check this out: http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/258425
> From: kche...@xplorium.com > To: asterisk-users@lists.digium.com > Date: Tue, 21 Jun 2011 13:25:39 +0300 > Subject: Re: [asterisk-users] Asterisk call limitation > > Any update ? > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@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@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@localhost ~]# > > -----Original Message----- > From: Khaled W. Chehab [mailto:kche...@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@localhost ~]# ulimit > unlimited > [root@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-boun...@lists.digium.com > [mailto:asterisk-users-boun...@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" <kche...@xplorium.com> > wrote: > > > > > I tried the ulimit > > > > [root@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@from-trunk:4 Up Playback(ss- > > noservice) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- > > noservice) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- > > noservice) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- > > noservice) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- > > noservice) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- > > noservice) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- > > noservice) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- > > noservice) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- > > noservice) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- > > noservice) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- > > noservice) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) > > > > 100 active channels > > 100 active calls > > 6407 calls processed > > > > [root@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: kche...@xplorium.com > > MSN ID :khalidche...@hotmail.com > > Web Site: http://www.xplorium.com > > > > -----Original Message----- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@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" <kpflem...@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:kche...@xplorium.com> kche...@xplorium.com > >>> > >>> MSN ID :khalidche...@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: kflem...@digium.com | SIP: kpflem...@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? 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