similar to: Latest Stable

Hi list! Do anybody has success histories about using the Beronet app_saynumber application? For those that are hearing about for the first time, it?s a piece of software that enables the use of another language in say_number commands in asterisk dialplan or AGI scripts. Link to download: http://www.beronet.com/download/app_saynumber-beta-rc1.tar.gz I?m trying to compile it in

We are running * V1.0.9 on a demo box. We have set up everything in our dialplan and we have a directory where we store individual extension settings. That directory is called "extensions-phones.d" and it contains a number of .conf files. In my extensions.conf file I have put a #include "extensions-phones.d/*.conf" in my [globals] context If we reload and restart *, and

Hi all, I'm struggling some time now with this problem. Googling and searching on this topic did not deliver the answer yet, so my last hope is this list. Analogue to the things which are possible with modem.conf, where I can configure the MSN's to act on, I would like to have similar functionality. This is the idea: I have 1 ISDN line, it can be reached by 4 different MSN's. I have

Hi list, I have an asterisk box running on a via C3 motherboaard/Debian Sarge. Installed version was the Debian packages one 1.0.7-bristuff. I use this box with the console dial command and it was working fine. Cards info are: cat /proc/assound/cards 0 [V8235 ]: VIA 8233 - VIA 8235 VIA 8235 at )xe400, irq 11 Now I installed the bristuff+asterisk 1.0.9 and always have in my logs

I previously had Asterisk 1.0.7 running on a Linux 2.4.x kernel with ztdummy. I was able to do things like meetme and music on hold. I recently installed Asterisk 1.0.9 on a different machine with a Linux 2.6.x kernel running ztdummy. I installed and configured everything the same way, but when I try to call into a conference room I get the error message stating, "that is not a valid

Hello! Im pretty new of asterisk world: my goal would be using iaxcomm to call ppl over POTN. Yesterday i configured asterisk to be able to hear a song (with Playback() command) with iaxcomm, and it was wonderful. Now, today i tried to configure asterisk to use my ISDN4Linux supported card (a plain HFC card) with asterisk. i configured modem.conf this way, after reading

Hi everyone. I've installed Asterisk PBX using apt packages, but i don't have actually any Digium card, so i want to use ztdummy. I've tried to modify the Makefiles in the debian source package, i don't get any error, but still the ztdummy module doesn't get compiled. Does anybody has idea how to get the ztdummy module using the debian package system? Thanks for your help.

I'd like to these three things about asterisk: 1. How the asterisk program can be configured to run as a different user from root. 2. what directories and files it must have read and right access to 3. Setup an asterisk group, which also has some of the rights the asterisk user has rights to, and what else it can be used for. Are these some info sources which go into these areas in depth?

Hi, is there any good calculator/table/reference about proper dimensioning? I read the wiki and they basically say "xx users run fine in yy hardware" http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning. SO far I read that: -Run up to 4 E1s per CPU (which one? an i386 or a dual core? -it is very CPU intensive to do transcoding. Try to minimize it. -you can help the CPU

Hi all, I just installed an TDM02B. My system is a dell pc with linux 2.6.12-1.1456_FC4 asterisk-1.2.0-beta1 zaptel-1.2.0-beta1 libpri-1.2.0-beta1 in /etc/zaptel.conf I have (all others are default): fxsks=3-4 <--- I saw light in the ports channels=1-2 <--- change it to 3-4 has same result but... [root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel

Folks: does anybody have an idea? how to capture the DTMF digits to a file, after an extn asnwer? then POST it to a url? Regards, JR

Hi. I have installed asterisk 1.0.7. I need to send dtmf with tone duration. This functionality is in the cvs version. Can any body send me the app_senddtmf.so binary compiled for i386 or pentium IV to replace the 1.0.7 version.? I want to preserve the rest of 1.0.7 version of asterisk, because I use the debian package and I want to maintain the asterisk upgrades and patches. Thanks.

If anyone could tell me what this error is all about, I would be very grateful. Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not permitted Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not

Hello, I have been noticing the following behaviour with the monitor command.. Normally it records to the default location and then uses soxmix to create the correct wav file. But for some reason sometimes it doesn't use /var/spool/asterisk/monitor/.. but //var/spool/asterisk/monitor/.. (notice the 2 // in front!) Here is some logging: monitor executing ( nice -n 19 soxmix

Hi, I have read the wiki entries on IAX(2), but I'm afraid, it still have some questions: I have a working connection between two Asterisk-Servers (Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on Debian 3.1) via IAX. Does this connection work with IAX or IAX2? When I try to load chan_iax2.so, I get the error message chan_iax2.so]Oct 11 10:09:52 WARNING[2288]: loader.c:258 ast_load_resource:

Hi! I'm using Debian Sarge with the asterisk-packages (1.0.7) installed. I want to use the cdr_addon_mysql, but there is no DEB. So I tried to compile it, but w/o success. I get an error of a missing asterisk.h. I searched for that file on my system , but it is non-existent, although I installed asterisk-dev. Is there something which I still have to install or do I have to compile the

Hi, I wouldlike use webvmail on my asterisk, I use debain Sarge with asterisk 1.0.7 package. I have installed package "asterisk-web-vmail" but when i go to http://MyAsteriskBOx, i have a page of presentation of Apache. Could you help me please? Thanks

Hi, i am running * 1.0.9 with a newer version of the TE405P. Modprobe wct4xxp and ztcfg are OK. zap show channels only shows me the following. my zapata.conf: [pstn] switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown usecallingpres=yes busydetect=no ; not need on pri callprogress=no ; was yes but wiki says experimatley could be produce hangups callwaitingcallerid=yes ; show

/bin/sh: -c: line 0: syntax error near unexpected token `;' /bin/sh: -c: line 0: `if [ -n "" ]; then if [ -f ]; then mv -f .bak ; fi; cat .bak | grep -v "alias char-major-250" | grep -v "post-install torisa /sbin/ztcfg" | grep -v "post-install wcfxsusb /sbin/ztcfg" | grep -v "alias wctdm" | grep -v "post-install wctdm

Hi all! I'm running a SuSE Linux 9.3 on a little book pc which is based on a VIA CPU and Chipset: cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 7 model name : VIA Samuel 2 stepping : 3 cpu MHz : 532.776 cache size : 64 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no