search for: asterisk18

...asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted [ OK ] (reverse-i-search)`d': /etc/init.d/asterisk restart [root at asterisk1 ~]# rpm -qa | grep asterisk asterisk-sounds-core-en-gsm-1.4.21-1_centos5 asterisk18-1.8.4.4-1_centos5 asterisk18-core-1.8.4.4-1_centos5 asterisk18-doc-1.8.4.4-1_centos5 asterisk18-dahdi-1.8.4.4-1_centos5 asterisk18-configs-1.8.4.4-1_centos5 asterisk18-voicemail-1.8.4.4-1_centos5 [root at asterisk1 ~]# uname -a Linux asterisk1 2.6.18-238.el5 #1 SMP Thu Jan 13 15:51:15 EST 2011 x86_...

Hello, Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM version from Asterisk repo I found that asterisknow-version is needed by package asterisk18-core-1.8.7.0-2 How could this be explained? Best regards, Ioan ######### [root at localhost ~]# yum update asterisk18* -x asterisknow-version Loaded plugins: fastestmirror, km...

Hi there, Per the instruction from http://www.asterisk.org/downloads/yum , I setup the yum repository on my Centos 5 x86_64 machine and did a yum install asterisk18 asterisk18-configs then I startup the asterisk (with no changes to config) just to see if it runs, but see below errors in the /var/log/asterisk/messages: [Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module 'res_pktccops': /usr/lib/asterisk/modules/res_pktccops.so: cannot open...

...asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted [ OK ] (reverse-i-search)`d': /etc/init.d/asterisk restart [root at asterisk1 ~]# rpm -qa | grep asterisk asterisk-sounds-core-en-gsm-1.4.21-1_centos5 asterisk18-1.8.4.4-1_centos5 asterisk18-core-1.8.4.4-1_centos5 asterisk18-doc-1.8.4.4-1_centos5 asterisk18-dahdi-1.8.4.4-1_centos5 asterisk18-configs-1.8.4.4-1_centos5 asterisk18-voicemail-1.8.4.4-1_centos5 [root at asterisk1 ~]# uname -a Linux asterisk1 2.6.18-238.el5 #1 SMP Thu Jan 13 15:51:15 EST 2011 x86_...

Still a couple of questions...... I did configure extconfig.conf ... ;iaxusers => odbc,asterisk ;iaxpeers => odbc,asterisk ;sipusers => odbc,asterisk sipusers => mysql,asterisk,sip_devices sippeers => mysql,asterisk,sip_devices ;sippeers => odbc,asterisk ;sipregs => odbc,asterisk ;voicemail => odbc,asterisk ;extensions => odbc,asterisk ;meetme => mysql,general

I used to use the local channel to create a global variable (dialplan) [default] exten => s,1,Set(GLOBAL(LSESSION)=${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)}) to that end, I modified cli.conf [startup_commands] originate local/s extension s at default = yes But now I upgraded to Asterisk18 and there is no longer a local channels Does anybody have any idea of a workaround? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230816/9fcf7cc3/attachment.html>

Hi All, While I'm certainly comfortable compiling from sources, I'm trying to do an rpm only asterisk install on CentOS 5.7. I'm using the asterisk repositories and I installed all the asterisk18 rpms, but find that chan_gtalk and res_jabber are missing. Is there a separate rpm that includes support for gtalk? Thanks in advance. -Gaurav -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111127...

...omeone calls the queue the first time the call is sent to the agent without issue. The issue is that any calls after the first are placed in the queue and never sent to the agent who is logged in and available. Before I call the queue I do a "show queue" and it shows the agent as Asterisk18*CLI> queue show irock.com has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/9013XX9XX8 (dynamic) (Not in use) has taken no calls yet No Callers Then the call comes into the queue and the callee just sits in the...

HI, I tried to configure Asterisk 1.8 on one of my test-hosts. I've installed from centos-asterisk.repo (http://packages.asterisk.org/centos/$releasever/tested/$basearch/): Nov 26 15:34:56 Installed: asterisk-sounds-core-en-gsm-1.4.20-1_centos5.noarch Nov 26 15:34:59 Installed: asterisk18-core-1.8.0-1_centos5.i386 Nov 26 15:35:02 Installed: asterisk18-voicemail-1.8.0-1_centos5.i386 Nov 26 15:35:03 Installed: libpri-1.4.11.5-1_centos5.i386 Nov 26 15:35:04 Installed: libtonezone-2.4.0-2_centos5.i386 Nov 26 15:35:04 Installed: asterisk18-dahdi-1.8.0-1_centos5.i386 Nov 26 15:35:04 Insta...

On Wed, 16 Aug 2023, Federico wrote: > But now I upgraded to Asterisk18 and there is no longer a local channels Are app_originate.so and res_clioriginate.so loaded? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST https://www.linke...

...e > > (dialplan) > > [default] > > exten => s,1,Set(GLOBAL(LSESSION)=${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)}) > > to that end, I modified cli.conf > > [startup_commands] > > originate local/s extension s at default = yes > > But now I upgraded to Asterisk18 and there is no longer a local channels > > Does anybody have any idea of a workaround? > > -- http://help.nyigc.net/

Hi I have followed https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS%2FRedHat%29, to my surprise there is only one config file by the name zapata.conf under /etc/asterisk/ There are no other config files. Any thing i am missing ? Please suggest/guide. Regards, Kaushal

...ut] exten => s,1,Set(agent=${EXTEN:4}) exten => s,n,Set(queue=irock.com) exten => s,n,NoOp(Queue logout agent ${EXTEN:4} from queue ${phone}); exten => s,n,RemoveQueueMember(${queue}); exten => s,n,Playback(agent-loggedoff) When I do a "queue show" I see the following Asterisk18*CLI> queue show irock.com has 0 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 0s talktime), W:0, C:0, A:15, SL:0.0% within 0s Members: SIP/9012211610 (dynamic) (Unavailable) has taken no calls yet No Callers So after I have logged in the agent by dialing *50...

Hello, I can do simple, "yum install asterisk18-*" and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, *"You do not appear to have the source for the 2.6.32-4-pve kernel installed".* * * 1-...

...t; On Behalf Of Steve Edwards Sent: Wednesday, August 16, 2023 11:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Alternative to Local channel On Wed, 16 Aug 2023, Federico wrote: > But now I upgraded to Asterisk18 and there is no longer a local channels Are app_originate.so and res_clioriginate.so loaded? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST https://www.linke...

...> (dialplan) > > [default] > > exten => s,1,Set(GLOBAL(LSESSION)=${STRFTIME(${EPOCH},,%Y-%m-%d > %H:%M:%S)}) > > to that end, I modified cli.conf > > [startup_commands] > > originate local/s extension s at default = yes > > But now I upgraded to Asterisk18 and there is no longer a local > channels > > Does anybody have any idea of a workaround? > > -- http://help.nyigc.net/

i have installed asterisk 1.8 following this doc http://www.asterisk.org/downloads/yum i installed the package asterisk18-voicemail-imapstorage-1.8.2.2-1_centos5 in order to store voicemail in imap but the application voicemail is not available when i type core show application ? in the asterisk log file i have these messages [Feb 3 19:00:20] WARNING[14311] loader.c: Error loading module 'app_voicemail_imapstor...

Hi, I'm having a weird problem with asterisk (asterisk18-1.8.12.2_1). Every call on the system, whatever it comes from the PSTN or from local extensions, when we hit the '#' button to transfer the call, asterisk just disconnects it, without any error or log, here is my current features configuration: Builtin Feature Default Current ------------...

Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call

Hi all, A week or so down the list, i read that not many people were using realtime on an Asterisk18, so i had this afternoon a go at it... [sorry for the inconveneant line-wraps] First i did: mysql> create database asterisk; mysql> grant all on asterisk.* to 'voipadmin'@'localhost' identified by next i used the info from the wiki: CREATE TABLE `sip_devices` ( `id` int(1...