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

[Richard Mudgett]

The facility message sent by DAHDISendCallreroutingFacility() is intended to be used before the call is answered as the call is being deflected.  The Jtec is probably complaining about the CallRerouting facility message being sent after the call was answered.

I have seen in the code for the application that it will always return nonzero.  Thus, it will always terminate dialplan execution and hangup the call after sending the facility message.  I don't think that this is much of a problem since if the call is deflected it will be hung-up anyway.  If the deflection fails however, the call will just be dropped because the application returns non-zero.

DAHDISendCallreroutingFacility() in released code supports Q.SIG only.  I have recently extended it to work for ETSI(EuroIsdn) PTP and PTMP.  The extension is only available on asterisk svn trunk and libpri svn 1.4 branch code.

Richard
----- Original Message -----
From: "Alec Davis" <sivad.a at paradise.net.nz>
To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
Sent: Thursday, November 19, 2009 2:04:41 AM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-dev] Can asterisk PRI/BRI support redirect calls

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

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