[asterisk-dev] Can asterisk PRI/BRI support redirect calls

Don Kelly dk at donkelly.biz
Thu Nov 19 07:08:27 CST 2009


It sounds to me like you need to use a straightforward Two B-Channel
Transfer (TBCT), an ISDN feature. I haven't used it in Asterisk, but
discussion on the users list a while back indicates it was implemented (not
sure what version) and will work when you do a simple transfer if you have
your configuration set correctly and the switch handling your PRI is set to
handle it.

You answer the incoming call, determine what the caller wants via the auto
attendant (invite them to wait a moment if required), make an outgoing call
on another channel to the appropriate destination number (make an
announcement if required) then complete the transfer--Asterisk sends a
facility message to the ISDN switch requesting the transfer. The switch
completes the transfer and disconnects both channels on the PRI to the
Asterisk system.

Questions like this concerning the operation of your Asterisk implementation
would be better sent to the users' list: asterisk-users at lists.digium.com

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax


-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Saúl Ibarra
Sent: Wednesday, November 18, 2009 5:28 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Can asterisk PRI/BRI support redirect calls

Hi Alec,

On Wed, Nov 18, 2009 at 10:42 AM, Alec Davis <sivad.a at paradise.net.nz>
wrote:
> We have asterisk for a small group of users in our head office, and still
a
> Fujtisu PBX for the majority of users.
>
> The request have been can we get asterisk to be the Automated Attendant
for
> incoming calls from the PSTN.
> IE. Press
>    1 for sales
>    2 for service
>    3 for admin
>    ...
>    0 for reception
>
> The answer so far is, of course asterisk can. But as I understand it will
> bridge the call to the Fujitsu PABX.
>
> We need to transfer the call back out of asterisk down the E1 line and to
> the MAIN PABX, and free up the 2 trunks used. As I understand this,
redirect
> the call.
>
> Tthe setup is as below.
>
> SALES       SERVICE/ADMIN/...
> ASTERISK    FJPABX
>   |        |
>   E1       E1
>   |        |
>   ISDN SWITCH (Jtec 5015 - Nice but obsolete)
>        |
>        E1
>        |
>       TELCO
>       PSTN
>
> Developers: Guide me in the right direction, and if it's not supported,
> what's the likely hood? Or do I need rmudgett on the case.
>

If I understood correctly what you need is called "Call path
replacement" which is not currently supported in Asterisk. However, I
contacted Dialogic as their Diva cards seemed to support this
(according to their website).

For that you need chan_dialogicdiva, which at the time I checked it
did support call path replacement but NOT in NT mode. You may ask
again if support has been added.


Regards,


-- 
/Saúl
http://www.saghul.net | http://www.sipdoc.net

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