Internal calls (where call pickup comes into play) is handled by the
proxy. It's always beena  goal of the project to intentionally not use
a b2bua for every phone connection in order to achieve peer to peer
media and scalability.

I would really not put FS in that role, it's intention is a media server.
On Fri, Sep 16, 2011 at 3:09 PM, Niek Vlessert <[email protected]> wrote:
> Hello all,
> Some of you might know that Call Pickup and BLF with Aastra phones don't
> work on SipX because the Aastra firmware is not compatible.
> We already fixed the BLF issue
> (http://track.sipfoundry.org/browse/XTRN-113?focusedCommentId=55875#action_55875)
> but now we need Call Pickup.
> The problem is that the phone won't respond well to the call pickup SIP
> stuff. Is there a way to get control over the call in another way?
> Something like this; instead of dialing *78<extension> (which is call
> pickup) we dial *79<extension>. In Sipx, add a gateway to local port 15060,
> which is freeswitch, and a route to get the *79<extension> in Freeswitch.
> Freeswitch can execute any script. Is there a way to get to the SIP header
> and bridge the call to the phone which did the *79? I know, not beautiful at
> all, but it's a way…
> Some other direction we are thinking is that Freeswitch registers itself as
> a phone on the same extension as the Aastra phone, the dual forking feature
> in SipX. So if the number is dialed, the Freeswitch phone will also 'ring'.
> Maybe we can then bridge that call to the user who did the *79<extension>?
> But it we do that, then every Aastra phone needs a seperate SIP account in
> Freeswitch. Freeswitch can handle that, but that's even less beautiful, I'd
> say very ugly. ;)
> Anyone got a trick?
> Regards,
> Niek Vlessert
> Telecats
> _______________________________________________
> sipx-dev mailing list
> [email protected]
> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>



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