similar to: Error of FreePBX after installing from Yum Repository of Asterisk

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

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

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

Every few seconds I get the following message: == Parsing '/etc/asterisk/manager.conf': Found == Connect attempt from '127.0.0.1' unable to authenticate I'm trying to track down where it's coming from. I've used TCPDUMP & NGREP to monitor 127.0.0.1, no data's flowing. I've tried loading Asterisk with no modules, tried loading with a naked

Hello, I've found a couple of previous posts on this subject, but with no posted resolution... I'm attempting to run * as a non-root user (asterisk), following the guidelines on the wiki: http://voip-info.org/tiki-index.php?page=Asterisk%20non-root I can run * as my new user with "/usr/sbin/asterisk -vvvvc" without problem. However, I'm unable to run * using

Hi All, Whenever i try to start FOP using script ./op_panel_redhat.sh start given in directory /usr/local/op_panel-snapshot/init I got the following error: Starting Flash Operator Panel: execvp: No such file or directory [FAILED] Please let me know the reason for this. Thanks in Advance With Regards, newbie

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

Hi, is it possible to force FOP to reload its configuration files (op_buttons.cfg and op_style.cfg) while it is working? I tried to click on the refresh icon but nothing happens. TIA Giorgio Incantalupo

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Hi, Anyone got AFOP working. The install instructions tell you to copy all of the files extracted under the 'html' directory to a subdirectory under your main web directory (in my case this is /var/www/html/panel/) and then the instructions talk about modifying the 'op_server.cfg' file but they do not tell you were to put this file. There is something wrong with the

I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Asterisk is started with the default safe_asterisk script with a console on TTY9. The coloured text on this console is made up of weird characters instead of normal. Please see http://www.softins.co.uk/dsc00018.jpg for an example. If I do "asterisk -rvvvvv" on a normal login, either via the

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:

Hi Guys, I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: [Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy, avgtone: 75, avgsilence 135 [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [h at macro-dialout-trunk:1] Macro("SIP/2111-b6a400b0", "hangupcall|") in new stack [Jul

Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing "sip set debug peer PROVIDER": Sending to 123.123.123.123 : 5060 (no NAT) ^^^^ That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see

Hello, I have written an agi script that uses google voice API for voice recognition. The script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable. More info and dialplan examples can be found in the README file:

Hi Guys, I am looking to use System() function along with some bash scripting to determine if a Trunk is being used during certain time of the day or not. Here is what I have in mind. Please guide me if you know a better way: exten => s,1,answer exten => s,n,System(/tmp/check.sh) check.sh: check EPOCH time => do an IF for certain times => Allow mutiple calls in certain times and

I'm pleased to announce that Flash Operator Panel 0.26 has been released! FOP is a GPL'd switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in FreePBX, Asterisk@Home, DeStar, startShop, and several other projects both free and commercial. You can grab the

I have an *ipphone 7965G* which has to be connected to Asterisk. It has been flashed with SIP firmware but the config file doesn't seem to work maybe I am missing something in it. I appreciate it if you can share your working sample config file with me. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Everyone, I am stack between two identical systems (2U Twin2, 4 nodes, SuperMicro) servers that have the same exact specs except for HDDs. These nodes will all either have Asterisk installed with CentOS or will have Asterisk install in virtual environment. Option 1: *12* x 3.5" HDD (3 HDDs per node) Option 2: *24* x 2.5" HDD (6 HDDs per node) **both options come to the same price.

Hi Everyone, We have three different Queues set to "leastrecent" strategy and from time to time I hear someone complain that they receive short rings (partial ring cycle) and since it's not their turn even if they pickup the phone the call is not given to them since the Queue is actually hitting someone else at the same time. Is this short ring an indication of some sort for