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

[Alec Davis]

Previously incorrectly sent to asterisk-dev list, sorry.

I tried today while connected to a Jtec QSIG E1 card, with
DAHDISendCallreroutingFacility with the following test dialplan:

Extension 4888 is on the Fujitsu 

[incoming]
exten => 8688,1,Answer()
exten => 8688,n,Playback(connecting)
exten => 8688,n,DAHDISendCallreroutingFacility(4888,8688)
exten => 8688,n,Playback(goodbye) 

With the following in chan_dahdi.conf
...
context=incoming
facilityenable=yes
transfer=yes
switchtype=qsig
signalling=pri_cpe
channel => 1-15,17-31

Is this how DAHDISendCallreroutingFacility is expected to be setup?

After dialing into the E1 and hearing 'connecting' the result was an
immediate hangup as the transfer was started, the only a warning regarding
'reason' and defaulting to unknown.

The Facilty messaqge sent to the Jtec was 86 bytes long, is there a way to
construct a minimal facilty message, as the Jtec debug, although I don't
have it tonight, reported an error with one of the 'message types' in the
facility message.

I have the option of swapping out the QSIG card in the Jtec for a non QSIG
card, and change to switchtype=euroisdn in chan_dahdi.conf. Would
DAHDISendCallreroutingFacility then do the equivalent ETSI methods to
reroute the call?

I may be able to test this over the weekend, in the mean time, I thought I'd
ask, if this was the correct way, or if mattf, rmudgett or others had 'team'
branch that is a work in progress that we can perhaps have a look at.

Alec Davis

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Saúl Ibarra
Sent: Thursday, 19 November 2009 12:28 a.m.
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

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev