[Asterisk-Dev] 1.2 Feature freeze.

[Jerris, Michael MI]

> Harald Milz
> 
> Jerris, Michael MI <mjerris at ofllc.com> wrote:
> 
> > There has been talk of 1.2 for quite some weeks or months 
> now both on 
> > the dev list and on the dev conference calls.  If somone 
> has not been
> 
> Please don't omit the externhost feature. It's a must for 
> many people running an Asterisk behind a DSL NAT router with 
> a dynamic IP, as it is usual in Germany.
> 
> And while we're at it, I still see no official way to 
> determine the called number on incoming calls. Example:
> 
> - Sipgate gives you only a single DID number, e.g. 01234-567891
> - if someone dials 01234-567891-12, 01234-567891 will be 
> called but the
>   whole number will be sent in the SIP header including the 
> suffixed -12
>   (a SER feature as it seems). This allows me to route incoming calls
>   directly to a specific extension of my attached ISDN PBX (which in
>   turn is just an 8-way analog adapter if you will) depending on the
>   suffix. It's like an ISDN PtMP setup.
> - I see no official way to see the affixed -12 in an extension. I
>   submitted a patch for CALLEDNUM but there has been little 
> interest so
>   far, although this is a _very_ interesting feature. This 
> patch simply
>   determines the called number from the SIP header and 
> creates a variable
>   CALLEDNUM that can be used in the dial plan.
> 
> It would be very useful to include this feature - or document 
> how that can be done the "official" way. The "s" extension 
> doesn't give you access to the number that was actually 
> dialed, and if I use an explicit extension in the "register" 
> line in sip.conf, only this extension will be available.
> Effectively, you only get a clue _that_ you were called, not 
> _how_ you were called. I never see the actually called number 
> including the suffix. Any idea???
> 

This thread was not really meant as a feature request list for 1.2.  If
you would like specific features, please post a patch to mantis.  As for
your sipgate multiple trunk matching issue, it is a known difficulty
with asterisk, and we welcome patches to clean up the way that works.
There are workarounds for it in some detail in mantis if you search
recent bugs for "broadvoice".  To my knowledge, no functionality has
been "removed" for the upcoming 1.2 but there has most certainly been
significant changes, you milage may vary, you will need to look at cvs
head for your answers on specifics.  As for your patch, drum up
interest, and get testing activity.  

Mike