Hi

I recently started installing FreePBX over Asterisk Business Edition C3.0.3, I had good results with FreePBX 2.5 and ABE C 2.1.2 and thought it would bring more stability to my deployments...

During the installation everything goes well but now every time the machine boots up there are no calls between extensions, nor can a call from an IVR go to the desired destination. The only way to fix this is by running amportal restart which solves the issue immediately.

Does anyone know where to locate a list of what has been updated according to each release? I'm thinking about updating from v2.2.1 but wanted to see what's new.

Thanks!

Hi All,

I have recently installed Asterisk 1.4.23, Dahdi and FreePBX

everything is running fine.. I need to know about configuring a Digium board TE410P...Do I just modify /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf to specify all the parameters?

I googled it and I came across a thread that said I have to use FreePBX and do not modify chan_dahdi.conf, just add
#include dahdi-channels.conf
#include chan_dahdi_additional.conf

to chand_dahdi.conf and use FreePBX to do the rest...

we have a dedicated server and want to run a 2nd instance of asterisk mapped to another ip to run a ip-pbx for my brothers office. ideally also with it's own freepbx frontend too. is this possible and what is the best way to do it ? must i use xen or is it possible just configuring asterisk to do this ?

martin

I've recently upgraded or better reinstalled my asterisk-machine.
I'm now using the asterisk-packages from debian (1.4.21.2~dfsg-3) and FreePBX 2.6.0.RC2.1.
I've had 2.5 on the system and read about something that brought me to upgrade to the latest 2.6.

The error stays the same so I havent reverted that yet:
app_voicemail.c: No entry in voicemail config file for '23'

The voicemail.conf does have the (proper) line:
23 => 1234,Klaus M Brantl,kmb@deam.org,,attach=yes|saycid=no|envelope=yes|delete=no

I wonder if there is a dialparties.agi problem still - all the extensions show up on the FOP as hung-up:

== Manager 'admin' logged off from 127.0.0.1
-- Executing [1633@from-sip-external:1] NoOp("SIP/192.168.243.24-01034200", "Received incoming SIP connection from unknown peer to 1633") in new stack
-- Executing [1633@from-sip-external:2] Set("SIP/192.168.243.24-01034200", "DID=1633") in new stack
-- Executing [1633@from-sip-external:3] Goto("SIP/192.168.243.24-01034200", "s|1") in new stack
-- Goto (from-sip-external,s,1)

Hi,

I'm trying to configure a SIP trunk to a provider that admits they lack experience when it comes to Asterisk based PBX systems config, and I'm not much help.

I've have FreePBX running with a couple of Xlite phones on two extensions, and they work fine locally.

If i configure the SIP account settings of the Xlite phones to access the SIP trunk provider, they send and receive calls fine.

So my question is, how do i take the working settings from the Xlite phones, and build the proper PEER Details and USER Details in a FreePBX SIP Trunk?

I have several systems on trixbox prior to the fork and a few 2.8.1 after. Can I replace with 2.5 or 2.6rc without any problems and how?
Thanks

The menu item of "module admin" is also missing.
Now what?

What is that manual method of fixing all this? One would **think** it would be in the "Module Documentation" of the documentation on this site, but I cannot find it anywhere.

Thanks

GB

Hello,

I have this special requirement,

I bought a international call, I need to make a special dial plan for this new trunk seem that my old form to make international call.

My requirement are :

Now for make international call I have this dial patterns :

9|005.
9|007.
9|009.

Now , with my new provider I will need :

For USA

9005|1NXXNXXXXXX system call to : 1NXXNXXXXXX
9007|1NXXNXXXXXX system call to : 1NXXNXXXXXX
9009|1NXXNXXXXXX system call to : 1NXXNXXXXXX