similar to: AMP with SUSE 9.2

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:

Hello all, The "Secret Agent" final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script

Does anyone know of any obscur reference for detecting an incoming fax. I currently have AMP running and everything else is working great. Installed the spandsp patches and software... using the default AMP extensions.conf, I start sending a fax, I hear it pick up and transfer to voicemail after 20s. Fax is set for system... Here is the detail from the extensions.conf [global] FAX_RX = system

I have a fresh install of AMP. In the AMPortal, Setup, Devices or Users, I get: Cannot connect to Asterisk Manager with user/password (set respectively) This module requires access to the Asterisk Manager. Please ensure Asterisk is running and access to the manager is available. I checked /etc/amportal.conf, /var/www/html/panel/op_server.cfg, and most of the conf files in /etc/asterisk/ Am

If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

I know almost nothing linux, and don't really know that much about Asterisk (proper).. but I was 'pulled' by this subject and grabbed an <mailto:Asterisk@home> Asterisk@home installation CD (version 1.3) and just went with it. Newbie doesn't quite describe it, I'm a banker.. this simply goes to show how easy Asterisk is becoming (I certainly hope this direction was meant

Hello All, I'm new to the list and the whole voip server side. I'm trying to setup Asterisk to just do internal dialing, no access out to the pstn is required/wanted at the moment. I'm running Fedora Core 3 with Cisco 7960's phones (running SIP 6.3). I've set it up following these guides: http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3

Dear forum, I have a install of asterisk using AMP. I followed the install guide off the AMP site. http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.4.pdf When I start using amportal start or asterisk -ccccv I received this in my log. The last line is that ogg failed. I have found nothing on the web about this, and I am not even sure where to start troubleshooting. Any help

Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know". My third repost: Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has

Hi All, Does anyone know if AMP can work with Fedora Core 3? I've tried to install AMP, and * on FC3, but found FOP failed to start when you use the command amportal start. Please advice if anyone got the solution. Chichi --------------------------------- Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more. -------------- next part -------------- An

I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you suggest and why please ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.com <http://www.successfulhosting.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041128/8e282c51/attachment.htm

Greetings. I am trying to get AMP up and going on my Asterisk server. I can access the admin pages on my asterisk server via a web browser. I can add and edit things via the web browser and it edits the database accordingly. Everything seems fine except when I try to run 'amportal start'. Below is what I get (Plus tail /var/log/asterisk/full, but the tail of the 'full' log

Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden stuff, no dice. Any ideas? Thanks... -Ken unidencom.txt: OverwriteLocalSettings

I'm looking at deploying some Polycom 501's here, but one thing that still needs confirmation before I can move forward is global paging. I figure that I can couple polycom auto-answer (http://www.voip-info.org/tiki-index.php?page=Polycom+auto-answer+config) with this script: http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html However, that script was posted over a

Hi, I've compiled Asterisk cvs stable (few days ago) unde Suse 9.2 without any problems. We're using te110p and wcte11xp module that is autoloaded by Suse 9.2. Card goes green after reboot, but this meesages appear in logs: Mar 22 11:28:51 linux kernel: Zapata Telephony Interface Registered on major 196 Mar 22 11:28:51 linux modprobe: FATAL: Error inserting torisa

I installed oh323 and everything seemed to go smoothly (compile & everything upto calling through using oh323). I must admit, there is some behavior that's doesn't seem right but generally, I'm able to dial-out of any oh323 device whether to an extension or to a trunk. Audio is sometimes muted when dialing out until the extension or dialed number answers. Sound quality is good

Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards "standardizing" installed modules, functionalities, tools etc. The "wall" we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to

Hi everyone, i am face with an asterisk use management interface, at the pressent, i am using AMP (asterisk Management Portal: http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 ). Does anyone know a better and more documented management interface for * ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at

Hello all, A new version of the Asterisk Management Portal is available for download. Please visit the AMP homepage at http://amp.coalescentsystems.ca Upgrade instructions are at http://amp.coalescentsystems.ca/UPGRADE Use our Sourceforge mailing list and forum for discussions about AMP. 1.10.006 ChangeLog: - Use extensions_custom.conf for customizations. Sample included. - Added option