similar to: AMP 1.10.010 Config Problem

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:

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 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

Hi, I have the newbie guide from AMP's website and (fair enough) it is all about whitebox linux. Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ? Any help appreciated. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050125/6b7a2f61/attachment.htm

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

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

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

Hi!I've installed Asterisk@home and I have a ISDN card,(Cologne Chip Design GmbH ISDN network controller [HFC-PCI](rev 0.2) This is how I installed bristuff: how to install hfc card after unload asterisk and amportal whit amportal stop type "setup" unselect zaptel in system service... and set the lan --->reboot<--- cd /usr/src wget

Hi to all, I use FreePBX version 2.7.0.2 with dahdi. The first problem is with dahdi: At the system startup i can't find a way to start correctly Asterisk with Dahdi. My boot configuration is the following: /etc/rc.d/after.local /usr/sbin/rcdahdi start & sleep 15 /usr/local/sbin/amportal start & /sbin/route add -host 85.38.234.9 gw 192.168.2.1 & /usr/bin/python

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension

Hi!I've this problem to another asterisk@home machine, without digium cards, but only with a bri isdn card.It doesn't connect in the amportal graphical,(it's stopped), if I make tha amportal start command this is the result: STARTING ASTERISK Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 Asterisk died

Hello guys, I'm using Trixbox and I have a PHP application that updates a value in the MySQL asterisk database as an interface to have a dynamic customizable IVR. After execute the UPDATE SQL query, the php application is supossed to reload asterisk or restart amportal in order to get the change working, but nor asterisk -rx reload nor amportal restart got the change working. So, the

Hi there, I want to use asterisk for sip comminication with max 1000 users Which gui shuld i use for adding users and managing asterisk? I tried AMPortal, it added extensions to mysql but asterisk did not find users i added ? installed asterisk 1.2.2 on FC4 Toygun

Hello, This is what what I see after a Yum install asterisk16 asterisk16-config freepbx: Use of uninitialized value in string ne at /var/www/html/panel/op_server.plline 4997. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5439. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5440. Use of uninitialized value

Hi there, We did a clean install the AsteriskNOW 1.7.0 64 bits ISO and configured it. On the main screen (Crtl-ALT-F1) we keep seeing the following lines when making a call 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

Hello all, After compiling successfully Asterisk and AMPortal, I cannot make the fax module work. Asterisk does not start (unless I remove the modules or mark them as Noload in modules.conf) with the following error: Jun 3 20:55:25 VERBOSE[3328]: [app_rxfax.so]Jun 3 20:55:25 WARNING[3328]: /usr/local/lib/libspandsp.so.0: undefined symbol: dds_modf Jun 3 20:55:25 WARNING[3328]: Loading

Hello, I am implementing a small call center with 1 to 4 agents. For some reason I don't understand, if an agent is busy, and it is his/her turn in the queue round, asterisk gives an "all destinations are busy" message and hangs up the call. Agents are SIP lines registered with an audiocodes MP108FXS which registers each line independently. Ringing strategy is RoundRobin (most of

Hi, I am using Asterisk personally at home. My SIP client (SPA 3000) supports MWI with SIP NOTIFY messages. With a previous version of Asterisk I had no problems with MWI. But now I am using the following version which comes with Trixbox 2.8.0.1, and I have problems with MWI. Asterisk 1.6.0.9-samy-r27 Problem description: When a voicemail is left on the extension, a SIP NOTIFY message is sent

When I drag and drop a call from the PSTN to a SIP phone (the SIP phone's icon) using FOP .25 with * 1.0.9 with the intent of transferring it, the called party gets transferred rather than the calling party. This is controlled by the reverse_transfer parameter in op_server.cfg but the behavior is exactly the same whether the parameter is set to 0 or 1. This is after the call is picked up by

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