Installation

Help with installation/upgrading Freepbx

Calling system on IAX through Asterisk to phonesystem on SIP.

Johnnyphive's picture

Hi.
We are trying to get our calling system to connect to our helpdesk through Asterisk.
We are using FreePBX 2.8.0.2

Helpdesk can utlilize 8 channels and the system is set up like this.
Callingstations => callingcentral=>Asterisk=>AlcatelSIPcentral=>Helpdeskphones.
And vice versa.

How do I set up Asterisk to handle the IAX calls from Callincentral through to the SIP sentral and Helpdesk phones.

We are located in Norway, in case that is a crutial info.

All input is much appriciated.

Thanx all.

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Polycom Ip 330

ka9tgn's picture

Ok, I downloaded the Asterisk Now distro, 1.7.0 from Asterisk.org I can add the extensions using Freepbx that came with it but cannot do anything with the phone. I was able to add a soft phone and that came up just fine, but the polycom ip 330 does not want to dial anything. I have checkd and rechecked all settings. I looked at the Status on the phone and it shows it is registered but yet Freepbx dashboard status shows 0 ip phones online. If I add the soft phone it shows as 1 IP phone online but It should show the polycom as well (2 devices) and does not.

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Now what should I do?

thalesrr's picture

Hello everybody.
Newbie here!
I'm using AsteriskNOW and its and AMAZING system.
Well, not properly using it, yet...

So, I follow the instructions on Asterisk website and my system is up and running.
My question is:
Now what? What should I do? How can I test the system?

I have a Digium VB0408PCI
FreePBX 2.7.0.5
AsteriskNOW 1.7.0

I can access the system via browser.
When I go Basic >> DAHDI
I got this:

Digital Hardware

Span Alarms Framing/Coding Channels Used/Total Signalling Action
Analog Hardware

Type Ports Action
FXO Ports --
FXS Ports --

Forums: 

FreePBX 2.8.0 installation error

rgazali's picture

The following errors is created when installing FreePBX 2.8.0 with asterisk 1.4.35:

Installing new FreePBX files..OK (668 files copied, 0 skipped)
Configuring install for your environment..amportal..fpbx..freepbx_engine..done
apply username/password changes to conf files..done
creating missing #include files..OK
OK
Setting permissions on files..OK
Checking for upgrades..48 found
Upgrading to 1.10.007beta1..
-> Running SQL script /usr/src/asterisksrc/freepbx-2.8.0/upgrades/1.10.007beta1/tables.sql

Forums: 

IAX Trunk in High Availability solutions

kris04051980's picture

Hi,

I need help to fix the IAX trunk issue in High availability solution. I installed the following in centOS 5.4.

asterisk 1.6.2.2
freepbx-2.7
drbd
heartbeat.

I managed to install two asterisk servers with above mentioned modules. With that floating IP my fail over solution working fine. All the phones and SIP gateways registered to primary server using floating IP and its working fine.

Forums: 

Cannot change ARI password [Solved]

lelik67's picture

Modified
#nano -w /var/www/html/recordings/includes/main.conf.php
and edited $ARI_ADMIN_PASSWORD="new password"

Still can log to ARI with admin/ari_password

restarted web server
#/etc/rc.d/init.d/httpd restart

restarted asterisk
#amportal restart

rebooted the server

still the same: admin/ari_password works, and my new password does not.

Verified:
#nano -w /var/www/html/recordings/includes/main.conf.php
and $ARI_ADMIN_PASSWORD="new password".

Am I missing something?

Forums: 

Unresponsive User Portal Options (Menu item)

tislam's picture

Since I upgraded to the current version of FreePBX, the selectable options in the User Portal (Recording) no longer work. Anything I choose (click) like Voicemail, Phone Features, Follow Me, Feature Codes, Settings, etc., the portal just terminates (exit/logout). It does not matter who logs in, whether a regular user or an admin.

What changed for which it is behaving in this manner and how do I fix it?

Thanks so much.

Sincerely
Towhid Islam

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Please urgent help !

Hayssam's picture

Hello there !
myserver was working fine , but sometimes i get problem connecting the devices to the server i always need to rebbot the server and the adsl modem many times to be able to let extensions connect again , i have public static ip does this made any problem please help !

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Zap exten not dial tone fxs

techmike's picture

Hi all,

I have a system asterisknow , and as this is with freepbx maybe I can get some help here.

I have a tdm400p analog card with 3 fxo and 1 fxs.

I want to connect and configure an analog phone to fxs port and make call to and from sip phones..

I add a zap extension but I can not get dial tone.. should I configure a trunk before?

Here is some information:

Quote:
$lsdahdi

### Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)
1 FXO FXSKS (SWEC: MG2) RED
2 FXO FXSKS (SWEC: MG2) RED
3 FXO FXSKS (SWEC: MG2) RED

Forums: 

No metrics for Network under System Statistics

tislam's picture

I noticed that no network activity such as eth0 receieve, eth0 transmit has been logged under the System activities, after my initial install. I believe, they stopped working after I performed the very first update. Could anybody please tell me what killed that configuration and how to correct it? Thanks.

Sincerely
Towhid Islam

Forums: 

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