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. > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
