[asterisk-users] Asterisk call limitation

Khaled W. Chehab kchehab at xplorium.com
Tue Jun 21 05:25:39 CDT 2011


Any update ?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Tuesday, June 21, 2011 12:40 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk call limitation


The problem remains  even when 

I add to /etc/init.d/asterisk
ulimit -n 65536

[root at localhost ~]# ulimit -a
core file size          (blocks, -c) 0
data seg size           (kbytes, -d) unlimited
scheduling priority             (-e) 0
file size               (blocks, -f) unlimited
pending signals                 (-i) 65536
max locked memory       (kbytes, -l) 32
max memory size         (kbytes, -m) unlimited
open files                      (-n) 1024
pipe size            (512 bytes, -p) 8
POSIX message queues     (bytes, -q) 819200
real-time priority              (-r) 0
stack size              (kbytes, -s) 10240
cpu time               (seconds, -t) unlimited
max user processes              (-u) 65536
virtual memory          (kbytes, -v) unlimited
file locks                      (-x) unlimited
[root at localhost ~]#

-----Original Message-----
From: Khaled W. Chehab [mailto:kchehab at xplorium.com]
Sent: Tuesday, June 21, 2011 12:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Asterisk call limitation

Can  you please specify more 

1-how to set the ulimit on
[root at localhost ~]# ulimit
unlimited
[root at localhost ~]# ulimit --help
-bash: ulimit: --: invalid option
ulimit: usage: ulimit [-SHacdfilmnpqstuvx] [limit]
---------------------------------------------------------------------
How to set the ulimit command on in  /etc/init.d/asterisk Since there is  no
parameter for ulimit in the file

Thanks in advance

Regards



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Satish Patel
Sent: Tuesday, June 21, 2011 12:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk call limitation

Oh! Wait you set ulimit for running shell    You should set ulimit on  
asterisk. Also you can set ulimit command on asterisk startup file /
etc/init.d/asterisk and restart asterisk also you can set in limit.conf file

I had this issue before and I solved that way.

--
Sent from my iPhone

On Jun 20, 2011, at 4:47 PM, "Khaled W. Chehab" <kchehab at xplorium.com>
wrote:

>
> I tried the ulimit
>
> [root at localhost ~]# ulimit
> Unlimited
>
> Then
> sipp -sn uac -d 10000 -s 2005 127.0.0.1 -l 150
>
> SIP/127.0.0.1:5061-0 s at from-trunk:4       Up      Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:4       Up      Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:4       Up      Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:4       Up      Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:4       Up      Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:4       Up      Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:4       Up      Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:4       Up      Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:4       Up      Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:4       Up      Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:4       Up      Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s at from-trunk:5       Up      SayAlpha(2005)
>
> 100 active channels
> 100 active calls
> 6407 calls processed
>
> [root at localhost ~]#
> I find in  /var/log/asterisk/full
>
> [Jun 20 09:43:17] NOTICE[9705] pbx_ael.c: AEL load process: verified 
> config file name '/etc/asterisk/extensions.ael'.
> [Jun 20 09:43:17] VERBOSE[3068] chan_unistim.c:  Reloading 
> unistim.conf...
> [Jun 20 16:43:33] WARNING[12353] file.c: Failed to write frame [Jun 20 
> 16:43:34] WARNING[12389] file.c: Failed to write frame [Jun 20 
> 16:43:35] WARNING[12394] file.c: Failed to write frame [Jun 20 
> 16:43:43] WARNING[12484] file.c: Failed to write frame [Jun 20 
> 16:43:44] WARNING[12488] file.c: Failed to write frame [Jun 20 
> 16:43:52] WARNING[12573] file.c: Failed to write frame [Jun 20 
> 16:43:57] WARNING[12625] file.c: Failed to write frame [Jun 20 
> 16:44:07] WARNING[12723] file.c: Failed to write frame [Jun 20 
> 16:44:14] WARNING[12789] file.c: Failed to write frame [Jun 20 
> 16:44:22] WARNING[12872] file.c: Failed to write frame [Jun 20 
> 16:44:26] WARNING[12908] file.c: Failed to write frame
>
> Khaled  Chehab
>            NGN Eng.
>
>
>      Operations Office - Lebanon
>      Office : +961 1 868686 ext 115
>      Mobile: +961 3 045212
>      E-mail: kchehab at xplorium.com
>      MSN ID :KhalidChehab at hotmail.com
>      Web Site: http://www.xplorium.com
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Satish 
> Patel
> Sent: Monday, June 20, 2011 11:24 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk call limitation
>
> It could be your OS limit try ulimit command.
>
> --
> Sent from my iPhone
>
> On Jun 20, 2011, at 2:21 PM, "Kevin P. Fleming" <kpfleming at digium.com>
> wrote:
>
>> On 06/20/2011 01:09 PM, Khaled W. Chehab wrote:
>>> Dears,
>>>
>>>
>>>
>>> i am using sipp to test asterisk(1.6.22) performance ,but when i 
>>> limit the calls to 150 ,only 100 active calls on asterisk found ?why
>>>
>>> sipp -sn uac -d 10000 -s 2005 127.0.0.1 -l 150
>>
>> You did not provide any log output, or anything that could be used to 
>> try to help you understand your problem. Without any details, any 
>> reply you get would be just a guess, nothing more.
>>
>>>
>>>
>>>
>>>
>>>
>>> Regards
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Khaled  Chehab
>>>
>>>           NGN Eng.
>>>
>>>
>>>
>>> Description: xplorium
>>>
>>>     Operations Office - Lebanon
>>>
>>>     Office : +961 1 868686 ext 115
>>>
>>>     Mobile: +961 3 045212
>>>
>>>     E-mail:<mailto:kchehab at xplorium.com>  kchehab at xplorium.com
>>>
>>>     MSN ID :KhalidChehab at hotmail.com
>>>
>>>     Web Site: http://www.xplorium.com
>>
>> Please refrain from including 20-line signature blocks in your 
>> messages to the Asterisk mailing lists (or really, anywhere). Your 
>> message had three lines of content and 30+ lines of non-content.
>>
>> --
>> Kevin P. Fleming
>> Digium, Inc. | Director of Software Technologies
>> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype:
>> kpfleming
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at 
>> www.digium.com & www.asterisk.org
>>
>> --
>> _____________________________________________________________________
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>
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