similar to: What web interfaces are available today for debian based Asterisk installation?

Hello everyone I'm trying to make asterisk get a call out using the .call system. The setup is A@H 2.6 This is the content of the file is : <<< Channel: Zap/g0/052MYPHONE MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that your local extensions are kept in the # context called [extensions] # Context: ext-local Extension: 210 Priority: 1 >>> I'm

Hello all, I'm looking for a solution to offer Virtual PBX, to my clients. I just saw software with multi-tenant support and I tested it, but no one likes me enought. Finally, I want to offer this service like a kind of hosting. Has you experience with multi-tenant software? Which has you tested? Has anyone experience about "vhost", "vserver", or something similar to run

Hi! I have an automated dialout via a call file to a mobile. Can I detect when the call is not answered but forwarded to the mobile operator voicebox? I would like to stop the dialout if this is the case. TIA, Mike

Hi everybody, I need to know about sound quailty issues from those who have experience with Tormenta2 PRI Interface. Also how to make it work with new versions of Asterisk and Zaptel. And also suggestion if it is a good idea to switch to some newer card from Sangoma or Digium, or Tormenta should work fine. I have sound breaking many times a day over the trunks. Server is AMD Athlon 2.4 GHz with

Hello everyone The PBX is connected using 4 Line FXO card to the PSTN. I wish to send calls that come to extension X to an external phone number, i.e. call the comes from Line1 would go out using Line{2,3,4}. I wish the user that the extension belongs to him be able to set it. Can this be done ? Can it be done from the user's phone (Sipura 841) ? Can I as the wwwadmin user set it ? Thank

Hello list We've installed asterisk@home 2.6 at the office :) I'm trying to set call hijacking for users The way this should work is this: When call comes in, a user would dial some phone code (like *8# - what we had in the old 1.0.9 setup) and pick up the call. How can I do this for the 2.6 setup ? Can it be done from the AMP web managment portal ? Our setup uses the zapata.conf file

Dear Sirs, let me repost my question again, probably the last one was lost in a huge amount of messages during weekend. I'm actually looking for web-based tool which can do two level of administration: 1) high level, Administrators, can create "domains" 2) lower level, Users, can manage extensions within certain domain. much like asterisk2billing. so, I want users to manage

dear all is there any GUI application with support asterisk 1.2 version i am useing 1.2 and i have fine more about GUI base configuration but i didnt got any GUI package for asterisk 1.2 --------------------------------- Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. -------------- next part -------------- An HTML attachment

I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks

People, I have a Debian box with Asterisk and I've installed the Destar package in order to get web managing of my voip system. After I installed Destar, it runs on "localhost:8080", but my server does not have X-Window to access to it so I can engter the web interface.. So how can I change localhost:8080 to IP_ASTERISK:8080 in order to access Destar via web from another PC ???

Hi, I made the mistake of upgrading both my Linux box (to Ubuntu 6.06) and Asterisk (to 1.2.9.1) at the same time. Now, when trying to compile mpg123 - using the tried and true "make mpg123" -, the build fails with an error make[3]: Entering directory `/usr/src/asterisk-1.2.9.1/mpg123-0.59r' make[3]: *** No rule to make target `\ ', needed by `mpg123'. Stop. Maybe

Hi, I'm trying to compile asterisk 1.2.14 on a Debian Sarge amd64 with kernel 2.6.8-12-amd64-k8 make[2]: Entering directory `/usr/src/asterisk-1.2.14/codecs/gsm' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -fomit-frame-pointer -fPIC -c -DNeedFunctionPrototypes=1 -funroll-loops

Dear all, I'll implement a VoIP system using Asterisk + SIP with softphones; I need to connect LAN and VPN users (about 100-150). What version/installation of asterisk do you recommend tyo me ??? Does Asterisk@Home or Trixbox match to my scenario ???? By the way, I use Debian Etch as OS server. Really thanks. Alejandro --

Hello folks. I've got a strange issue. When I modprobe TE220 I do not see mesages like Launching card: 0 <..> Setting up global serial parameters. You can see how I loaded and unloaded the card for several times - http://asteriskpbx.ru/pastebin/11 lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware

Hi All... I installed the Debian Sarge Asterisk package and in the docs it had the licensing terms for the MOH, explaing that Digium (or someone) had licensed the mucic for distribution as MOH only. That's fine, but I can't find the music! Does anyone know where it can be found? Is there another source of free MOH that sounds good with Asterisk? Thanks...

Hi, I am looking for a simple solution to do this. I wish to have the user to enter their preferred method of connection i.e. for the cheapest solution to their desktop phone or mobile phone, then plan callfile based on the number that user provided and dial to the user. Any suggestions? CK -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey Guys, Wanted a Suggestion..Howz this Xorcom Asterisk?I am using it and till now its fine as currently it is in testing stage with 3-4 users. Any Ideas??? Thanks Sharadindu Mohanty --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment

> you should mv the file (and in the same filesystem, so 'rename' is used) > You might want to chmod or even chown the file first as well. I wrote a little script that does all of this before the .call file is mv'd into the outgoing directory: cp /tmp/test3.call /tmp/test1.call chmod 666 /tmp/test1.call chgrp asterisk /tmp/test1.call chown asterisk /tmp/test1.call mv

Hi Everyone... I am running Asterisk 1.2.13 on Debian "Etch". I installed it from the package. I also installed the web voice mail package, which installed Apache2 and a bunch of other stuff. When I point my browser at my PBX machine, the web page says "It Works!" but of course it does not. It does not seem that Apache is configured to run the vmail.cgi script. In the

Hello Has someone written a web page (preferably PHP) that simply shows what extensions are currently online? Thank you.