[asterisk-users] evaluate SIP response codes in dialplan

[Philipp Kempgen]

Johansson Olle E schrieb:
> 19 jan 2009 kl. 11.10 skrev Philipp Kempgen:
>> Johansson Olle E schrieb:

>>>> I still think we need a SIP_CAUSE channel variable. :-)
>>>>
>>> Then we need to start working on aggregation rules, like what if one
>>> IAX channel answers and one SIP channel is busy?
>>>
>>> For SIP-only calls, we need to add a lot of code from proxy rules for
>>> call forking and response aggregation. It's not an
>>> easy task.
>>
>> I know it's not an easy task if you'd want it to be done properly.
>> But then again Asterisk is not a SIP softswitch but a PBX.  :-)
>> I've never seen people who are asking for SIP_CAUSE expect it
>> to work under all circumstances. All the use cases are pretty
>> simple:
> Well, but if we implement a half-done implementation, we will get a
> ton of bug reports within days... We can't do it like that, Philipp.

I guess you're right.
Give them an inch and they will request a mile.
SIP_CAUSE_HALFBAKED could do the trick. ;-)

> (Well, looking at TLS/TCP in 1.6 I guess we can do anything... ;-) )

;-)


   Philipp Kempgen

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