similar to: Software only Asterisk PBX (commercial)

I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI

Sherwood, Your intentions are noble and your desire to build this, fullfills an immediate need for business. If your intention is just to build a GUI for Asterisk, read no further. If your desire is to build something more purposeful, your best bet would be to see the existing commercial GUI/HostedPBX offerings like Pbxware and Switchware from bicomsystems.com ( http://www.bicomsystems.com)

Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP:

List, The D'Link phones are not reliable at this time. I am trying to get them fix their Firmware to my specifications. It is half done so far. However there are still hurdles. below email is self explanatory. At present if you want to use these phones, you need to buy D'Link's SIP Server and run this as one of your SIP servers in the blend to call to Asterisk. Seshu Kanuri "G

Hey all, First-time poster and been following the list for some time. Great postings, all... Currently we're implementing the voipconnection vs-1 for a customer. It's a decent device, and the included admin interface leaves a lot to be desired... The idea of the storing settings compact flash is nice, although sometimes they choose not to stick. All in all, we're thinking about

I am trying to download the latest release of Asterisk from: ftp://ftp.digium.com/pub/asterisk/ The link provided by Digium is incorrect for the Asterisk Tarball as there is no such file at ftp://ftp.digium.com/pub/asterisk/asterisk-1.0.7.tar.gz However the links for the Asterisk-Addons and other Tarballs is OK ftp://ftp.digium.com/pub/asterisk/asterisk/asterisk-addons-1.0.7.tar.gz Does anyone

There was some discussion in the past about which one is the best Content Management System that can be used in conjunction with Asterisk. Mambo was supposed to be the best out there under GPL. The guys who developed Mambo have a new product now - Joomla. I am using this and it appears to be better than Mambo in many respects. Read the gist about Joomla below. ------------- If you've read

Folks! Netweb Group, Inc. fully supports connectivity to any Asterisk PBX systems you have and can provide A-Z termination with immediate effect. Any volume is good enough for us, even an amount as small as $1.00 a day will do for us. We will provide connectivity from our Softswitch IP 216.162.116.46. If anyone is interested, add this to your Asterisk IPBX and then email me for setting up a

I have setup Grandstream to connect to my Asterisk Server. All the digits 0-9 are accepting dtmf. But When I try to send the call by Pressing # Key, nothing happens. Does anyone has a standard configuration for Asterisk and Grandstream as a PDF file or something to see. How do you send the connect signal? Seshu Kanuri -----Original Message----- From: asterisk-users-admin@lists.digium.com

I am using PBXware for configuring users and extensions. Pbxware uses Internal script called init.sh to process the calls based on its own version of extensions.conf defined in the GUI. I have IAX2 Extensions 56 and 101 and SIP extensions 50 and 51. I have used IAX2 extension 101 and dialed SIP Extension 51 But the PBXWare's Init.sh AGI command identifies the DNIS as another IAX

Why is no one suggesting any solution here for this problem, which has been lingering for a while. Are we not supporting H.323 on Asterisk? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of ruixun wu Sent: Thursday, July 22, 2004 4:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] still can't

Folks! Let me clarify something to the Asterisk community about the CDR tool. 1) This is *not* my code to start with. I picked the original code from this forum here... http://www.voip-info.org/tiki-index.php?page=Asterisk%20CDR%20Areski%20GUI 2) The original code was not working (for most part, as the MySql portion has bugs) and I fixed this and added a few bells and whistles. 3) The

I just bought two new Linksys WRT54G routers. Sveasoft has loaded Linux on this router and included a bunch of Linux tools, one of which is Bandwidth Management. The QoS aspect of this is supposed to be much more granular than the previous solution (Wonder Shaper). I have not been able to find any suggestions for how to impliment QoS (Bandwidth Management) using the web interface of Alchemy.

I have followed the Idiots' guide for installation, but still could not make it work. When I try to login at the web page coming from /var/www/html/areski , I get the following errors: Can some body give me some hints where and what to check for this error?. I am looking for info on the changes we have to make for 1) the database name 2) user name 3) password 4)connection name (server

Abel, In have the same issue when I have burned the image to an 800MB CF Disk. All it displays is GRUB CLI in a continuous stream. Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of abel Sent: Monday, June 06, 2005 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Folks! We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found. Here is the config I have used: ------------------------------- Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2 User1 is in USA on Broadband Cable User2 is in India on 64Kbps ISDN Line User1 using SIPURA SPA 2000 user2 using Xten professsional(X-pro)

Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated

Eric Wrote: ----------- The trick is not to use options you don't understand. "show application dial" will show you what the t and T options are for. Most people use the transfer feature of their phone, rather than using the T/t hack on the Dial line. Sounds like you are using CVS-HEAD and so will have to configure stuff in /etc/asterisk/features.conf. /Snip/ Eric, Thanks for

Hello everyone, I am developing an online SMDR / call log system for asterisk. This is going to take the form of an executable with embedded sql and webserver, pdf generation, excel generation, graphs. Actually, we have been selling this for a while now with great success and now I am starting work on the integration with Asterisk. Its a windows executbale and the executable is just about 1MB.

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --