[asterisk-users] evaluate SIP response codes in dialplan

[Johansson Olle E]

15 jan 2009 kl. 13.02 skrev John covici:

> That is very nice, but where are the HANGUPCAUSE values documented?
That's the issue...

include/asterisk/causes.h is a good reference for now.

/O
>
>
> Thanks.
>
> on Thursday 01/15/2009 Johansson Olle E(oej at edvina.net) wrote
>>
>> 14 jan 2009 kl. 14.02 skrev Klaus Darilion:
>>
>>> Hi!
>>>
>>> Is it somehow possible to evaluate the SIP response code inside the
>>> dialplan?
>>>
>>> I have an Asterisk server which forwards requests to various PSTN
>>> gateways with SIP. If the Dial() attempt is not successful I want to
>>> differ at least these 3 options:
>>> - called destination is busy (486): e.g. activate auto-redial
>>> - called destination does not exist, unassigned number (404)
>>> - gateway is broken, error, circuit busy (e.g. 503)
>>>
>>> 486 is mapped to DIALSTATUS=BUSY
>>> but both 503 and 404 is mapped to DIALSTATUS=CONGESTION
>>>
>>> As when Asterisk forwards the response with SIP to the caller the  
>>> same
>>> response code is used, I suspect this information must be stored
>>> somewhere inside the channel variable. So, are there any means to
>>> access it?
>>
>> Check the HANGUPCAUSE, it's much more detailed than DIALSTATUS.
>>
>> We do map the SIP (and all other protocol errors in various channel
>> drivers) codes to ISDN hangup causes, which gives you much more
>> information about
>> why a call failed.
>>
>> The conversion we're using follows the RFC, and where that doesn't
>> cover it, Cisco's documentation.
>>
>> /Olle
>>
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>
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
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* Olle E Johansson - oej at edvina.net
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