If you have upgraded from freepbx 2.4 to freepbx 2.5 and now have problems with your inbound routing not working then read on.

If you have callerid inbound routing for some of your calls you may have found that it has stopped working since upgrading to freepbx 2.5.

After some testing this is because the any/any catch all is doing what it should and catching all of the calls. This is no good if you have other inbound routes setup as these will no longer work.

The fix for me was to add the following in the any/any route

in the DID add _.

yep thats it _. and then save

You should now find that all of your inbound call routes work again

written by TMP \\ tags: Asterisk, freepbx, inbound, routing

Calling back a person that left you a voicemail is easy.

When setting up VoiceMail for the extension, be sure to include callback=from-internal in the vm-options field in freePBX.

Then, once you listen to a voicemail message, choose advanced options (3) and then option 2 to return the call.

written by NewsFeeder \\ tags: freepbx, freepbx asterisk voicemail

If you are seeing this error I am getting an error upon running ./install_amp from a fresh svn source.
It’s complaining about a non-existent MySQL column being addressed:
Generating Configurations.conf..
Checking for PEAR DB..OK
Checking for PEAR Console::Getopt..OK
Checking for /etc/amportal.conf..OK
Reading /etc/amportal.conf..OK
Reading /etc/asterisk/asterisk.conf..OK
Connecting to database..OK
SELECT extension, directdid, didalert, faxexten, faxemail, answer, wait, privacy man FROM users WHERE directdid IS NOT NULL AND directdid != ” [nativecode=1054 ** Unknown column ‘faxexten’ in ‘field listGenerating AMP configs..OK
Restarting Flash Operator Panel..OK
Please Reload Asterisk by browsing your server.

Then you need to issue the following from the command line ./install_amp –force-version 2.1.1

written by NewsFeeder \\ tags: Asterisk, freepbx

Put the following in the dial field of the custom extension:   SIP/trunk/number|timeout|rD(digits)   For example:   SIP/telasip-gw/1234567890|30|rD(ww2w110)   “r” is for generating the ring for the calling party and “D(digits)” are the digit to play (DTMF) to the called party before bridging the calls. It can be used to dial an extension. It’s nice because we can add dial options directly in freepbx without having to hack config files. Another example of this is you could create a custom extension 211 and in the “dial” text box you could enter: Local/18005551212@outbound-allroutes If you dial ext 211 then the system will dial 18005551212 as an out bound number. This can be quite handy if you want to dial a mobile number

written by TMP \\ tags: custom extensions, freepbx