similar to: How to exchange/get $variables from/to each channel on cmd Dial

Hello, This question is probabely recurrent, i apologize, but i haven't found a limpid explanation (for me) in mail list, google, and hum.... source code): When use the Command DIAL to ring a group, WHERE is stored the name of the 'winner' who pick up the call ? ($variable = ?), and, step beyond: WHEN (or on EVENT = ?) could we get this variable ? many threads but nothing very

I am trying to use ImportVar to get some information out of a SIP/ZAP channel. I cannot seem to find an example of the syntax, or what variables I can access. Basically, I would like to output which person is being called. i.e: SIP/25 calls SIP/21. 25 executes a macro, and the result is SIP/21. The info that I want is stored in the channel's "Direct Bridge" variable. I have

Hello, I read a thread on the asterisk dev list (call file handling suggestion) May i have some comment/opinion on these two ways below to place a call file in the future ? (from the wiki and the asterisk book but added typos and stupidity come from me) The best is ? (and should work ?) tmsp = the delay in future.. say 100 seconds exten=> ra,n,System(NOW='date %S') exten=>

Hello List, customers telled me problem when moving folders in thunderbird which i can confirm. Following setup on my side: dovecot 1.2.11 with mbox. Thunderbird 3.1 with ready mailaccount and "Server supports folders that contain sub-folders and messages" is disabled. i create folder1 for subfolders and in folder1 a folder2 for mails. Now i create folder3 on toplevel like just like

* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND VIDEO TO MICROBLOGGING! In a surprising move, Digium in partnership with Edvina today released a new channel driver for Asterisk, chan_tweet. The driver connects seamlessly to several microblogging platforms, including Twitter, Facebook, Laconi.ca/Identi.ca and GSM text/SMS. The main feature of this new module is to

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

Recently I wrote about this problem, but it still exist and I can't dial my Xlite SIP Phone So here is the Notice Jul 5 17:14:07 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'Damian Minkov <sip:damian@10.1.1.2>' failed for '10.1.1.11' The * box(10.1.1.2) and the PC(10.1.1.11) on which is the XLite are in the same network Here is part from sip

Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Also, ztcfg shows 4 channels configured, as it should. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below

Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them ......... and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are

I apologize, my English isn't better than the last year So, - why Queue has some options like 'caller can continue in his dialplan' (Dial has g and F options for callee/caller) and not he same option for 'callee' ? - no option in queue to send message (for only calle/caller) when bridged as Dial has - idem to announce periodically 'X minutes before hangup' (in Dial,

Hello group I just update to the newest CVS now I'm not able to get asterisk to start. No error during the make or make install I did a make clean before the make;make install Any help would be great!!!! Here is the output asterisk -vvvvvgcd Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf == Binding realtime_ext to mysql/realtime/extensions_table == Binding

I made a similar question in a previous thread, but there was no answer, so I think I was not very clear making the question. What I need is some configuration that works like "promiscredir=yes" in sip.conf that enables me to do the same thing with transfer (REFER), letting me transfer a sip call to a non local sip address. Thanks in advance, Thiago Abra sua conta no Yahoo!

Hello everyone, Can any body please tell me the compiler options to optimise the speex code for ARM TE architecture. Fixed point is enabled. It currently takes about 700ms to encode 5 sec worth of voice. I need to get it down to under 10ms. Is the option --enable-arm5te-asm supported? I am using speex 1.1.12 version. The configure.ac file does not seem to support --enable-arm5te-asm

Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. > Thanks Olle, > > So am I to understand that you

I just downloaded the newest stable version of WINE and tried to run a .exe file. The program I am trying to run only uses 50mb or RAM and I am getting an error message that says I need at least 50 to run it. I have a 26 inch iMac with a quad core i% processor and 8gb of ram so there should be no memory problem. PS. I am new to Macintosh, so I hope this is'nt too much of an obvious question

Looking at the error I tried moving chan_modem* out of the modules folder and asterisk started and its working again... Not sure what changed in the chan_modem_i4l.so but removing it from the folder fixed my problem. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Hall Sent: Sunday, January 23, 2005

Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is

I added the playback line to my [macro-stdexten] context but when I dail an extension I don't get the "please hold while I try that extension" message. It just dials the extexsion. Do I have a syntax problem somewhere ? exten => 8005,1,Macro(stdexten,8005,Zap/2) exten => 8006,1,Macro(stdexten,8006,Sip/8006) [macro-stdexten] ; ; Standard extension macro: ; ${ARG1} -

Hallo I have a big Problem with oleaut32.dll. Allways i got the Messages: "Unimplemented Function oleaut32.OleLoadPictureEx called at address xy Do you wish to debug it?" I have tryed to link with ln -s /windows/D/Windows/System/oleaut32.dll ~/.wine/fake_windows/Windows/System/oleaut32.dll but no success is there. I have linking all VB6lib and ocx. Programms, which don't

Anyone have any idea what this is talking about. Here is my zapata.conf [channels] switchtype=5ess signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=default musiconhold=default faxdetect=incoming channel => 1-23 Here is my zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for