asterisk users - Jun 2008 - QSIG transfer of calls away from Asterisk?

I have a question concerning an Asterisk box connected to another
(non-Asterisk) PBX using QSIG signalling over an E1 connection.

Certain extension numbers of the external PBX are routed to the
Asterisk box, which in turn routes calls to its own extension phones.
Dialplan routing is also in place so that users of those phones can
also make calls routed over the E1 to extensions of the external PBX.

I think that if an incoming call (PBX->Asterisk) is then transferred
to an external extension, the call continues to pass through the
Asterisk box, consuming two B-channels on the E1.

Is it possible (a) in the QSIG protocol and (b) in the Asterisk
implementation of it, to arrange it so that a transfer such as
described above causes the call to revert back to being handled
completely by the external PBX, and release the two B-channels?

Cheers
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org

Hi!
> Is it possible (a) in the QSIG protocol and (b) in the Asterisk
> implementation of it, to arrange it so that a transfer such as
> described above causes the call to revert back to being handled
> completely by the external PBX, and release the two B-channels?
The QSIG keywords that you are looking for a "call diversion" and most
probably also "path replacement". 

In the Q.931 world there is "call diversion" for Point-to-Multipoint
and
"call reroute" for Point-to-Point connections. The bristuff patches
come
with call diversion, while chan_capi support has both (but that's 
depending on the ISDN card you are using); chan_sirrix also supports both 
methods, as well as basic Q.SIG features. 

In how far any of these - more or less fully - implement QSIG as well I 
don't know, however it appears that the Sirrix people would be ready to 
help if you pay them... ;-) And for what concerns chan_capi you might 
want to take a look at this:

http://www.melware.org/ChanCapiQsig

Cheers, Philipp

Tony Mountifield wrote:
> I have a question concerning an Asterisk box connected to another
> (non-Asterisk) PBX using QSIG signalling over an E1 connection.
> 
> Certain extension numbers of the external PBX are routed to the
> Asterisk box, which in turn routes calls to its own extension phones.
> Dialplan routing is also in place so that users of those phones can
> also make calls routed over the E1 to extensions of the external PBX.
> 
> I think that if an incoming call (PBX->Asterisk) is then transferred
> to an external extension, the call continues to pass through the
> Asterisk box, consuming two B-channels on the E1.
> 
> Is it possible (a) in the QSIG protocol and (b) in the Asterisk
> implementation of it, to arrange it so that a transfer such as
> described above causes the call to revert back to being handled
> completely by the external PBX, and release the two B-channels?
a.) Yes, it is possible in the protocol.

b.) With the most recent version of libpri there is support for this.  I 
have heard mixed success with it, so your mileage may vary.  You simply 
need to have both channels be present on the same PRI, your switchtype 
must be Q.SIG, you must have transfer=yes for both channels involved (in 
zapata.conf) and you must also not be using any Dial() command flags 
that require monitoring of media (HhTt, and a few others) since it is 
accomplished via the native bridging code.

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.