similar to: error in FreePBX

Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound

Hi, i try install FreePbx by tuturial in http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Paso&view_comment_id=13443 but i have this error when i try install freepbx: #pear install db No releases available for package "pear.php.net/db" Cannot initialize 'db' , invalid or missing package files Package "db" is not valid install

Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: root@hernandezz-laptop:/home/hernandezz# asterisk -rvvvvvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i

HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has

I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP

HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get this error: "Couldn't load variables.txt?aldope=xxxxx " I search in the google and see a sugestion to edit line flash_dir=/var/www/html/panel/flash in file op_server.cfg. Any Sugestion please? --

Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the

Hi, I don't explain very well what my problem, but i can't make calls. i analise my log full and i found two errors Jul 3 19:02:08 ERROR[4670] res_config_mysql.c: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. Jul 3 19:02:08 VERBOSE[4670] logger.c: -- Added extension 'exit-FAILED' priority 1 to macro-vm Jul 3 19:02:08

Hi All, kind of need help on the conference module, i'm using freepbx and enabled conferencing, i created a conference number, 6000. when i dial to it, my phone says it is connected but i'm hearing nothing, maybe logs below can help you. also, when i hang up the phone, the conference did not disconnect me. how can i end a conference? thank you -- Executing

Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani

Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of my FXS zap extension created. dialparties.agi: Starting New

Hi, how i have to do for receive a email with a alert from my voice mail? My doubt is what I put in ?serveremail? in file voicemail.conf. I think is a email server, but can be see anyone? I searching one in the internet? Thanks and sory my english -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I upgraded ocfs from 1.2 to 1.4 after the update I launched tunefs.ocfs2 to enable the new ocfs2 features (sparse files and unwritten extents). tunefs.ocfs2 is now running since 2 days (12T partition) and I need my system back to production, can I safety abort tunefs.ocfs2? thanks Nicola

Hi, i use asterisk with freePbx for all configurations. Now i want use a openser with asterisk but also with freepbx. I pretend use the asterisk whit freepbx, but autentications for users in openSer.. it's possible?? thanks -- Carlos Jer?nimo

I have a problem with my internal extensions, I'm using Asterisk 1.6.2.5 and freePBX 2.6. When I call betwen extensions these don't connect. There is a long silence and finally hang up. I have an E1 whit r2 and I use openr2, but I don't have problems to do calls to the PSTN......my problem it only with internal extension. Please help!!!!...it's an Asterisk bug??..... Thanks :S

I have Asterisk 1.2.29, Zaptel 1.2.24 and Freepbx Setup for a queue up to 15 agents through a PRI line, it was working fine for more than 1 year, suddenly, when there is a load on the queue, the asterisk service disconnects and the calls are dropped. And the service starts again after few seconds, and so on. I am not using fax. I checked PRI by zttool and there are no alarms. The cdr logs

Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow.

Hello. System: Mandrake 10.1 (spanish) Wine: 0.9 Program: GTalk (google talk) http://www.google.com/talk/ I recently installed GTalk (i was almost sure installation will fail, but it didn't). The package was built with NullsoftPimp. The program runs excepcionally fine (the windows looks perfect no problem etc.) but i got 2 problems. I cannot connect with the server. Wine displays a

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before

<speex-dev at xiph.org> Hi all, I'm a begginer with DSP and i need your help and suggestions. I'm trying to use the Speex and a DSP DM642 to implement a solution with voice! I'm using the 1.2 beta 3 distribution and the TI's Code Composer Studio v3.1 simulator. I coosed speex_C64_test.pjt and modified the speex_C64_test.cmd to only use the DM642 external RAM memory. I'm