----- "Klaus Darilion" <[email protected]> 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 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.
--
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
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