[asterisk-users] evaluate SIP response codes in dialplan

[Klaus Darilion]

Joshua Colp schrieb:
> ----- "Klaus Darilion" <klaus.mailinglists at pernau.at> wrote:
> 
>> Philipp Kempgen schrieb:
>>> Klaus Darilion schrieb:
>>>> Is it somehow possible to evaluate the SIP response code inside
>>>> the dialplan?
>>> No. Part of the reasoning is that Asterisk is meant to be a
>>> multi- protocol PBX, not a SIP softswitch.
>> This is IMO a stupid limitation. There are dozens of ISDN cause
>> codes,
>> 
>> dozens of SIP response codes and similar in other protocols, but 
>> Dial() only exports BUSY or CONGESTION ......
>> 
> 
> Right, app_dial condenses down the information it gets into some
> basic string representations. You can also access a more specific
> Q.931 representation by using the ${HANGUPCAUSE} dialplan variable.
> While this is not the SIP response code this gives you more
> information. You can also control the SIP response code by passing a

I see. I thought HANGUPCAUSE works only with zaptel. I will give it a try.

thanks
klaus

> Q.931 value to the Hangup() application itself. Unfortunately the
> mappings of SIP response code <-> Q.931 are hard coded in chan_sip
> though so that is where you can find what maps to what.
>