[Asterisk-Users] T1 passthrough/middleman

[Mimmus]

In zapata.conf I have, among other things: 

; Incoming only
group=0   ; Zap/g0
signalling=pri_cpe
context=from-pstn
channel => 1-10
; Outgoing (only?)
group=1   ; Zap/g1
channel => 11-15,17-21
; To/From Alcatel
group=2   ; Zap/g2
signalling=pri_net
context=from-alcatel
channel => 32-46,48-62

Then in extensions.conf I have:

[from-pstn]
include => ext-did
include => from-pstn-timecheck  ; this has to be included otherwise it
overrides ext-did

[ext-did]
; My DID has 3 numbers (scrambled to protect innocents): 987654ZXX
exten => _987654ZXX,1,Set(FROM_DID=_984899ZXX)
exten => _987654ZXX,2,Set(NumberCalled=${EXTEN:6})
exten => _987654ZXX,3,Goto(custom-ext-did,${EXTEN:6},1)

[custom-ext-did]
; use only 'include' here!!!
include => ext-local
; change trunk number below if trunks order changes!!!
include => outrt-003-alcatel

[from-pstn-timecheck]
...
 (if incoming call doesn't match DID then do whatever you like...)
...

[outrt-003-alcatel]
; trunk '3' is Zap/g2 (To/From Alcatel)
exten => _ZXX,1,Macro(dialout-trunk,3,${EXTEN},)
exten => _ZXX,2,Macro(outisbusy)        ; No available circuits

[from-alcatel]
; allow Alcatel phones to call Asterisk extensions
include => ext-local
; allow Alcatel phones to call PSTN numbers
include => from-alcatel-ext
exten => s,1,Macro(hangupcall)
exten => h,1,Macro(hangupcall)

[from-alcatel-ext]
exten => _X.,1,SetTransferCapability(SPEECH)
; trunk '2' is Zap/g1 (outgoing)
exten => _X.,n,Macro(dialout-trunk,2,${EXTEN},)
exten => _X.,n,Macro(outisbusy)

That's all (more or less)
Please send me a beer if it works for you!

Bye from Italy

M.


> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Nathan Bell
> Sent: Monday, June 12, 2006 6:41 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] T1 passthrough/middleman
> 
> That sounds exactly like what I want to do. I've don't have a 
> PRI line (although I'm going to press for getting one soon), 
> but for now I would just like a couple of pointers in getting 
> Asterisk's dial plan set up to just pass the calls from one 
> T1 to another.
> 
> Thanks a million in advance.
> 
> Mimmus wrote:
> 
> >I used this approach to gradually migrate from a legacy Alcatel PBX:
> > PSTN E1 PRI --- Asterisk --- Crossed E1 cable --- Alcatel PBX At 
> >first, Asterisk did nothing, only passing calls to/from Alcatel.
> >Then I started to use a bunch of SIP phones directly 
> connected to Asterisk.
> >Now I have the great part of extensions as SIP phones and 
> the old PBX 
> >is working as a channel bank only for a few of analog devices.
> >
> >Configuring the dialplan to do this dirty job is not 
> difficult but now 
> >I'm not able to help you because it's saturday evening and 
> I'm at home!
> >Re-try next Monday.
> >
> >DV
> >
> >
> >  
> >
> >>-----Original Message-----
> >>From: asterisk-users-bounces at lists.digium.com
> >>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf 
> Of Nathan 
> >>Bell
> >>Sent: Friday, June 09, 2006 10:34 PM
> >>To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>Subject: [Asterisk-Users] T1 passthrough/middleman
> >>
> >>Is it possible to act as a middle man on a T1 line?
> >>
> >>My installation currently has an aging Inter-Tel Axxess box 
> with a T1 
> >>coming in (16 in, 8 out). Rather than adding and replacing 
> phones and 
> >>cards as they die, I would like to slowly migrate to a asterisk SIP 
> >>installation.
> >>
> >>I want to take the incoming T1 line, use any available 
> outgoing lines 
> >>for outgoing SIP, intercept any incoming lines and either send them 
> >>off to a SIP line or pass them through to other T1 line 
> (going to the 
> >>Axxess box), and finally take in outgoing calls from the 
> Inter-Tel box 
> >>and either send them to SIP or send them to the outside T1 line.
> >>
> >>How will a dual T1 card be set up in this situation? Would it be 
> >>easier to use an FXO channel bank (or card) and connect 
> analog lines 
> >>to the FXS analog lines on the Inter-Tel box?
> >>_______________________________________________
> >>--Bandwidth and Colocation provided by Easynews.com --
> >>
> >>Asterisk-Users mailing list
> >>To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>    
> >>
> >
> >_______________________________________________
> >--Bandwidth and Colocation provided by Easynews.com --
> >
> >Asterisk-Users mailing list
> >To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >  
> >
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>