No you need to move to FreePBX 2.9. There are also patches to Asterisk we do in the Distro to make it work in all cases as I know in some instances it will not work.
I would suggest your get your vendor to get off 2.8 since that has been EOL and not even getting security fixes that we put in 2.9 and 2.10
Unfortunately for you, Elastix package their own forked up version of FreePBX 2.8, you will have to go to them for help. No-one here will be able to help you.
As Tony said also, you need FreePBX 2.9, this will break Elastix totally, so you are in a Catch-22 situation. I suggest you cut your losses and move to the FreePBX distribution or one that uses the real FreePBX
The patches I was referring to was Asterisk patches not FreePBX. You need to move to FreePBX 2.9. As far as the Asterisk Patch it has already been submitted back to Digium and I believe included in Asterisk 10.
To note the first comment made. If you want to update the dial plan for the hack (true Connected Line is so much better) you can use the custom contexts in extensions_additional.conf. Simply reference them in extensions_custom.conf, that is what they are for, user code.
If you need to rewrite a standard FreePBX macro simple copy the entire extensions into extensions_custom.conf and make your changes. Since extensions_custom.conf is loaded after extensions_additonal.conf (go read the notes in extension.conf and http://www.freepbx.org/configuration_files) any extension in extensions_custom.conf have precedence.
For upgrading to freepbx 2.9 or to wait until elastix incudes 2.9 is the last solution.
But right now i had tried with code Connected Line. plus we need to change sip_additonal.conf also.
In each sip users its showing Callerid="device " in sip_additional.conf. This should be change to callerd id name where we are giving in web gui while creating sip users. then only the things work properly.
As per the previous instruction, we can make extra code in extension_custom.conf for conencted line to work. But for the case of “device” for callerid to name what we can do?
. . . you need FreePBX 2.9, this will break Elastix totally, so you are in a Catch-22 situation. I suggest you cut your losses and move to the FreePBX distribution or one that uses the real FreePBX.
With an aweful lot of work you might be able to add those changes to sip_custom_post (read sip.conf’s penultimate paragraph), but maintenance would be a nightmare in anything but a completely static system.
Yah, i dream about that. Let me try from one to one. I looked into the changes happening while reloading… its totally a night mare as you said. But i have to do that.