[Asterisk-Dev] 1.2 Feature freeze.

[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???

TIA!

Ciao,
hm

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