Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] QSIG transfer of calls away from Asterisk?


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
tony at softins.clara....
Guest





PostPosted: Mon Jun 02, 2008 11:10 am    Post subject: [asterisk-users] QSIG transfer of calls away from Asterisk? Reply with quote

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
Back to top
klitzing at pool.infor...
Guest





PostPosted: Mon Jun 02, 2008 12:31 pm    Post subject: [asterisk-users] QSIG transfer of calls away from Asterisk? Reply with quote

Hi!

Quote:
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... Wink And for what concerns chan_capi you might
want to take a look at this:

http://www.melware.org/ChanCapiQsig

Cheers, Philipp
Back to top
creslin at digium.com
Guest





PostPosted: Mon Jun 02, 2008 1:38 pm    Post subject: [asterisk-users] QSIG transfer of calls away from Asterisk? Reply with quote

Tony Mountifield wrote:
Quote:
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.
Back to top
tony at softins.clara....
Guest





PostPosted: Mon Jun 02, 2008 1:58 pm    Post subject: [asterisk-users] QSIG transfer of calls away from Asterisk? Reply with quote

In article <48443E0C.1020606 at digium.com>,
Matthew Fredrickson <creslin at digium.com> wrote:
Quote:
Tony Mountifield wrote:
Quote:
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.

Thanks Matt, it's good to know there is something worth trying.

Since you said "most recent", would it be correct to infer that this is
only possible in 1.6, not 1.4?

Cheers
Tony
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org
Back to top
creslin at digium.com
Guest





PostPosted: Mon Jun 02, 2008 3:49 pm    Post subject: [asterisk-users] QSIG transfer of calls away from Asterisk? Reply with quote

Tony Mountifield wrote:
Quote:
In article <48443E0C.1020606 at digium.com>,
Matthew Fredrickson <creslin at digium.com> wrote:
Quote:
Tony Mountifield wrote:
Quote:
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.

Thanks Matt, it's good to know there is something worth trying.

Since you said "most recent", would it be correct to infer that this is
only possible in 1.6, not 1.4?

It is possible in 1.4 version of Asterisk as well as in 1.6. You simply
need to be using at least version 1.4.4 of libpri.

--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services