[asterisk-users] evaluate SIP response codes in dialplan

[Johansson Olle E]

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