Help with installation/upgrading Freepbx

New 2.8 installation issue on Ubuntu 10.04 (Lucid)

mjonescalpoly's picture

I'm installing new freepbx 2.8 on Ubuntu 10.04 (Lucid) and when executing the following:
sudo ./install_amp --username=asteriskuser --password=secret

Getting the following error

Upgrading to 2.8.0alpha2..
-> Running PHP script /usr/src/freepbx-2.8.0/upgrades/2.8.0alpha2/tables.php
PHP Fatal error: Cannot redeclare __parse_DialRulesFile() (previously declared in /usr/src/freepbx-2.8.0/upgrades/2.6.0beta1/tables.php:87) in /usr/src/freepbx-2.8.0/upgrades/2.8.0alpha2/tables.php on line 239


Do I need Asterisk?

lafaverp's picture

My goal is to have a free single residential phone line.
Having read documents until my eyes bled I have FINALLY managed to get a Grandstream HT-386 working with sipgate. Although not completely free it is pretty cheap. The one caveat is that do to port forwarding problems I had to connect directly to the Comcast Modem/Gateway and use a 10.x.x.x from it instead of going through my Belkin router and switch to use my 192.168.0.x LAN. I don't know how much of a problem that might be. Now, my questions:

1) Do I even need Asterisk for what I am trying to achieve.


Streaming MOH

witdirect's picture

Streaming MOH
Sorry for post, but I have read all the threads and links on here and other sites.

I cannot get streaming MOH to work at all.
Please somebody can you give me a detailed step by step instruction on how to set this thing up.

I cannot seem to get my head around this thing.


Newbie: SIP Trunking to Tandberg VCS Control

Shampoo Monkey's picture

Hey Everybody,

I am going to start off by saying, I am a complete noob when in comes to this technology. A few months ago I got put in charge of a new project. Part of the project is integration of a phone system with a video conferencing system. I have the video conferencing system up and running and just started on the phone system. To save money, we have decided to go with an asterisk server and a SIP Trunk for our phone system. I used AsteriskNow 1.7.0 32bit.


CentOS 5
Asterisk 1.6


Warning: Module 'pdo_sqlite' already loaded in Unknown on line 0

ted's picture

I every body,

I am currently installing freepbx with sqlite.
It works fine with mysql but whe I try tu use sqlite3 it doesn't work.
I have the following message : Warning: Module 'pdo_sqlite' already loaded in Unknown on line 0

Yet in my /etc/amportal.conf I have

What does it mean ?
Can somebody help me ?

thanks in advance



Can't configure zaptel in Mandriva

silvaphoenix's picture

I'm trying to install an X100P card. It's recognised in the hardware system but genzaptelconf doesn't work
[root@localhost sbin]# genzaptelconf
grep: /etc/asterisk/zapata.conf: No such file or directory
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
Please can anyone help


Upgrade to freepbx 2.8

rvlewis's picture

Before I upgraded to FreePBX 2,8 an analog trunk hooked to a Digium card would ring in, but Asterisk did not answer the trunk unless the destination extension answered the phone, this allowed the voice mail on the analog trunk to answer instead of the asterisk voice mail, which was the way I wanted it on this trunk. Since the upgrade Asterisk picks up on the first ring. How do I change this back to not answering the trunk?


Sip Registration Intermittent

stephenl's picture


Build - FREEPBX, with Asterisk (Ver.

Im finding my sip registration is very intermitant with my VOIPFONE Trunk

If I save the VOIPFONE Trunk this resolves the problem, but only for a short while, after this save, using the sip show registry command i can see that the registration is happening after 45 seconds

pbx*CLI> sip show registry

Host dnsmgr Username Refresh State Reg.Time Y 3016093 45 Registered Mon, 02 Aug 2010 07:07:31
1 SIP registrations.


x100p clone card (from and dahdi ??

japnuts's picture

Hi all Smile

Apologies in advance for dumb noob questions and lack of info.. I'm totally new to the linux/asterisk/freepbx world.

I have an old dell pc (256mb RAM, 20gb HDD, P3 processor) on which I have installed the latest version of asteriskNOW 32 bit and freePBX

I want to use this as a home & small business PBX with one incoming legacy POTS line and potentially 2 or more SIP trunks in the future.

My first task was to get the system up and running with the existing POTS line.. and I seem to be failing dismally !! Sad