Is this an Adit 600?
On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> The output from the CLI when I put in an inbound call is the following:
>
> -- Starting simple switch on 'Zap/25-1'
> -- Executing GotoIf("Zap/25-1", "1?from-pstn-reghours|s|1:") in new stack
> -- Goto (from-pstn-reghours,s,1)
> -- Executing GotoIf("Zap/25-1", "0?from-pstn-reghours-nofax|s|1:2") in new
> stack
> -- Goto (from-pstn-reghours,s,2)
> -- Executing Answer("Zap/25-1", "") in new stack
> -- Executing Wait("Zap/25-1", "1") in new stack
> -- Executing SetVar("Zap/25-1", "intype=EXT-412") in new stack
> -- Executing Cut("Zap/25-1", "intype=intype|-|1") in new stack
>
> It then goes on to call the extension I have setup. I think it's coming in
> on Channel 25, but I'm not sure what the -1 is for in Zap/25-1.
>
> Not sure if this is relevant or not, but I'm using a Carrier Access
> Corporation (CAC) channel bank, with 1 FXO card and 1 FXS card. The analog
> line is definitely hooked to the FXO card, and I definitely have the T1
> plugged in to the FXO card.
>
> Thanks,
>
> James
>
>
> C F wrote:
> > Looks like channel 25 is not the one hooked up to your POTS, when an
> > incoming call arrives, what channel does the CLI report?
> >
> >
> > On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> >> Thanks for the reply. I have tried adding anywhere between 1 and 6 w's to
> >> the dial string, but still no luck. I hooked up and listened on the line
> >> when the call went out, and never heard any DTMF's. I'm sure this must be
> >> something simple, I just can't seem to figure out for the life of me what
> >> it is. What other information can I provide to help sort this out?
> >>
> >> Thanks again,
> >> James
> >>
> >> ------------------------------
> >> You could insert a pause by adding a w before the number to be dialed,
> >> like this:
> >> Dial(zap/25/w1234567890) iirc each w puts a 500ms pause.
> >>
> >>
> >> On 1/30/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> >>>> I am experimenting with an asterisk setup in my office. The last bit I
> >>>> have to test is working with analog lines. I have TE411p digium card,
> >>>> with an ISDN line plugged into the first, a channel bank plugged into
> >>>> the second port, and the last two ports empty. I have the following
> >>>> setup in my zaptel.conf:
> >>>>
> >>>> span=1,1,0,esf,b8zs
> >>>> bchan=1-23
> >>>> dchan=24
> >>>>
> >>>> span=2,0,0,d4,ami
> >>>> fxsks=25
> >>>>
> >>>> And in zapata.conf, I have:
> >>>> group=2
> >>>> language=en
> >>>> context=from-pstn
> >>>> signalling=fxs_ks
> >>>> channel=>25
> >>>>
> >>>> I have one analog line plugged in for testing. If I dial that analog
> >>>> number, the inbound call arrives, and it works great. However, when I
> >>>> place an outbound call, I get the following output:
> >>>> -- Called g2/5148346
> >>>> -- Zap/25-1 answered SIP/412-9b72
> >>>>
> >>>> However, my number never rings. After about 30 seconds, I get a message
> >>>> saying my call could not be completed as dialed. Almost like it didn't
> >>>> get all of the digits. Is there a way to inject a pause before dialing?
> >>>> Any other thoughts? Any help is greatly appreciated.
> >>>>
> >>>> _______________________________________________
> >>>> --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
> >>
> > _______________________________________________
> > --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