[asterisk-users] evaluate SIP response codes in dialplan

[Marcus Hunger]

Hi,
sorry for joining the discussion so lately. I'd like to ask you to check
http://bugs.digium.com/view.php?id=14810. The patch tries to address the
issue using channel-variables to propagate the hangup-cause to the calling
channel.

Best regards, Marcus

On Fri, Jan 23, 2009 at 3:08 PM, Johansson Olle E <oej at edvina.net> wrote:

>
> 21 jan 2009 kl. 11.49 skrev Klaus Darilion:
>
> > Hi Olle!
> >
> > Currently we have the problem that due to
> > SIP<->hangupcause<->SIP<->hangupcause.... conversions the original
> > hangupcause gets lost in a chain of Asterisk servers using SIP.
> >
> > In chan_sip there is already code for adding the X-Asterisk-Hangupcode
> > header. What about reading this header on the receiving side for
> > setting
> > the hangupcause instead of doing SIP->hangupcause mapping ?
> In this case we could do that, but there has to be an option to enable
> it
> since it will change the behaviour in existing networks.
>
> Good idea!
> /O
>
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-- 
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Marcus Hunger - hunger at sipgate.de
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