First of, let me start by saying this is my First PBX install and the first time i have worked with a linux based system in over 12 years. I was hoping this would have been very simple, but i think i am now going grey because of this…
My Voip provider is sipgate, I have been able to install free PBX using the newest distro version. Currently i have been able to setup connection between sipgate and the pbx server. But i am having trouble connecting phones to the network.
The first phone i am trying to connect is a 3CX softphone. I have created a extension but i have no idea how to link that extension to the phone, The phone just does not appear anywhere within the system! Any ideas where i should look?
Also the webserver keeps going offline! really starting to tick me off as well, how important is the web server to the general running of software? if this is important and unstable then i need to fix this first!
I use the 3CX softphone and it does work, but it doesn’t just ‘show up’. You have to configure the softphone application with the extension, server and password (at least) in order for it to connect. As for the webserver…it is important to be able to configure the system, but Asterisk runs without it. I noticed that the latest version of FreePBX sometimes seems to flash yellow on the webserver icon as I switch between the FreePBX pages, particularly when making several changes. Don’t know if that is your issue or not. On what hardware is it running? Could it just be slow response?
The network connection is stable, an your correct about it changing colour, only seems to happen working on the gui.
I have setup the 3CX softphone in the profile and settings section, it reports back as not connected. i also think its saying authentication failed just before the not connected sign, For the credentials i have entered the correct extension number and password but i am a bit unsure what the ID is meant to be.
Where do i set the ID or view ID for each extension?
THe ID is the extension number and the password is the secret. Often you need to enter the ID twice, as an auth ID (for authentication) and the actual SIP ID.
Well i am happy to report the inbound calls now go to “ring group” and they all ring and work perfectly well.
My next problem is making out bound calls, I have created an outbound route but it is currently blocking calls. I have done the following so far,
Route Name: Name
Route CID: "Blank"
Route Password: “Blank”
Do i need to my VoIP details here? (I though they was all ready connected?
Maybe its the dial patterns. I have done the following,
Prepend “Blank” + “0” Match Pattern: “Blank” Caller ID: “Blank” I have added all the local start numbers, EG 0, 07, 08, 02, 01. Maybe i have entered these details in the incorrect location?
Thanks for all your help so far guys! I was close with most of the settings but close does not make it work. But i am learning and starting to get comfortable with it.