sip_additional not being written

I’ve searched and found a similar entry in the trak:

SIP changes made in the gui (ie, add or delete an extention) are not happening in aterisk.

If I run /var/lib/asterisk/bin/ and then a ‘sip reload’ in the asterisk CLI, then it works fine.

This is an ‘unsupported’ configuration I’m sure, but any help or pointers would be appreciated. As always, if I find the answer first, I’ll post it back up here.

FreePBX 2.3.0beta 1.9 (framework 2.3.0beta 1.5)
Debian 4.0 (2.6.18-4-686 #1 SMP i686 GNU/Linux)

Traced to a problem with the paging module, error was being given about a sip header (sorry, didnt write it down, gone from scroll buffer now) so I removed the paging module and restarted asterisk.

That seemed to do it. The bizarre thing is I’ve reinstalled the paging module and the error did not return.

While typing this I realised that I’d not added any paging groups so I went ahead and did so and now the problem is back. Changes made in the gui are not being written back to asterisk.

Here’s the output of /var/lib/asterisk/bin/retrieve_conf

[code]Checking for PEAR DB…OK
Checking for PEAR ConsoleGetopt…OK
Checking for /etc/amportal.conf…OK
Bootstrapping /etc/amportal.conf…OK
Parsing /etc/amportal.conf…OK
Parsing /etc/asterisk/asterisk.conf…OK
Connecting to database…OK
Connecting to Asterisk manager interface…OK

Fatal error Class ‘ext_sipaddheader’ not found in /var/www/html/admin/modules/paging/ on line 91[/code]

Looks like a bug?

please post these on the beta list - they are more likely to be noticed and an answer provided to you:

Yeah, I thought of that as I clicked submit, too late I know.

The thing is, I dont know if it’s the beta of freepbx or the module thats causing the problem? The changelog for the module does mention something about this sip header.

Anyway, if you can move the thread to a more suitable place that’d be great.
Cheers Philipe

it’s the paging module. The short term fix would be to disable paging, hopefully I’ll have the beta out today or tomorrow at the latest, or you can just grab the file I mentioned in the other thread.

Is this related? Or is there something else to do

Just made a backup of my production box, copied the backup over to a new box. Everything looked OK, but phones don’t seem to register properly.

They can make outgoing calls, but incoming calls go straight to VM.

Looking at the DATABASE, I do not see any entry for (at least) SIP and Device. I’m able to recreate an extension from scratch. And it works OK

If I go into any extension setup page, I can save and realod an existing extension and the extension then seems to work…

Is there a command that can rebuild the realtime DATABASE without having to touch every extension?

there seems to be something broken in the backup restore stuff although I was sure I had it fixed several months ago. The backup tarball has a filed called something like astdb.dump or something similar and there is a function in the AMPBIN directory calls restoreastdb.php that can restore it. The problem is, it was written really badly and expects the file to be in a very specific place that the backup stuff sets up (if you look in the restore script you will see). That script could easily restore all your settings (as it should) if you can get things properly setup, or modify the script so you can just pass in the name of the dump file which you can get out of the backup tarball.

backup is on the list of problematic modules to review again - problem is, it really needs a rewrite…

Probably the easiest thing to do at this point is to Just Copy the file from one box to the other!