similar to: How to set fromuser in sip.conf so each user gets it's own callerid?

This one puzzles me a lot: [thba at vink layout]$ nc script/xw_functions.ample NEdit: getcwd() fails: No such file or directory NEdit: getcwd() fails: No such file or directory [thba at vink layout]$ ll script/xw_functions.ample -rw-rw-r-- 1 thba thba 16829 Oct 25 16:59 script/xw_functions.ample [thba at vink layout]$ pwd /home/thba/workarea/colibri/design/ana/layout [thba at vink layout]$

Hi all, how can I pickup a call from a queue? Which context parameter do I have to specify? The context that calls the queues application is ext-queues. This is what I tried so long (777 is the extension of the queue I want to pickup from): exten => _**ZXX,1,Noop(Attempt to Pickup ${EXTEN:2} by ${CALLERID(num)}) exten => _**ZXX,n,Pickup(${EXTEN:2}@from-did-direct) exten =>

I have a quad core CPU running Centos5. When I use top, I see that running processes use 245% instead of 100%. If I use gkrellm, I just see one core being used 100%. top: PID USER PR NI VIRT RES SWAP SHR S %CPU %MEM TIME+ COMMAND 18037 thba 31 15 304m 242m 62m 44m R 245.3 4.1 148:58.72 ic Also in the log of some programs I see this strange factor: CPU Seconds = 2632

Hello, I'm having troubles while using the "Goto" function in a realtime extension. Here is the error message : -- Executing Goto("SIP/siemens1-081f56b0", "script_13_0|s|1") -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context

On my CentOS5 boxes the automounter fails after the last update to autofs.x86_64 1:5.0.1-0.rc2.177.el5 No update is seen on CentOS6 and things still work there. The home directory of users is setup using an auto.home map which is distributed using NIS: * fsutrecht02:/export/home/& /nobackup -noacl nasutrecht01:/nobackup/& /snapshot -ro fazant:/home_backup/ This maps a users

Hello, I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have some trouble with the CDR userfield that is not changed when using the SET command in the realtime dialplan. In my dialplan (extensions.conf, the file) I'm setting the userfield like this : exten => s,n,Set(CDR(userfield)="X") Later, my dialplan switches to the realtime part and this is an

So I am compling the newest release of samba 3.0.14a on Solaris 2.8. I can get it to compile but I don't get all the built in modules that I should and smbd when fired up bombs out with something like, ------- derek@supernova:/opt/UMsmb/sbin# ./smbd -c /etc/samba/smb.conf -i smbd version 3.0.14a started. Copyright Andrew Tridgell and the Samba Team 1992-2004 No builtin nor plugin backend for

Hello, I wanted to set my own time intervals for shutdown. {poweroff,return,paused.return}, so I had to change sources, where it has been set in a hard way. poweroff: Sxx\r return: Zxx\r paused.return: SxxRyyyy\r Solution A: Why not adding an info about these commands in a generic way and issuing shutdown.{poweroff.xx,return.xx,paused.return.xxyyyy} Solution B: Why not creating a special

I have working with a polycom IP500 phone. I like the idea of having each line button on the phone as a separate sip device. If I understand it right, each phone could have three extensions (one for each line.) This would be great since I could then use the dialplan to forward calls to the desired extension. I envision something like this: Extenson 101 - Company-A Extension 102 - Company B

I just upgraded from ver. 0.9 to 1.0.5 on Gentoo (from portage) and when I try to reload my settings I get: == Unregistered 'mysql' CDR backend == Parsing '/etc/asterisk/cdr_mysql.conf': Not found (No such file or directory) My phone or SPA-3000 will not register with asterisk either. I've copied the Sip.conf; extenson.conf and iax.conf from old version to 1.0.5 -- #Joseph

Hello, It seems that i spent days and days trying to make Asterisk do automatic dial-outs and i am clueless as i have tried everything i could. I would like Asterisk to automatically dial-out a specific number and leave a goodbye message. However, the outbound call isn't triggered. Even after i reload asterisk and restart it many times. I am using a TDM400 card from Digium with 4 Fxo ports

Hello everybody, I have a strange comportment with oh323 and asterisk, I'start testing asterisk but with this I can't understant plesae help me ! Thanks Eltorio ---------------------------------------------------------- 1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a Modem[i4l] line ---------------------------------------------------------- Nothing happens

Since upgrading to b129, I''ve noticed that my system clock is ahead by five hours, exactly how much I am behind UTC (US Eastern Time). The strange thing is it only happens when I have xen enabled. If I ''svcadm disable milestone/xvm; reboot'' the clock is correct again. If I correct the time manually, ntp knocks it right back ahead again. Oddly enough, even ntp knows

Strange things are happening at my asterisk box :) I've got asterisk setup to dail out with sip to my SIP provider. I've got NO fromuser/fromdomain stuff setup in my sip.conf When I place a call with my Siemens Optipoint 400 SIP phone everything is allright, the From: header is stating the username of the Siemens phone. When I place a call with X-Lite the From: header is altered and reads

How can I force my users to be obliged to give a 'fromuser' and 'fromdomain' -parameter in their SIP-configuration ?? Is this set in the [general] -section of sip. conf ?? Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091109/43dbedb0/attachment.htm

Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user762 at

Hello ! I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18 The things work up to 80%, I can transfer the call by BLF button and I can see the green (free) status and red (busy) status. What I cannot do is to accept the call when someone rings a remote extension. The BLF button starts to blink in red telling me that the call is ringing on remote extenson, but if I press it,

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The Wine development release 1.9.23 is now available. What's new in this release (see below for details): - HID bus support for macOS. - Color font support in Direct2D. - More color formats in Direct3D. - Uninstallation fixes in MSI. - Various bug fixes. The source is available from the following locations: http://dl.winehq.org/wine/source/1.9/wine-1.9.23.tar.bz2