[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
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