Hello,
I installed AsteriskNow this evening without a hitch and was able to login to FreePBX without any problems. The problems started when I tried to make my IP phones work.
I have a number of Grandstream GXP-2020 handsets but between the web config utiliy for the phone and FreePBX just could not do anything to get the phone to register. On selecting an account on the handset the screen just said that it wasnt registered and so no calls could be made.
Could somebody please confirm exactly what has to be configured to make a phone register out of the box. I assume that no files have to be manually edited as this defeats the object of using FreePBX?
I wasted hours on this so any help would be great,
Cheers,
Joel.
Phone Provisioning is something that FreePBX does not do, I don’t remember if AsteriskNow has any provisioning for Grandstream phones. Assuming not then yes you will need to edit the files that the phone looks for on boot (or enter all the needed information correctly via the phones interface). Items that each phone will need are: the user Id (Normally the extension), the password which is the secrest you defined for the extension number, the Server IP/name for SIP proxy, and registrar. You also probably want to enter the users display name and/or screen name.
Due to the number of phone manufactures and large quantity of phone models produced per manufacture FreePBX just does not provide this feature. Many Distros will provide provisioning interfaces for some manufactures and not others.
Okay, so I understand that the IP phone needs to have some information which corresponds to the extension setup via FreePBX.
I discovered a tutorial for the GXP-2020 and Asterisk here - http://www.asteriskguru.com/tutorials/gxp2020.html but im a little confused at a disgrepency between the sip.conf file in the tutorial and the sip.conf file generated via FreePBX on my particualr machine.
Basically, when i create a generic SIP extension with FreePBX, if i then open sip.conf I can see that the extension has been created with the exception of the line “username=xxxxxxxx” which isn’t created. Do I need to manually add this line for each extension created in FreePBX in order for the phone to work?
Out of the fields to ce completed on the phone then i would imagine the 3 required ones are
"SIP Server " - here you must type the address of the Asterisk server you want to use.
“Authenticate ID” - this is the ID which your user uses to login.
“Authenticate Password” - this is the password which your user uses to login.
So in the sip.conf file under the extension, Username= should be the same as Authenticate ID on the phone setup but i just cant undertand why this line is missing.
Any clues?
Cheers.
ok that tutorial is for configuring the phone if you are only using asterisk, which you are not and if you open the sip.conf and extensions.conf files you will see that it tells you to not edit them as they are now owned by FreePBX.
The authentication ID should be the extension number.