[Asterisk-Users] PRI in and out

[altus]

I got the same setup,sort of
I connected a single port sangoma to my pbx
My ony problem is,when a call comes in and it gets transfered back out
that it does not detect the hangup?So that channel keeps being open
Any ideas why


On Wed, 2005-09-07 at 01:40 -0600, Rich Adamson wrote:
> > I am wanting to front-end a legacy PBX with an asterisk box. I have done plenty 
> > of asterisk work over the last 6 months to PRI circuits, but not with a PBX 
> > being involved.
> > 
> > I know I can use asterisk and digium cards in this manner, but do I need 
> > separate cards for the PRI -> Asterisk side to the Asterisk -> PBX side, or will 
> > a 4-port PRI card do the job? (I already have a spare one of these).
> 
> The 4-port card will work just fine.
> 
> > In other words, can I use SPAN 1 as a timing source, then provide timing to the 
> > PBX connected to SPAN 2 of the same card?
> 
> Yes. In fact, the 4-port card will be a slight advantage over two 
> single port cards as all ports on the 4-port card will have their
> clocks in sync with your external timing source.
> 
> Keep in mind that all T1/E1 spans having timing embedded in their
> transmit legs; you can't turn that off even if you tried. The clock
> timing source is always an engineering decision as to chosing which
> "receive leg" to use for clock sync. (Obviously, the span from the
> pstn would be your timing source and not the span to the pbx. If
> you already are using the PRI with the PBX, then no changes required
> on the PBX side for clock sync.)
> 
> The config examples in zapata.conf and the wiki are good. Not much
> to configure really.
> 
> You will probably want to focus more on options that your pstn 
> provider can/will impact such as the number of digits to be sent 
> from them to you, which channel is the d channel, the digits they 
> expect from you for each call (whether prefixed with "1", "0" or 
> whatever), etc.
> 
> As sort of a side note, the 4-port card gives you another slight
> advantage from an ongoing support perspective. The third (or forth)
> port could be connected to a "test" asterisk box on which you can
> stage/test future asterisk code before moving it into the production
> box. Think about reserving a couple of DID numbers for the test
> box if you'll be using DID.
> 
> 
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Thanks
Altus Snyman
Stormcorp Network Solutions
+27 11 8071141 exten 301