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? 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
