[asterisk-users] Asterisk call limitation
Khaled W. Chehab
kchehab at xplorium.com
Mon Jun 20 15:47:00 CDT 2011
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
More information about the asterisk-users
mailing list