) 2009 Free Software Foundation, Inc.
 License GPLv3+: GNU GPL version 3 or later 
 http://gnu.org/licenses/gpl.html
 This is free software: you are free to change and redistribute it.
 There is NO WARRANTY, to the extent permitted by law.  Type show copying
 and show warranty for details.
 This GDB was configured as i386-redhat-linux-gnu.
 For bug reporting instructions, please see:
 http://www.gnu.org/software/gdb/bugs/...
 Reading symbols from /usr/sbin/asterisk...done.

 warning: .dynamic section for /usr/lib/libidn.so.11 is not at the
 expected address

 warning: difference appears to be caused by prelink, adjusting
 expectations
 [New Thread 3212]


 SOME OF THE LINES IN the end of GDB Error:

 Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done.
 Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so
 Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no
 debugging symbols found)...done.
 Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so
 Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done.
 Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so
 Core was generated by `/usr/sbin/asterisk -f -vvvg 

Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-09 Thread Dan Austin
Manmohan wrote:

 My Web-MeetMe_v4.0.1, i followed the instructions in the 
 README File in the same package.
Good.  There are other instruction packages, but since I wrote
the README it is the one I am most familiar with.

 Are you using RealTime enabled app_meetme or app_cbmysql 
 from the WMM package?  
 i didnt get this actually what do i need to check here? Please 
 dont mind but m not so good in opensource world. I try to read and
 understand and on trial n error basis try  to implement things. 
 Though had very much interest in learning things.
Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk
was in a separate Asterisk application (app_cbmysql).  With version 4 of
WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe
application.

The README in 4.0.1 lists the steps to setup RealTime (database) support
for Asterisk and MeetMe.  This narrows down the possible problems, since
we do not need to consider app_cbmysql.

Did you install Asterisk from a package with yum, or did you compile it
yourself?  

Dan


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-09 Thread Manmohan Singh Jandu
Ahh here is the catch i was still using app_cbmysql for this.
now i had removed and just followed the README of 4.0 for WMM
and m getting following on ,my asterisk console.

Verbosity is at least 3
  == Using SIP RTP CoS mark 5
-- Executing [...@phones:1] MeetMe(SIP/492-, ) in new stack
-- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en')
  == Parsing '/etc/asterisk/meetme.conf':   == Found
[Jul 10 13:42:15] NOTICE[16906]: res_odbc.c:1427 odbc_obj_connect:
Connecting meetme
[Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1452 odbc_obj_connect:
res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source
name not found, and no default driver specified
[Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1273 ast_odbc_request_obj2:
Failed to connect to meetme
[Jul 10 13:42:15] ERROR[16906]: res_config_odbc.c:144 realtime_odbc: No
database handle available with the name of 'meetme' (check res_odbc.conf)
-- SIP/492- Playing 'conf-invalid.ulaw' (language 'en')
-- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en')
-- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en')
  == Spawn extension (phones, 493, 1) exited non-zero on 'SIP/492-'


(Initially i installed using yum, i was getting the same issue.
Than i scrapped everything and installed it manually.)



On Fri, Jul 9, 2010 at 8:39 PM, Dan Austin dan_aus...@phoenix.com wrote:

 Manmohan wrote:

  My Web-MeetMe_v4.0.1, i followed the instructions in the
  README File in the same package.
 Good.  There are other instruction packages, but since I wrote
 the README it is the one I am most familiar with.

  Are you using RealTime enabled app_meetme or app_cbmysql
  from the WMM package? 
  i didnt get this actually what do i need to check here? Please
  dont mind but m not so good in opensource world. I try to read and
  understand and on trial n error basis try  to implement things.
  Though had very much interest in learning things.
 Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk
 was in a separate Asterisk application (app_cbmysql).  With version 4 of
 WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe
 application.

 The README in 4.0.1 lists the steps to setup RealTime (database) support
 for Asterisk and MeetMe.  This narrows down the possible problems, since
 we do not need to consider app_cbmysql.

 Did you install Asterisk from a package with yum, or did you compile it
 yourself?

 Dan


 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
Thanks  Regards
Manmohan Singh Jandu
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-08 Thread Manmohan Singh Jandu
Hello Team,

I was looking for audio conferencing solution where i got Web-meetme.
I had installed Asterisk 1.6.2.9 on Centos  5.4. Its perfecting working
fine. I tried using Meetme even meetme app is working perfectly fine.
I installed Webmeetme 4.0 and integrated with my asterisk. When i try to
dial the conference number it take me to an IVR wherein it asks for the
conference number. The time i provide the conference number, asterisk
crashes giving segmentation fault.
I have been trying to google up and checked lot of forums but didnt get any
solution for this yet.

Kernel version -- 2.6.18-194.3.1.el5PAE


-- 
Thanks  Regards
Manmohan Singh Jandu
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-08 Thread Chandrakant Solanki
On Thu, Jul 8, 2010 at 12:21 PM, Manmohan Singh Jandu
manmoha...@gmail.comwrote:

 Hello Team,

 I was looking for audio conferencing solution where i got Web-meetme.
 I had installed Asterisk 1.6.2.9 on Centos  5.4. Its perfecting working
 fine. I tried using Meetme even meetme app is working perfectly fine.
 I installed Webmeetme 4.0 and integrated with my asterisk. When i try to
 dial the conference number it take me to an IVR wherein it asks for the
 conference number. The time i provide the conference number, asterisk
 crashes giving segmentation fault.
 I have been trying to google up and checked lot of forums but didnt get any
 solution for this yet.

 Kernel version -- 2.6.18-194.3.1.el5PAE


 --
 Thanks  Regards
 Manmohan Singh Jandu

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Hi

If you get Segmentation fault. One of core.$ file is created.

Try to use # gdb asterisk core.$ and use bt command.

And then paste error here.

-- 
Regards,

Chandrakant Solanki
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-08 Thread Paul Belanger
On Thu, Jul 8, 2010 at 2:51 AM, Manmohan Singh Jandu
manmoha...@gmail.com wrote:
 crashes giving segmentation fault.

Read doc/backtrace.txt on how to capture and generate a backtrace.

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-08 Thread Dan Austin
Manmohan wrote:
 I was looking for audio conferencing solution where i got Web-meetme.
 I had installed Asterisk 1.6.2.9 on Centos  5.4. Its perfecting working
 fine. I tried using Meetme even meetme app is working perfectly fine.
 I installed Webmeetme 4.0 and integrated with my asterisk. When i try
 to dial the conference number it take me to an IVR wherein it asks for
 the conference number. The time i provide the conference number, asterisk
 crashes giving segmentation fault.
 I have been trying to google up and checked lot of forums but didnt get
 any solution for this yet.

Which instructions did you follow for the integration?  Are you using
RealTime enabled app_meetme or app_cbmysql from the WMM package?  Which
exact version of WMM?

Dan

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Re: [asterisk-users] Asterisk crashes : Failed to start PBX

2009-11-20 Thread Giorgio Incantalupo
Hi Neo,

have you checked your log files? It sometimes happened to me that 
Asterisk crashed without a reason. I discovered my logrotate didn't make 
its dirty work so I had huge log files. I lowered Asterisk log level and 
forced logrotate to work and now I have no more crashes.

Hope it may help. :)

Giorgio.

Neo Anderson wrote:
 Hello,

 I am using Asterisk 1.4.24.1 version in production.
 OS is Centos 5.3 64 bit  RAM is 8 GB.
 I am facing crash in asterisk approx each 12 hour.
 When it crashes I see  below lines in asterisk logs.
 [Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread
 [Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :(
 I debugged asterisk source code in details   I found that it happens 
 because it can not allocate memory to create thread.

 Another thing is, when I check coredump using gdb, it's not showing 
 any debug symbols.

 Would you please let me know how to prevent or resolve this?

 Thanks in advance!!

 --
 Regards,
 voipexpert


 

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Re: [asterisk-users] Asterisk crashes : Failed to start PBX

2009-11-20 Thread Neo Anderson
Hi,

I have already setup to rotate logs hourly  Debug level is 3.
Is there any other possibility of crash?

Thanks in advance!!


--
Regards,
Voipexpert






From: Giorgio Incantalupo gincantal...@fgasoftware.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Fri, November 20, 2009 10:06:06 PM
Subject: Re: [asterisk-users] Asterisk crashes : Failed to start PBX

Hi Neo,

have you checked your log files? It sometimes happened to me that 
Asterisk crashed without a reason. I discovered my logrotate didn't make 
its dirty work so I had huge log files. I lowered Asterisk log level and 
forced logrotate to work and now I have no more crashes.

Hope it may help. :)

Giorgio.

Neo Anderson wrote:
 Hello,

 I am using Asterisk 1.4.24.1 version in production.
 OS is Centos 5.3 64 bit  RAM is 8 GB.
 I am facing crash in asterisk approx each 12 hour.
 When it crashes I see  below lines in asterisk logs.
 [Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread
 [Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :(
 I debugged asterisk source code in details   I found that it happens 
 because it can not allocate memory to create thread.

 Another thing is, when I check coredump using gdb, it's not showing 
 any debug symbols.

 Would you please let me know how to prevent or resolve this?

 Thanks in advance!!

 --
 Regards,
 voipexpert


 

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[asterisk-users] Asterisk crashes : Failed to start PBX

2009-11-18 Thread Neo Anderson
Hello,

I am using Asterisk 1.4.24.1 version in production.
OS is Centos 5.3 64 bit  RAM is 8 GB.
I am facing crash in asterisk approx each 12 hour.
When it crashes I see  below linesin asterisk logs.
[Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread
[Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :(
I debugged asterisk source code in details   I found that it happens because 
it can not allocate memory to create thread.

Another thing is, when I check coredump using gdb, it's not showing any debug 
symbols.

Would you please let me know how to prevent or resolve this?

Thanks in advance!!

--
Regards,
voipexpert


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[asterisk-users] asterisk crashes when calling gtalk user

2009-10-23 Thread Giorgio Incantalupo
Hi all,

I'm using Asterisk 1.4.26.2. Every time I call a gtalk user, Asterisk 
crashes:

-- Executing [6...@inbound:1] NoOp(SIP/8-084894d8, jabbertest) 
in new stack
-- Executing [6...@inbound:2] Dial(SIP/8-084894d8, 
Gtalk/asterisk/pippopi...@gmail.com) in new stack
Segmentation fault

Is anybody experiencing the same? Found any workaround?

Thank you.

Giorgio Incantalupo

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[asterisk-users] asterisk crashes!!!

2009-08-07 Thread Oguzhan Kayhan
Hi,
I got ast. 1.6.0.10 working for a few weeks without a problem.
A few mins ago..I got the following msgs on ast-cli and asterisk service
crashed.

I coudlnt find anything that might cause this problem.
Any ideas??

[Aug  7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein:
Invalid GSM data (1)
[Aug  7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did
not update samples 0
[Aug  7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein:
Invalid GSM data (1)
[Aug  7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did
not update samples 0
[Aug  7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein:
Invalid GSM data (1)
[Aug  7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did
not update samples 0
[Aug  7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein:
Invalid GSM data (1)
[Aug  7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein:
Invalid GSM data (1)
[Aug  7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein:
Invalid GSM data (1)
[Aug  7 13:54:40] WARNING[9517]: file.c:718 ast_readaudio_callback: Failed
to write frame
asterisk1*CLI
Disconnected from Asterisk server



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[asterisk-users] Asterisk crashes

2007-05-11 Thread Elman Efendiyev
Hello,

I have very annoying problem with asterisk 1.4.4:
Every evening when I have peak load asterisk crashes, peak load is only
over 20-30 sip-to-h323 simultaneous calls. Nothing special in logs after
crash. Load average never was higher than 0.3, asterisk never uses more than
12% CPU (according to top). Tried SVN versions - same result. Both h323 and
sip peers has only one codec allowed - g729 - so no conversion. There is no
conferences, call recordings or something like this - very simple setup.

Software config:
Linux Slackware 11.0
Kernel 2.6.21.1
Asterisk 1.4.4 (native h323 channel from asterisk tarball)
Libpri-1.4.0
Zaptel-1.4.2.1 (using ztdummy for internal sync, no zaptel hardware)
pwlib_v1_10_0
openh323_v1_18_0

Hardware config:
Intel SE7210TP1 motherboard
P4 3GHz HT 1Mb cache CPU
1Gb RAM (dual channel, two same DIMMs from intel recommended list)
80Gb SATA HDD
No zaptel hardware or even any PCI cards

There isn't overheating and voltage problems with a hardware (controlling
over IPMI), this hardware (with another HDD and software versions) worked
fine about year with asterisk restarts manually only for a version upgrade.

Could somebody point me the way to debug this problem?

Thank you!

--
Sincerely,
Elman Efendiyev

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[Asterisk-Users] Asterisk crashes at startup

2006-05-31 Thread Jean-Michel Hiver

Hi List,

Yesterday night after a power off due to a faulty UPS my asterisk 
doesn't want to start anymore. Here is what I get on the CLI:


Asterisk Ready.
*CLI
Disconnected from Asterisk server: Bad file descriptor.
Executing last minute cleanups
 == Destroying musiconhold processes
Asterisk uncleanly ending (0).

I use 1.2.7 I think on a debian sarge and cdr_pgsql too.

Any ideas?

Cheers,
Jean-Michel.

--
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Découvrez la Réunion des Technologies IP  Telecom
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Re: [Asterisk-Users] Asterisk crashes at startup

2006-05-31 Thread Robert Rawlinson

You may have file damage. Run the file repair.
Bob Rawlinson

Jean-Michel Hiver wrote:

Hi List,

Yesterday night after a power off due to a faulty UPS my asterisk 
doesn't want to start anymore. Here is what I get on the CLI:


Asterisk Ready.
*CLI
Disconnected from Asterisk server: Bad file descriptor.
Executing last minute cleanups
 == Destroying musiconhold processes
Asterisk uncleanly ending (0).

I use 1.2.7 I think on a debian sarge and cdr_pgsql too.

Any ideas?

Cheers,
Jean-Michel.



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RE: [Asterisk-Users] Asterisk crashes at startup

2006-05-31 Thread Brian C. Fertig
run asterisk with asterisk -c   and see if it gives anymore 
information.  You can also get it to produce a core dump and see if it gives 
you anymore information.

brian


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver
Sent: Wednesday, May 31, 2006 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk crashes at startup

Hi List,

Yesterday night after a power off due to a faulty UPS my asterisk 
doesn't want to start anymore. Here is what I get on the CLI:

Asterisk Ready.
*CLI
Disconnected from Asterisk server: Bad file descriptor.
Executing last minute cleanups
  == Destroying musiconhold processes
Asterisk uncleanly ending (0).

I use 1.2.7 I think on a debian sarge and cdr_pgsql too.

Any ideas?

Cheers,
Jean-Michel.

-- 
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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Re: [Asterisk-Users] Asterisk crashes at startup

2006-05-31 Thread Vladimir Montealegre

look the list of hardware!

http://www.voip-info.org/wiki/view/How+To+Debug+and+Troubleshoot+VOIP

- Original Message - 
From: Jean-Michel Hiver [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Wednesday, May 31, 2006 11:09 AM
Subject: [Asterisk-Users] Asterisk crashes at startup



Hi List,

Yesterday night after a power off due to a faulty UPS my asterisk doesn't 
want to start anymore. Here is what I get on the CLI:


Asterisk Ready.
*CLI
Disconnected from Asterisk server: Bad file descriptor.
Executing last minute cleanups
 == Destroying musiconhold processes
Asterisk uncleanly ending (0).

I use 1.2.7 I think on a debian sarge and cdr_pgsql too.

Any ideas?

Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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Version: 7.1.394 / Virus Database: 268.8.0/353 - Release Date: 31/05/2006




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Re: [Asterisk-Users] Asterisk crashes at startup

2006-05-31 Thread Matt Riddell (IT)
Jean-Michel Hiver wrote:
 Hi List,
 
 Yesterday night after a power off due to a faulty UPS my asterisk
 doesn't want to start anymore. Here is what I get on the CLI:
 
 Asterisk Ready.
 *CLI
 Disconnected from Asterisk server: Bad file descriptor.
 Executing last minute cleanups
  == Destroying musiconhold processes
 Asterisk uncleanly ending (0).
 
 I use 1.2.7 I think on a debian sarge and cdr_pgsql too.
 
 Any ideas?

Maybe the /var/run/asterisk.ctl file still exists and points nowhere?

-- 
Cheers,

Matt Riddell
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Re: [Asterisk-Users] Asterisk crashes at startup

2006-05-31 Thread Jean-Michel Hiver

Robert Rawlinson a écrit :


You may have file damage. Run the file repair.


Rob, thanks for your response.

Which tool would you use to do that?

--
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Découvrez la Réunion des Technologies IP  Telecom
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Re: [Asterisk-Users] Asterisk crashes at startup

2006-05-31 Thread Jean-Michel Hiver

Brian C. Fertig a écrit :


run asterisk with asterisk -c   and see if it gives anymore 
information.  You can also get it to produce a core dump and see if it gives 
you anymore information.
 


That's pretty much what I did, and it says:

[func_uri.so] = (URI encode/decode functions)
 == Registered custom function URIDECODE
 == Registered custom function URIENCODE
 == Manager registered action DBGet
 == Manager registered action DBPut
 == Parsing '/etc/asterisk/enum.conf': Found
Asterisk Ready.
*CLI
Disconnected from Asterisk server: Bad file descriptor.
Executing last minute cleanups
 == Destroying musiconhold processes
Asterisk cleanly ending (0).

Cheers,
Jean-Michel.

--
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Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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Re: [Asterisk-Users] Asterisk Crashes after update

2005-07-24 Thread Scott Brown

Thanks Brian:

Sorry for the late reply.  That was helpful.  Problem solved.

Scott

At 10:43 PM 7/10/2005, you wrote:

You might want to recompile the res_config_mysql or configure
res_config_odbc which works via myodbc and is just as good!

/b
---
Anakin: You're either with me, or you're my enemy.
Obi-Wan: Only a Sith could be an absolutist.

On Jul 7, 2005, at 2:46 PM, [EMAIL PROTECTED] wrote:


After doing an update from SUSE 9.2 to 9.3 and Checking out the
latest from CVS, Asterisk crashes on startup with an apparent MySQL
(res_config_register) error:
# asterisk -vvvgc  asterisk_startup_error1.log
asterisk: symbol lookup error:
/usr/lib/asterisk/modules/res_config_mysql.so: un
   defined symbol: ast_cust_config_register
The log is shown below.  I've seen the posts from
1/25/05 and several more recent ones regarding this
same issue or a similar one with the
ast_cust_config_register being undefined, however
reverting to that build of 1/24/05 does not solve the
problem in my case.
Is there another issue with mySQL that may cause this
problem?  I'm using SUSE 9.3 on an Athlon 64 with 64
bit release 2.6 of Linux.  I've made sure that all the
ODBC and MySQL modules for SUSE 9.3 are installed.
I'm a rank noob with * and would appreciate any help.
Thanks!!!
Log Pasted below for more info:
[0;37;40m
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/asterisk.conf': Found
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/extconfig.conf': Found
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/asterisk.conf': Found
Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
== ===
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started
/var/log/asterisk/event_log
Asterisk Dynamic Loader loading preload modules:
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/modules.conf': Found
[1;30;40m  == [0;37;40mManager registered action
Ping
[1;30;40m  == [0;37;40mManager registered action
Events
[1;30;40m  == [0;37;40mManager registered action
Logoff
[1;30;40m  == [0;37;40mManager registered action
Hangup
[1;30;40m  == [0;37;40mManager registered action
Status
[1;30;40m  == [0;37;40mManager registered action
Setvar
[1;30;40m  == [0;37;40mManager registered action
Getvar
[1;30;40m  == [0;37;40mManager registered action
Redirect
[1;30;40m  == [0;37;40mManager registered action
Originate
[1;30;40m  == [0;37;40mManager registered action
Command
[1;30;40m  == [0;37;40mManager registered action
ExtensionState
[1;30;40m  == [0;37;40mManager registered action
AbsoluteTimeout
[1;30;40m  == [0;37;40mManager registered action
MailboxStatus
[1;30;40m  == [0;37;40mManager registered action
MailboxCount
[1;30;40m  == [0;37;40mManager registered action
ListCommands
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/manager.conf': Found
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/cdr.conf': Not found (No such file or
directory)
Jul  6 21:32:24 [1;33;40mNOTICE[0;37;40m[8492]:
[1;37;40mcdr.c[0;37;40m:[1;37;40m1162[0;37;40m
[1;37;40mdo_reload[0;37;40m: CDR simple logging
enabled.
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/rtp.conf': Found
[1;30;40m  == [0;37;40mRTP Allocating from port
range 1 - 2
Asterisk PBX Core Initializing
Registering builtin applications:
[1;30;40m [0;37;40m[AbsoluteTimeout]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mAbsoluteTimeout[0;37;40m'
[1;30;40m [0;37;40m[Answer]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mAnswer[0;37;40m'
[1;30;40m [0;37;40m[BackGround]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mBackGround[0;37;40m'
[1;30;40m [0;37;40m[Busy]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mBusy[0;37;40m'
[1;30;40m [0;37;40m[Congestion]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mCongestion[0;37;40m'
[1;30;40m [0;37;40m[DigitTimeout]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mDigitTimeout[0;37;40m'
[1;30;40m [0;37;40m[Goto]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mGoto[0;37;40m'
[1;30;40m [0;37;40m[GotoIf]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mGotoIf[0;37;40m'
[1;30;40m [0;37;40m[GotoIfTime]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mGotoIfTime[0;37;40m'
[1;30;40m [0;37;40m[ExecIfTime]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mExecIfTime[0;37;40m'
[1;30;40m [0;37;40m[Hangup]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mHangup[0;37;40m'
[1;30;40m [0;37;40m[NoOp]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mNoOp[0;37;40m'
[1;30;40m [0;37;40m[Prefix]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mPrefix[0;37;40m'
[1;30;40m [0;37;40m[Progress]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mProgress[0;37;40m'
[1;30;40m [0;37;40m[ResetCDR]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mResetCDR[0;37;40m'
[1;30;40m [0;37;40m[ResponseTimeout]
[1;30;40m  == [0;37;40mRegistered application

Re: [Asterisk-Users] Asterisk Crashes after update

2005-07-10 Thread Brian West
You might want to recompile the res_config_mysql or configure  
res_config_odbc which works via myodbc and is just as good!


/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jul 7, 2005, at 2:46 PM, [EMAIL PROTECTED] wrote:

After doing an update from SUSE 9.2 to 9.3 and Checking out the  
latest from CVS, Asterisk crashes on startup with an apparent MySQL  
(res_config_register) error:

# asterisk -vvvgc  asterisk_startup_error1.log
asterisk: symbol lookup error:
/usr/lib/asterisk/modules/res_config_mysql.so: un
   defined symbol: ast_cust_config_register
The log is shown below.  I've seen the posts from
1/25/05 and several more recent ones regarding this
same issue or a similar one with the
ast_cust_config_register being undefined, however
reverting to that build of 1/24/05 does not solve the
problem in my case.
Is there another issue with mySQL that may cause this
problem?  I'm using SUSE 9.3 on an Athlon 64 with 64
bit release 2.6 of Linux.  I've made sure that all the
ODBC and MySQL modules for SUSE 9.3 are installed.
I'm a rank noob with * and would appreciate any help.
Thanks!!!
Log Pasted below for more info:

  == Parsing
'/etc/asterisk/asterisk.conf': Found
  == Parsing
'/etc/asterisk/extconfig.conf': Found
  == Parsing
'/etc/asterisk/asterisk.conf': Found
Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
== 
===

  == Parsing
'/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started
/var/log/asterisk/event_log
Asterisk Dynamic Loader loading preload modules:
  == Parsing
'/etc/asterisk/modules.conf': Found
  == Manager registered action
Ping
  == Manager registered action
Events
  == Manager registered action
Logoff
  == Manager registered action
Hangup
  == Manager registered action
Status
  == Manager registered action
Setvar
  == Manager registered action
Getvar
  == Manager registered action
Redirect
  == Manager registered action
Originate
  == Manager registered action
Command
  == Manager registered action
ExtensionState
  == Manager registered action
AbsoluteTimeout
  == Manager registered action
MailboxStatus
  == Manager registered action
MailboxCount
  == Manager registered action
ListCommands
  == Parsing
'/etc/asterisk/manager.conf': Found
  == Parsing
'/etc/asterisk/cdr.conf': Not found (No such file or
directory)
Jul  6 21:32:24 NOTICE[8492]:
cdr.c:1162
do_reload: CDR simple logging
enabled.
  == Parsing
'/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port
range 1 - 2
Asterisk PBX Core Initializing
Registering builtin applications:
 [AbsoluteTimeout]
  == Registered application
'AbsoluteTimeout'
 [Answer]
  == Registered application
'Answer'
 [BackGround]
  == Registered application
'BackGround'
 [Busy]
  == Registered application
'Busy'
 [Congestion]
  == Registered application
'Congestion'
 [DigitTimeout]
  == Registered application
'DigitTimeout'
 [Goto]
  == Registered application
'Goto'
 [GotoIf]
  == Registered application
'GotoIf'
 [GotoIfTime]
  == Registered application
'GotoIfTime'
 [ExecIfTime]
  == Registered application
'ExecIfTime'
 [Hangup]
  == Registered application
'Hangup'
 [NoOp]
  == Registered application
'NoOp'
 [Prefix]
  == Registered application
'Prefix'
 [Progress]
  == Registered application
'Progress'
 [ResetCDR]
  == Registered application
'ResetCDR'
 [ResponseTimeout]

[Asterisk-Users] Asterisk Crashes after update

2005-07-07 Thread sbrown
After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from 
CVS, Asterisk crashes on startup with an apparent MySQL 
(res_config_register) error: 


# asterisk -vvvgc  asterisk_startup_error1.log
asterisk: symbol lookup error:
/usr/lib/asterisk/modules/res_config_mysql.so: un
   defined symbol: ast_cust_config_register 


The log is shown below.  I've seen the posts from
1/25/05 and several more recent ones regarding this
same issue or a similar one with the 


ast_cust_config_register being undefined, however
reverting to that build of 1/24/05 does not solve the
problem in my case. 


Is there another issue with mySQL that may cause this
problem?  I'm using SUSE 9.3 on an Athlon 64 with 64
bit release 2.6 of Linux.  I've made sure that all the
ODBC and MySQL modules for SUSE 9.3 are installed. 

I'm a rank noob with * and would appreciate any help. 

Thanks!!! 

Log Pasted below for more info: 



  == Parsing
'/etc/asterisk/asterisk.conf': Found 


  == Parsing
'/etc/asterisk/extconfig.conf': Found 


  == Parsing
'/etc/asterisk/asterisk.conf': Found 

Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium. 

Written by Mark Spencer [EMAIL PROTECTED] 

= 


  == Parsing
'/etc/asterisk/logger.conf': Found 


Asterisk Event Logger Started
/var/log/asterisk/event_log 

Asterisk Dynamic Loader loading preload modules: 


  == Parsing
'/etc/asterisk/modules.conf': Found 


  == Manager registered action
Ping 


  == Manager registered action
Events 


  == Manager registered action
Logoff 


  == Manager registered action
Hangup 


  == Manager registered action
Status 


  == Manager registered action
Setvar 


  == Manager registered action
Getvar 


  == Manager registered action
Redirect 


  == Manager registered action
Originate 


  == Manager registered action
Command 


  == Manager registered action
ExtensionState 


  == Manager registered action
AbsoluteTimeout 


  == Manager registered action
MailboxStatus 


  == Manager registered action
MailboxCount 


  == Manager registered action
ListCommands 


  == Parsing
'/etc/asterisk/manager.conf': Found 


  == Parsing
'/etc/asterisk/cdr.conf': Not found (No such file or
directory)
Jul  6 21:32:24 NOTICE[8492]:
cdr.c:1162
do_reload: CDR simple logging
enabled. 


  == Parsing
'/etc/asterisk/rtp.conf': Found 


  == RTP Allocating from port
range 1 - 2 

Asterisk PBX Core Initializing 

Registering builtin applications: 

 [AbsoluteTimeout] 


  == Registered application
'AbsoluteTimeout' 

 [Answer] 


  == Registered application
'Answer' 

 [BackGround] 


  == Registered application
'BackGround' 

 [Busy] 


  == Registered application
'Busy' 

 [Congestion] 


  == Registered application
'Congestion' 

 [DigitTimeout] 


  == Registered application
'DigitTimeout' 

 [Goto] 


  == Registered application
'Goto' 

 [GotoIf] 


  == Registered application
'GotoIf' 

 [GotoIfTime] 


  == Registered application
'GotoIfTime' 

 [ExecIfTime] 


  == Registered application
'ExecIfTime' 

 [Hangup] 


  == Registered application
'Hangup' 

 [NoOp] 


  == Registered application
'NoOp' 

 [Prefix] 


  == Registered application
'Prefix' 

 [Progress] 


  == Registered application
'Progress' 

 [ResetCDR] 


  == Registered application
'ResetCDR' 

 [ResponseTimeout] 


  == Registered application
'ResponseTimeout' 

 

[Asterisk-Users] asterisk crashes

2005-07-06 Thread Tulika Pradhan

hi !

i have the following in my extensions.conf

exten = 2000,1,Wait(60)
exten = 2000,2,Hangup

When i dial '2000' from my phone, I see 'Wait' being called. After 60 secs, 
I also se 'Hangup' being called. If I hangup the phone line before 60 secs 
are over ('Wait' command is probably interrupted in this case), asterisk 
crashes with segmentation fault.


Due to this problem, my 'campon' feature causes asterisk to crash often.

does anyone have an idea as to what this problem might be ?

tulika

_
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http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?74 Find your match on 
BharatMatrimony.com


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[Asterisk-Users] Asterisk crashes with sipp

2005-05-26 Thread Tulika Pradhan


with asterisk running, if i call

sipp ip address -s 9111 -d 6 -r 20 -t un -sn uac -m 60

all the calls get set up, and after a minute when asterisk receives the 1st 
BYE from uac, it responds with
200 OK and then crashes. If i restart asterisk, all the calls get terminated 
properly.


in the extensions, i have
[default]
exten = 9111222,1,Answer
exten = 9111222,3,Wait(600)
exten = 9111222,4,Hangup

please help as i am unable to continue with any load tests !

tulika

_
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vote now at IIFA.


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[Asterisk-Users] Asterisk crashes

2005-05-07 Thread Mark Johnson
Can someone please help me.  I am currently HEAD as of about 5 days ago 
(stable was giving me all sort of problems, upgraded per other users 
suggestions) on an Intel mainboard using a mix of Cisco 7960/40 SIP and 
7910 SCCP.   Can someone please explain what the following means?  When 
this happens, I am about 1 minute from Asterisk going downhill.  All of 
the SCCP phones quit, while the SIP phones can do calling to some 
degree.  I get kicked out of any consoles and can't reconnect without 
restarting asterisk.

Mark
May  7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 
'SCCP/118-001a'
May  7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 
'SCCP/118-001a'
May  7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 
'SCCP/118-001a'
May  7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 
'SCCP/118-001a'
May  7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 
'SCCP/118-001a'
May  7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 
'SCCP/118-001a'
May  7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 
'SCCP/118-001a'
May  7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 
'SCCP/118-001a'
May  7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 
'SCCP/118-001a'
May  7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 
'SCCP/118-001a'
May  7 01:03:13 WARNING[28400] channel.c: Avoided deadlock for 
'SCCP/118-001a', 10 retries!
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Re: [Asterisk-Users] Asterisk crashes on special Transfer with MGCP/ATA 186

2005-03-16 Thread [EMAIL PROTECTED]
Thomas Dingermann wrote:
Hi all,
i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with 
Cisco ATA-186 3.1.1 atamgcp

We are used to make an special ;) blind transfer like 
(Flash)Number(Hangup before anyone answers or ring).
Then * crashes (see below) if the man in the middle is an 
cisco-ata-186-mgcp

If one waits until the last one rings, then hangup, everything is fine.
If one waits until the last one  answers, then hangup, everything is 
fine, too.

Any hints?
mgcp debug on:
  -- Executing AGI(Zap/7-1, nuller.agi) in new stack
   -- Launched AGI Script 
/home/kpj/pbx/var/lib/asterisk/agi-bin/nuller.agi
   -- Accepting call from '01635571857' to '8551' on channel 0/1, span 3
   -- AGI Script nuller.agi completed, returning 0
   -- Executing Dial(Zap/7-1, MGCP/aaln/[EMAIL PROTECTED]||) in new 
stack
   -- MGCP mgcp_request(aaln/[EMAIL PROTECTED])
   -- MGCP cw: 0, dnd: 0, so: 0, sno: 0
   -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
   -- Called aaln/[EMAIL PROTECTED]
   -- MGCP/aaln/[EMAIL PROTECTED] is ringing
   -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
   -- MGCP/aaln/[EMAIL PROTECTED] answered Zap/7-1
gw-bzo*CLI mgcp debug on
Usage: mgcp debug
  Enables dumping of MGCP packets for debugging purposes
   -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf'
   -- Swapping 1 for 0 on aaln/[EMAIL PROTECTED]
   -- MGCP Muting 1 on aaln/[EMAIL PROTECTED]
   -- Started music on hold, class 'default', on Zap/7-1
   -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
   -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '8'
   -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '7'
   -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu'
   -- Stopped music on hold on Zap/7-1
Oct 15 13:32:58 NOTICE[100377]: chan_mgcp.c:1151 mgcp_fixup: 
mgcp_fixup(Zap/7-1, Zap/7-1MASQ)
   -- Swapping 0 for 1 on aaln/[EMAIL PROTECTED]
Oct 15 13:32:58 WARNING[14350]: chan_mgcp.c:3033 handle_request: 
Transfer attempt 
failed  
__
Hey There,
I had a similar problem running CVS HEAD 02/09/05. An Attended transfer 
that wasn't completed caused the channel to lockup. with a bunch of:
Mar 16 14:12:12 WARNING[8904]: channel.c:523 ast_channel_walk_locked: 
Avoided deadlock for 'MGCP/aaln/[EMAIL PROTECTED]', 10 retries!

messages.. Did you ever find a solution??
Thanks,
Brett
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RE: [Asterisk-Users] Asterisk crashes from time to time

2005-02-04 Thread Hecken, Guido

  (There are other debug modes, but not sure I'd use those to catch a
  production problem. The one's I know about are primarily intended for
  development debugging. Other folks might contribute hints here.)
 This reeks of a deadlock,
 http://voip-info.org/wiki-Asterisk+deadlock
 see this
 HowTo Debug a DeadLock in Asterisk
 i wrote up eons ago on the wiki
 http://voip-info.org/wiki-Asterisk+debugging

Thanks for your informations, I will try to follow the instructions on
debugging asterisk.
Since I'm not a programmer, I think I will get some fun with it ;-)

Guido

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[Asterisk-Users] Asterisk crashes from time to time

2005-02-03 Thread Hecken, Guido
Hello List,

we have 3 Asterisk boxes running under Fedora Core 2. Every box
hangs/crashes from time to time.
These installations are image based, means we made an image from our
testserver with an image tool, which is able to manage ext3 partitions and
deployed it to different server hardware.
These servers run very stable and I could not find any failures in the logs.
As these crashes appeared the first time, I thought rebooting these machines
by cronjob every night at 04:00 would solve the problems. It seemed to work
quite well for a couple of weeks. Today I saw our own asterisk production
server crash :-( .
These crashes are always the same, asterisk stops responding, the cli does
not give any reaction on command input, you have to manually kill -9 all
asterisk and moh processes.
Asterisk logs are empty.

We don't use any isdn/fxs/fxo/e1/t1 cards in these servers.
Our connections to PSTN is only made by Patton/Inalp SmartNode Gateways,
connected to asterisk via sip protocol.
Scince these crashes appear on three servers with different hardware, and
the main installation is always the same, I would think there are only two
possible sources to find the failure:

Operating System Fedora Core 2 Kernel 2.6.8-1.521
Asterisk CVS-HEAD-01/08/05

Has anybody out there similar problems, and if yes, how did he fix them?
Is there any working solution, having asterisk control itself perhaps by
using a script that drops a test call in /var/spool/asterisk/outgoing and if
this call wasn't processed successfull the script stops all running asterisk
and moh processes and restarts asterisk?

Any help would be appreciated, since I can't get no sleep with these
timebombs out there ;-)

Guido Hecken



 

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Re: [Asterisk-Users] Asterisk crashes from time to time

2005-02-03 Thread Rich Adamson
 we have 3 Asterisk boxes running under Fedora Core 2. Every box
 hangs/crashes from time to time.
 These installations are image based, means we made an image from our
 testserver with an image tool, which is able to manage ext3 partitions and
 deployed it to different server hardware.
 These servers run very stable and I could not find any failures in the logs.
 As these crashes appeared the first time, I thought rebooting these machines
 by cronjob every night at 04:00 would solve the problems. It seemed to work
 quite well for a couple of weeks. Today I saw our own asterisk production
 server crash :-( .
 These crashes are always the same, asterisk stops responding, the cli does
 not give any reaction on command input, you have to manually kill -9 all
 asterisk and moh processes.
 Asterisk logs are empty.
 
 We don't use any isdn/fxs/fxo/e1/t1 cards in these servers.
 Our connections to PSTN is only made by Patton/Inalp SmartNode Gateways,
 connected to asterisk via sip protocol.
 Scince these crashes appear on three servers with different hardware, and
 the main installation is always the same, I would think there are only two
 possible sources to find the failure:
 
 Operating System Fedora Core 2 Kernel 2.6.8-1.521
 Asterisk CVS-HEAD-01/08/05
 
 Has anybody out there similar problems, and if yes, how did he fix them?
 Is there any working solution, having asterisk control itself perhaps by
 using a script that drops a test call in /var/spool/asterisk/outgoing and if
 this call wasn't processed successfull the script stops all running asterisk
 and moh processes and restarts asterisk?

Far too many variables for anyone to even guess at the root cause. Problem
could be related to slight differences in o/s libraries between systems, 
coding problems within asterisk, etc.

There were some issues reported with cvs head in January relative to hangs,
etc.

Might consider changing /etc/asterisk/logger.conf and add debug to the list.
Then after a failure, at least look at /var/log/asterisk/debug messages.

For additional info, I'd suggest compiling the code on one of thse machines
to see if it complains about missing/inappropriate items.


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RE: [Asterisk-Users] Asterisk crashes from time to time

2005-02-03 Thread Hecken, Guido
Rich,

 Far too many variables for anyone to even guess at the root cause. Problem
 could be related to slight differences in o/s libraries between systems,
 coding problems within asterisk, etc.
You 're right, it could be every thing

 There were some issues reported with cvs head in January relative to
hangs,
 etc.
Are they reported in the bugtracker, or in the mailing list?

 Might consider changing /etc/asterisk/logger.conf and add debug to the
list.
 Then after a failure, at least look at /var/log/asterisk/debug messages.
Yes, this was the first thing, I did after the crash showed up. I simply
forgot to enable it, since this production server ran long time without
problems. But now, following murphy's law, the next crash will never happen
;-)

 For additional info, I'd suggest compiling the code on one of thse
machines
 to see if it complains about missing/inappropriate items.

After these machines were setup, we compiled new code on every machine,
since we started with an older version of Asterisk in November 2004. The
compiling of asterisk did not show me any relevant (?) errors. But I
remember there were some statements (Warnings) in the console output of the
make process, I didn't understand. Is this output logged in addition to the
console in a logfile somewhere?
If so, one could examine this output and hopefully get some hints...

Thanks for your help

Guido Hecken


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RE: [Asterisk-Users] Asterisk crashes from time to time

2005-02-03 Thread Rich Adamson
Inline...

  Far too many variables for anyone to even guess at the root cause. Problem
  could be related to slight differences in o/s libraries between systems,
  coding problems within asterisk, etc.
 You 're right, it could be every thing
 
  There were some issues reported with cvs head in January relative to
 hangs,
  etc.
 Are they reported in the bugtracker, or in the mailing list?

Not all for sure. If you watch the -cvs and -user list, you'd see folks
with seg faults, etc, and not too long after that you see a change
come through -cvs. Sometimes with comments like 'fix silly typo', etc.
Given that cvs head is actually development, at any point in time there
could easily be various problems (expected). To try to recreate
historically whether you caught a cvs head version that had errors is
almost impossible. 

That's why its important to run cvs head in some sort of pre-production
test environment before promoting the code into a customer's machine, etc.
(That implies beating the hell out of your test environment.)

  Might consider changing /etc/asterisk/logger.conf and add debug to the
 list.
  Then after a failure, at least look at /var/log/asterisk/debug messages.
 Yes, this was the first thing, I did after the crash showed up. I simply
 forgot to enable it, since this production server ran long time without
 problems. But now, following murphy's law, the next crash will never happen
 ;-)
 
  For additional info, I'd suggest compiling the code on one of thse
 machines
  to see if it complains about missing/inappropriate items.
 
 After these machines were setup, we compiled new code on every machine,
 since we started with an older version of Asterisk in November 2004. The
 compiling of asterisk did not show me any relevant (?) errors. But I
 remember there were some statements (Warnings) in the console output of the
 make process, I didn't understand. Is this output logged in addition to the
 console in a logfile somewhere?
 If so, one could examine this output and hopefully get some hints...

The only two (key) log methods that I know of is to run the cli with
several -'s, and turn on debugging in the logger.conf 
file (which may require you to config /etc/syslog.conf to catch them).
Then look at /var/log/asterisk/debug after a failure.

(There are other debug modes, but not sure I'd use those to catch a
production problem. The one's I know about are primarily intended for
development debugging. Other folks might contribute hints here.)


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Re: [Asterisk-Users] Asterisk crashes from time to time

2005-02-03 Thread TC
 (There are other debug modes, but not sure I'd use those to catch a
 production problem. The one's I know about are primarily intended for
 development debugging. Other folks might contribute hints here.)
This reeks of a deadlock, 
http://voip-info.org/wiki-Asterisk+deadlock
see this  
HowTo Debug a DeadLock in Asterisk
i wrote up eons ago on the wiki
http://voip-info.org/wiki-Asterisk+debugging
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Re: [Asterisk-Users] Asterisk crashes my router!?

2004-12-04 Thread Steve Totaro



On Thu, 2004-12-02 at 16:47, David Filion wrote:
  Hi,
  Does anybody else have problems like this.
  I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek
  Vigour 2600 ADSL router.
  My * box is configured with a public IP address which is presented on
  one of the switch ports on the rear of the router.
  When there is some SIP activity, incoming mainly, towards my * box,
  the router will lockup after a short period?!
 
 

 Maybe the router can't handle the traffic?  If you have a modem before
your
 router, try connecting * right to the modem and using rpppoe.

The Vigor 2600 should just do fine, I have 2 Vigor 2600VGi's in use with
Asterisk (IAX though) and Handytone's (SIP, not connected to the
Asterisk) in use (router PPPoE though, connected to NetSource here in
Ireland). Firmware is 2.5.3 (i think Draytek withdrew that again because
of GUI problems), but 2.5.2 was fine, too.

The router is SIP aware, so actually you shouldn't think much about NAT
setup etc, it should work straight away. Have you talked with Draytek
about that problem ? Maybe they have heard about it before and have a
solution.

Besides, is it the UK specific firmware you have loaded ?

Slán leat,
Martin List-Petersen
Dublin, Eire
(contact info on -- http://www.marlow.dk/)

Yes, try a firmware upgrade.  I actually saw a router one time that would
lockup if a client behind it ran a trace route

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Re: [Asterisk-Users] Asterisk crashes my router!?

2004-12-04 Thread Mike Dent
Hi Martin,
my router is a vanilla 2600, not the V model, as far as I know it has no special
SIP features, other than SIP seeming to crash it when a SIP call is made from
the internet to the * box here! :(

I mentioned the problem on the draytek forum but I;ve not contacted Draytek 
themselves per se.

One big difference is you are using PPPoE and I'm using PPPoA, unfortunately!

I've tried several different firmware, all UK specific, still the same.

thanks.

Mike


On Fri, 03 Dec 2004 23:39:50 +, Martin List-Petersen
[EMAIL PROTECTED] wrote:


 
 The Vigor 2600 should just do fine, I have 2 Vigor 2600VGi's in use with
 Asterisk (IAX though) and Handytone's (SIP, not connected to the
 Asterisk) in use (router PPPoE though, connected to NetSource here in
 Ireland). Firmware is 2.5.3 (i think Draytek withdrew that again because
 of GUI problems), but 2.5.2 was fine, too.
 
 The router is SIP aware, so actually you shouldn't think much about NAT
 setup etc, it should work straight away. Have you talked with Draytek
 about that problem ? Maybe they have heard about it before and have a
 solution.
 
 Besides, is it the UK specific firmware you have loaded ?
 
 Slán leat,
 Martin List-Petersen
 Dublin, Eire
 (contact info on -- http://www.marlow.dk/)
 

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RE: [Asterisk-Users] Asterisk crashes my router!?

2004-12-03 Thread Martin List-Petersen
On Thu, 2004-12-02 at 16:47, David Filion wrote:
  Hi,
  Does anybody else have problems like this.
  I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek
  Vigour 2600 ADSL router.
  My * box is configured with a public IP address which is presented on
  one of the switch ports on the rear of the router.
  When there is some SIP activity, incoming mainly, towards my * box,
  the router will lockup after a short period?!
  
 

 Maybe the router can't handle the traffic?  If you have a modem before your
 router, try connecting * right to the modem and using rpppoe.

The Vigor 2600 should just do fine, I have 2 Vigor 2600VGi's in use with
Asterisk (IAX though) and Handytone's (SIP, not connected to the
Asterisk) in use (router PPPoE though, connected to NetSource here in
Ireland). Firmware is 2.5.3 (i think Draytek withdrew that again because
of GUI problems), but 2.5.2 was fine, too.

The router is SIP aware, so actually you shouldn't think much about NAT
setup etc, it should work straight away. Have you talked with Draytek
about that problem ? Maybe they have heard about it before and have a
solution.

Besides, is it the UK specific firmware you have loaded ?

Slán leat,
Martin List-Petersen
Dublin, Eire 
(contact info on -- http://www.marlow.dk/)

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[Asterisk-Users] Asterisk crashes my router!?

2004-12-02 Thread Mike Dent
Hi,
Does anybody else have problems like this.
I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek
Vigour 2600 ADSL router.
My * box is configured with a public IP address which is presented on
one of the switch ports on the rear of the router.
When there is some SIP activity, incoming mainly, towards my * box,
the router will lockup after a short period?!

I've tried upgrading firmware in the router but it still locks up?

Thanks

Mike
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Re: [Asterisk-Users] Asterisk crashes my router!?

2004-12-02 Thread Sean Cook
Sounds like it is time for a different router... There are a few routers
out there with buggy nat engines... they are fine when you are doing
typical nat, but if you are trying to do 1:1 nat... get a good router or
make a BSD box to use as a router.  I would highly recommend
http://m0n0.ch/wall for a great do it yourself router with nice web
interface...  It does 1:1 nat as well as bridging packet filter...

Sean

On Thu, 2004-12-02 at 15:54 +, Mike Dent wrote:
 Hi,
 Does anybody else have problems like this.
 I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek
 Vigour 2600 ADSL router.
 My * box is configured with a public IP address which is presented on
 one of the switch ports on the rear of the router.
 When there is some SIP activity, incoming mainly, towards my * box,
 the router will lockup after a short period?!
 
 I've tried upgrading firmware in the router but it still locks up?
 
 Thanks
 
 Mike
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RE: [Asterisk-Users] Asterisk crashes my router!?

2004-12-02 Thread David Filion


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Mike Dent
 Sent: December 2, 2004 10:54 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Asterisk crashes my router!?
 
 Hi,
 Does anybody else have problems like this.
 I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek
 Vigour 2600 ADSL router.
 My * box is configured with a public IP address which is presented on
 one of the switch ports on the rear of the router.
 When there is some SIP activity, incoming mainly, towards my * box,
 the router will lockup after a short period?!
 
 I've tried upgrading firmware in the router but it still locks up?
 
 Thanks
 
 Mike
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Maybe the router can't handle the traffic?  If you have a modem before your
router, try connecting * right to the modem and using rpppoe.

David



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Re: [Asterisk-Users] Asterisk crashes my router!?

2004-12-02 Thread Robbie Hughes
I have exactly the same problem. It occurs with asterisk server based 
locally or external to the local net. My way around it is to use an 
external modem and then use a Draytek 2900 for the dual subnet routing. 
I'm hoping that will solve it as I'm getting dropped calls left right 
and centre... -- Message: 7 Date: Thu, 2 Dec 
2004 11:47:02 -0500 From: David Filion [EMAIL PROTECTED] Subject: 
RE: [Asterisk-Users] Asterisk crashes my router!? To: 'Mike Dent' 
[EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial 
Discussion' [EMAIL PROTECTED] Message-ID: 
[EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mike Dent
Sent: December 2, 2004 10:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk crashes my router!?
Hi,
Does anybody else have problems like this.
I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek
Vigour 2600 ADSL router.
My * box is configured with a public IP address which is presented on
one of the switch ports on the rear of the router.
When there is some SIP activity, incoming mainly, towards my * box,
the router will lockup after a short period?!
I've tried upgrading firmware in the router but it still locks up?
Thanks
Mike
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Maybe the router can't handle the traffic?  If you have a modem before your
router, try connecting * right to the modem and using rpppoe.
David
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Re: [Asterisk-Users] Asterisk crashes my router!?

2004-12-02 Thread Nick Bachmann
Sean Cook wrote:
Sounds like it is time for a different router... There are a few routers
out there with buggy nat engines... they are fine when you are doing
typical nat, but if you are trying to do 1:1 nat... get a good router or
 

I've been very happy with Netopia 3386-ENTs.
make a BSD box to use as a router.  I would highly recommend
http://m0n0.ch/wall for a great do it yourself router
This was so funny that I had to share it:
m0n0wall is probably *the first UNIX system that has its boot-time 
configuration done with PHP*, rather than the usual shell scripts, and 
that has *the entire system configuration stored in XML format*.

There's an excellent reason they're the first: those are both such 
unbelieveably terrible ideas, especially the PHP init scripts. 

I would reccomend IPCop, because their designers are a little more 
grounded.

Nick
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Re: [Asterisk-Users] Asterisk crashes my router!?

2004-12-02 Thread Sean Cook
 I've been very happy with Netopia 3386-ENTs.
 
 make a BSD box to use as a router.  I would highly recommend
 http://m0n0.ch/wall for a great do it yourself router
 
 
 This was so funny that I had to share it:
 m0n0wall is probably *the first UNIX system that has its boot-time 
 configuration done with PHP*, rather than the usual shell scripts, and 
 that has *the entire system configuration stored in XML format*.
 
 There's an excellent reason they're the first: those are both such 
 unbelieveably terrible ideas, especially the PHP init scripts. 
 

Wow, that is one of the most amazing statements I have ever heard... of
course you have worked with m0n0wall and have tested it thoroughly to
have come to such a conclusion.   Of course you have... 

Exactly what is terrible about php init scripts?  Is it that php is a
newer language that may not have originally been designed for such a
thing?  Please educate me...


Sean

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Re: [Asterisk-Users] Asterisk crashes my router!?

2004-12-02 Thread Dinesh Nair
On 03/12/2004 04:01 Nick Bachmann said the following:
There's an excellent reason they're the first: those are both such 
unbelieveably terrible ideas, especially the PHP init scripts.
I would reccomend IPCop, because their designers are a little more 
would you elaborate why these are terrible ideas ? i'm sure, of course, 
that you actually used m0n0wall and evaluated it before coming up with that 
statement.

--
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Re: [Asterisk-Users] Asterisk crashes my router!?

2004-12-02 Thread Steven Critchfield
On Fri, 2004-12-03 at 12:39 +0800, Dinesh Nair wrote:
 On 03/12/2004 04:01 Nick Bachmann said the following:
  There's an excellent reason they're the first: those are both such 
  unbelieveably terrible ideas, especially the PHP init scripts.
  I would reccomend IPCop, because their designers are a little more 
 
 would you elaborate why these are terrible ideas ? i'm sure, of course, 
 that you actually used m0n0wall and evaluated it before coming up with that 
 statement.

[EMAIL PROTECTED]:~$ ls -l /bin/bash -h
-rwxr-xr-x  1 root root 652K Nov 11 00:42 /bin/bash
[EMAIL PROTECTED]:~$ ldd /bin/bash
libncurses.so.5 = /lib/libncurses.so.5 (0x40028000)
libdl.so.2 = /lib/libdl.so.2 (0x40067000)
libc.so.6 = /lib/libc.so.6 (0x4006a000)
/lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x4000)
[EMAIL PROTECTED]:~$ ldd /bin/bash|awk '{print $3}'|xargs ls -lHh
-rwxr-xr-x  1 root root  88K Oct 13 14:40 /lib/ld-linux.so.2
-rw-r--r--  1 root root 1.2M Oct 13 14:40 /lib/libc.so.6
-rw-r--r--  1 root root 9.7K Oct 13 14:40 /lib/libdl.so.2
-rw-r--r--  1 root root 247K May 27  2004 /lib/libncurses.so.5

Or about a total of 2.2 megs

[EMAIL PROTECTED]:~$ ls -l /usr/bin/php4 -h
-rwxr-xr-x  1 root root 2.9M Oct  5 03:49 /usr/bin/php4
[EMAIL PROTECTED]:~$ ldd /usr/bin/php4
libcrypt.so.1 = /lib/libcrypt.so.1 (0x40028000)
libnsl.so.1 = /lib/libnsl.so.1 (0x40055000)
libexpat.so.1 = /usr/lib/libexpat.so.1 (0x4006a000)
libedit.so.2 = /usr/lib/libedit.so.2 (0x4008b000)
libncurses.so.5 = /lib/libncurses.so.5 (0x400a7000)
libpcre.so.3 = /usr/lib/libpcre.so.3 (0x400e6000)
libpanel.so.5 = /usr/lib/libpanel.so.5 (0x400f6000)
libdb-4.2.so = /usr/lib/libdb-4.2.so (0x400fa000)
libbz2.so.1.0 = /usr/lib/libbz2.so.1.0 (0x401d)
libz.so.1 = /usr/lib/libz.so.1 (0x401e)
libssl.so.0.9.7 = /usr/lib/i686/cmov/libssl.so.0.9.7 (0x401f2000)
libresolv.so.2 = /lib/libresolv.so.2 (0x40223000)
libm.so.6 = /lib/libm.so.6 (0x40235000)
libdl.so.2 = /lib/libdl.so.2 (0x40257000)
libc.so.6 = /lib/libc.so.6 (0x4025a000)
libcrypto.so.0.9.7 = /usr/lib/i686/cmov/libcrypto.so.0.9.7 (0x4038e000)
/lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x4000)
[EMAIL PROTECTED]:~$ ldd /usr/bin/php4|awk '{print $3}'|xargs ls -lHh
-rwxr-xr-x  1 root root   88K Oct 13 14:40 /lib/ld-linux.so.2
-rw-r--r--  1 root root  1.2M Oct 13 14:40 /lib/libc.so.6
-rw-r--r--  1 root root   19K Oct 13 14:40 /lib/libcrypt.so.1
-rw-r--r--  1 root root  9.7K Oct 13 14:40 /lib/libdl.so.2
-rw-r--r--  1 root root  132K Oct 13 14:40 /lib/libm.so.6
-rw-r--r--  1 root root  247K May 27  2004 /lib/libncurses.so.5
-rw-r--r--  1 root root   72K Oct 13 14:40 /lib/libnsl.so.1
-rw-r--r--  1 root root   64K Oct 13 14:40 /lib/libresolv.so.2
-rw-r--r--  1 root root 1006K Nov 14 13:43 /usr/lib/i686/cmov/libcrypto.so.0.9.7
-rw-r--r--  1 root root  194K Nov 14 13:43 /usr/lib/i686/cmov/libssl.so.0.9.7
-rw-r--r--  1 root root   61K Nov 24 18:23 /usr/lib/libbz2.so.1.0
-rw-r--r--  1 root root  857K Aug 21 00:27 /usr/lib/libdb-4.2.so
-rw-r--r--  1 root root  106K Aug 30 17:08 /usr/lib/libedit.so.2
-rw-r--r--  1 root root  127K Oct 19 19:34 /usr/lib/libexpat.so.1
-rw-r--r--  1 root root   12K May 27  2004 /usr/lib/libpanel.so.5
-rw-r--r--  1 root root   63K Mar 12  2004 /usr/lib/libpcre.so.3
-rw-r--r--  1 root root   66K Oct 30 13:49 /usr/lib/libz.so.1

Or about 7.2 megs. Do you gain enough by using php to explain an extra 5
megs or so over the normal bash. Of course you could go the busybox
route and be in at a total of 937k or over 6 megs less executables but a
crap load more functionality. 

So quickly you get the fact that on a minimalistic system such as a
firewall, you don't want all those libraries and crap. A true firewall
should be so minimal it would easily fit on a floppy image and be read
only so as not to be very exploitable.

And for a non technical argument, the use of php for the init scripts
smacks of someone who knew php and thought they would reinvent the
wheel(firewall) with the only technology they knew how to use. If true,
I would worry about security.
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] Asterisk crashes my router!?

2004-12-02 Thread Dinesh Nair
ahh, the famed steve critchfield has honoured me.
On 03/12/2004 13:21 Steven Critchfield said the following:
[EMAIL PROTECTED]:~$ ldd /bin/bash
libncurses.so.5 = /lib/libncurses.so.5 (0x40028000)
libdl.so.2 = /lib/libdl.so.2 (0x40067000)
libc.so.6 = /lib/libc.so.6 (0x4006a000)
/lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x4000)
[EMAIL PROTECTED]:~$ ldd /bin/bash|awk '{print $3}'|xargs ls -lHh
-rwxr-xr-x  1 root root  88K Oct 13 14:40 /lib/ld-linux.so.2
-rw-r--r--  1 root root 1.2M Oct 13 14:40 /lib/libc.so.6
-rw-r--r--  1 root root 9.7K Oct 13 14:40 /lib/libdl.so.2
-rw-r--r--  1 root root 247K May 27  2004 /lib/libncurses.so.5
Or about a total of 2.2 megs
interesting, because on m0n0wall,
#ls -al /usr/local/bin/php
-r-xr-xr-x  1 root  wheel  1060380 Jul 18 15:57 php
#ldd /usr/local/bin/php
php:
libcrypt.so.2 = /usr/lib/libcrypt.so.2 (0x2815d000)
libm.so.2 = /usr/lib/libm.so.2 (0x28176000)
libc.so.4 = /usr/lib/libc.so.4 (0x28191000)
# ls -al /usr/lib/libcrypt.so.2 /usr/lib/libm.so.2 /usr/lib/libc.so.4
-r--r--r--  1 root  wheel  580572 Jun 24 04:03 /usr/lib/libc.so.4
-r--r--r--  1 root  wheel   28432 Jun 24 04:03 /usr/lib/libcrypt.so.2
-r--r--r--  1 root  wheel  117024 Jun 24 04:03 /usr/lib/libm.so.2
which makes it 1.7MB by my calculations (1024*1024 bytes per MB).
Or about 7.2 megs. Do you gain enough by using php to explain an extra 5
megs or so over the normal bash. Of course you could go the busybox
so i guess, the question is, do you gain enough by using bash to warrant 
the extra 30% or so ?

So quickly you get the fact that on a minimalistic system such as a
firewall, you don't want all those libraries and crap. A true firewall
perhaps for a CLI based system. take a look at picobsd for something which 
fits off a single 1.44MB floppy. however, m0n0wall also provides VPN, 
captive portal, traffic shaper, DHCP and NAT functionality /with/ a 
web-based GUI for people who're /unfamiliar/ with the CLI.

wheel(firewall) with the only technology they knew how to use. If true,
I would worry about security.
once again, in what manner ? remember, we're discussing the init scripts.
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[Asterisk-Users] Asterisk crashes with Unicall

2004-11-19 Thread Leonardo Gomes Figueira
Hi,
For the last 40 days i've been using Unicall on an Asterisk connected to 
an Ericsson MD-110 PBX.

It was working fine for two weeks when there were just some random calls 
but for the last two weeks when the load increased to between 5 and 10 
simultaneous calls the system became unreliable with 2 main problems:

1- Some dropped calls when the call comes from Unicall: Unicall - 
IAX/SIP. When it comes from Zap (E1 PRI) there is no problem: Zap - 
IAX/SIP.

2- Asterisk crashes 2 or 3 times a day. Always when there is some 
Unicall channel active.

To be sure that the crashes are Unicall related I created an test 
enviroment:

2 servers with the same configuration:
- P4 2.8Ghz
- 512MB
- 1 Digium E100P (connected with each other using a E1 cross cable)
The test was: using an .call file to start a call from 1 server to the 
other on an extension that dial to the first server, that dial to the 
other and so on... until there is no more channels available.

The result: the calls start ringing in both servers until there is no 
more channels free, then they start to timeout and hangup. Until here 
there is no problem, but then suddenly one of the Asterisk servers 
crashes. Sometimes the server that initiated the calls, sometimes the 
other, there is no pattern. (I repeated the test several times and one 
time both Asterisk crashed).

If I change the signalling to E1 PRI and make the same test there is no 
problem (calls ring until no more channels are available and timeout 
after some time).

Some messages from the Asterisk that crashed follows below (Got only the 
last 200 lines, the complete log is 1800 lines / 192 Kb, too big for 
posting here).

Is there some debugging info i can extract from this test and post here 
to help ?

Thanks,
Leonardo
zaptel.conf:
span=1,1,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
unicall.conf:
[channels]
language=br
context=principal_in
rxwink=300
usecallerid=yes
hidecallerid=no
usecallingpres=yes
callprogress=no
restrictcid=no
immediate=no
callwaitingcallerid=no
threewaycalling=no
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
protocolclass=mfcr2
protocolvariant=br,4,4
protocolend=cpe #co on the other server
group=1
callerid=asreceived
context=principal_in
channel=1-15
channel=17-31
extensions.conf:
[principal_in]
exten = _.,1,SetCallerId()
exten = _.,2,Dial(UniCall/g1/${EXTEN},600)
Call file:
Channel: UniCall/g1/
Callerid: 
MaxRetries: 0
RetryTime: 600
WaitTime: 600
Context: principal_in
Extension: 777
Priority: 1
Core file:
Core was generated by `/usr/sbin/asterisk -fg'.
Program terminated with signal 11, Segmentation fault.
#0  0x407fa684 in ?? ()
No debugging symbols on asterisk binary cause it was installed from the 
RPM and the building process strips symbols. I can install other binary 
with debugging if it helps...

Asterisk messages:
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Detected
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Making a new call with CRN 
32769
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx bits 0xD   [2/ 
2/101/  0]
Nov 19 09:41:25 WARNING[7175]: UC event Detected
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 on  [2/ 
2/101/  0]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 on  [2/ 
2/101/  0]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 off [2/ 
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 off [2/ 
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 on  [2/ 
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 on  [2/ 
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 off [2/ 
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 off [2/ 
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 on  [2/ 
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 on  [2/ 
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 off [2/ 
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 off [2/ 
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 on  [2/ 
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 5 on  [2/ 
2/102/101]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 off [2/ 
2/102/105]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 5 off [2/ 
2/102/105]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 6 on  [2/ 
2/102/105]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 5 on  [2/ 
2/102/105]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 6 off [2/ 
2/102/103]
Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 5 off [2/ 
2/102/103]
Nov 19 09:41:26 WARNING[7175]: UniCall: mfcr2 Rx tone 6 on  [2/ 
2/102/103]
Nov 19 09:41:26 WARNING[7175]: UniCall: mfcr2 Tx tone 5 on  [2/ 
2/102/103]
Nov 19 09:41:26 WARNING[7175]: UniCall: mfcr2 Rx tone 6 off [2/ 
2/102/103]
Nov 19 09:41:26 WARNING[7175]: UniCall: mfcr2 Tx tone 5 off [2/ 
2/102/103]
Nov 19 

[Asterisk-Users] Asterisk crashes after call when running as non-root, bug???

2004-11-12 Thread Joost Kraaijeveld
Hi all,

I am using Debian Sarge (2.6) with ISDN4Linux. If I run asterisk as root 
everyting is OK. If I run asterisk as the user asterisk, the programm crashes 
after answering a call. No message in the asterisk logs but the 
/var/log/messages says:

Nov 12 20:42:01 localhost kernel: isdn: HiSax1,ch0 cause: E0010

Is this a bug or is this a known feature that can be solved by properly 
configuring asterisk?

TIA

Joost
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[Asterisk-Users] Asterisk crashes on special Transfer with MGCP/ATA 186

2004-10-15 Thread Thomas Dingermann
Hi all,
i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco 
ATA-186 3.1.1 atamgcp

We are used to make an special ;) blind transfer like 
(Flash)Number(Hangup before anyone answers or ring).
Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp

If one waits until the last one rings, then hangup, everything is fine.
If one waits until the last one  answers, then hangup, everything is 
fine, too.

Any hints?
mgcp debug on:
  -- Executing AGI(Zap/7-1, nuller.agi) in new stack
   -- Launched AGI Script /home/kpj/pbx/var/lib/asterisk/agi-bin/nuller.agi
   -- Accepting call from '01635571857' to '8551' on channel 0/1, span 3
   -- AGI Script nuller.agi completed, returning 0
   -- Executing Dial(Zap/7-1, MGCP/aaln/[EMAIL PROTECTED]||) in new stack
   -- MGCP mgcp_request(aaln/[EMAIL PROTECTED])
   -- MGCP cw: 0, dnd: 0, so: 0, sno: 0
   -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
   -- Called aaln/[EMAIL PROTECTED]
   -- MGCP/aaln/[EMAIL PROTECTED] is ringing
   -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
   -- MGCP/aaln/[EMAIL PROTECTED] answered Zap/7-1
gw-bzo*CLI mgcp debug on
Usage: mgcp debug
  Enables dumping of MGCP packets for debugging purposes
   -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf'
   -- Swapping 1 for 0 on aaln/[EMAIL PROTECTED]
   -- MGCP Muting 1 on aaln/[EMAIL PROTECTED]
   -- Started music on hold, class 'default', on Zap/7-1
   -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
   -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '8'
   -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '7'
   -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu'
   -- Stopped music on hold on Zap/7-1
Oct 15 13:32:58 NOTICE[100377]: chan_mgcp.c:1151 mgcp_fixup: 
mgcp_fixup(Zap/7-1, Zap/7-1MASQ)
   -- Swapping 0 for 1 on aaln/[EMAIL PROTECTED]
Oct 15 13:32:58 WARNING[14350]: chan_mgcp.c:3033 handle_request: 
Transfer attempt 
failed   

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Re: [Asterisk-Users] Asterisk crashes when trying to load G.729module.

2003-07-21 Thread Martin Pycko
Try to install the new codec code that is available in

ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so
place it in /usr/lib/asterisk/modules and restart asterisk (or try to
start it).

There is also a new command available g.729 show license usage and a few
fixes to the code.

Write back about the results.

regards
Martin

On Sun, 20 Jul 2003, Anton Tinchev wrote:

 Before few days i bought few g.729 licenses.
 When i try to load the codec, asterisk crahses.
 I tried with and without oh323 module, same result:
 --
 Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable 
 to initialize va stuff: -1
 --

 Here the ldd result:
 --
 [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so
 libc.so.6 = /lib/libc.so.6 (0x40039000)
 /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000)

 Version information:
 /usr/lib/asterisk/modules/codec_g729b.so:
 libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6
 libc.so.6 (GLIBC_2.2) = /lib/libc.so.6
 libc.so.6 (GLIBC_2.1) = /lib/libc.so.6
 libc.so.6 (GLIBC_2.0) = /lib/libc.so.6
 /lib/libc.so.6:
 ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2
 ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2
 ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2
 ---

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Re: [Asterisk-Users] Asterisk crashes when trying to load G.729 module.

2003-07-20 Thread Jeremy McNamara
You have to run a console with the G.729 due to the voice age library 
lameness.  We run safe_asterisk with a TTY and it seems to be fine.

Jeremy McNamara



[EMAIL PROTECTED] wrote:

Try launching asterisk like this:

screen -d -m asterisk -vvvcn

Aparently there is some bug in the codec.

- Justin

On Sun, 20 Jul 2003, Anton Tinchev wrote:

 

Before few days i bought few g.729 licenses.
When i try to load the codec, asterisk crahses.
I tried with and without oh323 module, same result:
--
Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to 
initialize va stuff: -1
--
Here the ldd result:
--
[EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so
   libc.so.6 = /lib/libc.so.6 (0x40039000)
   /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000)
   Version information:
   /usr/lib/asterisk/modules/codec_g729b.so:
   libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6
   libc.so.6 (GLIBC_2.2) = /lib/libc.so.6
   libc.so.6 (GLIBC_2.1) = /lib/libc.so.6
   libc.so.6 (GLIBC_2.0) = /lib/libc.so.6
   /lib/libc.so.6:
   ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2
   ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2
   ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2
---
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Re: [Asterisk-Users] Asterisk crashes when trying to load G.729 module.

2003-07-20 Thread Anton Tinchev
Is it stable enought?
I mean around 30-40 Incoming SIP connections.
Or i must trash the cisco and put Asteriisk/Digium/speex box?
Jeremy McNamara wrote:
 You have to run a console with the G.729 due to the voice age library 
 lameness.  We run safe_asterisk with a TTY and it seems to be fine.
 
 
 Jeremy McNamara
 
 
 
 [EMAIL PROTECTED] wrote:
 
 
Try launching asterisk like this:

screen -d -m asterisk -vvvcn

Aparently there is some bug in the codec.

- Justin


On Sun, 20 Jul 2003, Anton Tinchev wrote:

 


Before few days i bought few g.729 licenses.
When i try to load the codec, asterisk crahses.
I tried with and without oh323 module, same result:
--
Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable 
to initialize va stuff: -1
--

Here the ldd result:
--
[EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so
   libc.so.6 = /lib/libc.so.6 (0x40039000)
   /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000)

   Version information:
   /usr/lib/asterisk/modules/codec_g729b.so:
   libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6
   libc.so.6 (GLIBC_2.2) = /lib/libc.so.6
   libc.so.6 (GLIBC_2.1) = /lib/libc.so.6
   libc.so.6 (GLIBC_2.0) = /lib/libc.so.6
   /lib/libc.so.6:
   ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2
   ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2
   ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2
---

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RE: [Asterisk-Users] Asterisk crashes when trying to load G.729 module.

2003-07-20 Thread Matthew Hardeman
This is generally indicates a problem with the licensing process (which
is severely flawed and full of bugs) on your server...  Did you make it
through the registration process OK?

Matt Hardeman
PaperSoft

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Tinchev
Sent: Sunday, July 20, 2003 12:18 AM
To: [EMAIL PROTECTED]; [EMAIL PROTECTED];
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk crashes when trying to load G.729
module.

Before few days i bought few g.729 licenses.
When i try to load the codec, asterisk crahses.
I tried with and without oh323 module, same result:
--
Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413
(load_module): Unable to initialize va stuff: -1
--

Here the ldd result:
--
[EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so
libc.so.6 = /lib/libc.so.6 (0x40039000)
/lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000)

Version information:
/usr/lib/asterisk/modules/codec_g729b.so:
libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6
libc.so.6 (GLIBC_2.2) = /lib/libc.so.6
libc.so.6 (GLIBC_2.1) = /lib/libc.so.6
libc.so.6 (GLIBC_2.0) = /lib/libc.so.6
/lib/libc.so.6:
ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2
ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2
ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2
---

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[Asterisk-Users] Asterisk crashes when trying to load G.729 module.

2003-07-19 Thread Anton Tinchev
Before few days i bought few g.729 licenses.
When i try to load the codec, asterisk crahses.
I tried with and without oh323 module, same result:
--
Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to 
initialize va stuff: -1
--

Here the ldd result:
--
[EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so
libc.so.6 = /lib/libc.so.6 (0x40039000)
/lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000)

Version information:
/usr/lib/asterisk/modules/codec_g729b.so:
libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6
libc.so.6 (GLIBC_2.2) = /lib/libc.so.6
libc.so.6 (GLIBC_2.1) = /lib/libc.so.6
libc.so.6 (GLIBC_2.0) = /lib/libc.so.6
/lib/libc.so.6:
ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2
ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2
ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2
---

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