Hangupcause SIP implementation

Hi All,

I am very new to freepbx and Asterisk so please excuse my lack of knowledge at this stage.

I have setup a very basic installation of freepbx 1.4 with Asterisk version 1.6 32bit.

I have setup a SIP trunk with my ISP and can make inbound and outbound calls without an issue.

I would like to know how to implement hangupcause for SIP.

The problem I find is that if a user dials a number that is non existent the SIP provider gives us a SIP/2.0 404 Not found which in turn triggers a macro that classifies this code as being congestion and an all circuits are busy message is played to the user.

I would like to change this message to something more relevant either a disconnected message or dial tone.

I know we can replace the “all Circuits are busy” GSM audio file but this is very generic and dirty way of fixing the issue and not all codes are matched up.

I have seen another post (http://www.freepbx.org/forum/freepbx/users/potential-incorrect-handling-of-sip-404) with a patch for this particular issue but I am at a loss how to implement the patch or where the code is meant to go.

Any help would be greatly appreciated.

If you require me to post and config files please let me know.

Thanks in advance for your help.


I’m unsure if nicson ever submitted the patch. Interesting timing, I just came back to visit the forums today to see if there was any progress on the matter and chanced across this post :smiley:

Unfortunately I’m no programmer / developer so I’m not entirely sure how to implement it either, but I’ll keep an eye on this thread myself to see how you go :smiley: