[asterisk-users] Asterisk call limitation
Khaled W. Chehab
kchehab at xplorium.com
Mon Jun 20 16:25:00 CDT 2011
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
>>
>> --
>> _____________________________________________________________________
>> -- 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
>
> 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
>
> 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
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list