[Asterisk-Dev] Re: [Asterisk-Users] Call completion/error codes and extensions.conf call flow

John Todd jtodd at loligo.com
Tue Apr 13 20:53:33 MST 2004


At 11:06 PM -0500 on 4/6/03, Tilghman Lesher wrote:
>On Sunday 06 April 2003 14:09, John Todd wrote:
>>  There was a conversation last night on the IRC channel between
>>  myself, Corydon76, citats, and kram on the ability of a call
>>  process to access the error (or success?) codes underlying a
>>  call.  I'm uncertain if anything came out of it, but I'll
>>  re-hash here to solicit other comments.
>>
>>  My idea: I'd like to be able to get to error codes when a call
>>  passes through some kind of action (with perhaps the exception
>>  of "Trying" or other "non-end-of-message" results) so that I
>>  can play error messages or take actions that are appropriate
>>  to the event.  As an example, currently Asterisk only supports
>>  "busy" or "unavailable" call codes back from any channel type.
>>   However, chan_sip can provide a large array of codes that are
>>  more meaningful, such as "403 Forbidden" or "480 Temporarily
>>  unavailable" which can be more useful for both my internal
>>  logging as well as can trigger an appropriate recording to be
>>  played back to the user.  Why code individual cases inside of
>>  chan_blah when this can be extracted to allow the admin to
>>  handle them as required?
>>
>>  Two solutions were discussed, one method using an application
>>  and the other simply setting a channel variable.
>>
>>  METHOD 1:
>>  Corydon76 suggested creating an application that handled the
>>  redirection of the call process flow.  This would look
>>  something like this:
>>
>>  OnResultGoto(201:+100,405:+200,480:+250)
>>
>>  where "201", "405", and "480" are the numeric response codes
>>  corresponding to some useful error message from the channel.
>>  This would be called immediately the Dial application.  Jumps
>>  would be done within the existing context, to the priority X
>>  within the same context where X is represented by "+X".
>
>Here's the application and the patch needed in the pbx code to
>support the result code accessed from the application.  In
>addition to the relative branches mentioned in the channel, I
>also made absolute branches work (in the usual [[con]|ext]|pri
>format).
>
>Again, untested code.
>
>-Tilghman
>

[programs included in 
http://lists.digium.com/pipermail/asterisk-users/2003-April/009816.html 
]

Did anything ever happen with these (IMHO) very useful little 
patches/programs?  If not, you may consider submitting to the 
bugtracker.  I haven't heard of any new discussion about processing 
result codes in some sane way; this seems to be the best so far.  I 
am ashamed to say I haven't tested these, but came across them in 
reviewing my inbox.

Crap, that was a YEAR ago!?!?  That's simply not possible.

JT




More information about the asterisk-dev mailing list