[Asterisk-Users] SIP Responsecodes

[Olle E Johansson]

3 apr 2006 kl. 18.46 skrev Roger Schreiter:

> Douglas Garstang schrieb:
>> Wow. If Asterisk could return SIP response codes that would be  
>> AWESOME.
>
> ... and the remote IP address (which may differ from the
> address who registered).

...is available in the dialplan functions SIPCHANNEL and SIP_PEER

>
> Btw: Isn't the SIP response translated into a Q.931 code,
> which can be read by ${HANGUPCAUSE}?
We have adopted those codes as Asterisk generic cause codes.

We are trying to stick with that, as implementing dialplan routing
based on SIP response codes would not make any sense on your
PRI trunk or IAX2 channels. Asterisk is a multiprotocol PBX, and
the solutions we are implementing are always geared towards
multiprotocol solutions - if possible.

We are improving the cause code implementation in chan_sip
so that you will have the cause code available. If you find cases
where it's not availble, please alert me and we'll try to fix it.

Thanks,
/Olle

>
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* Olle E. Johansson - oej at edvina.net
* Asterisk european tour: http://www.meetasterisk.com