[Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't

[C F]

Is this an Adit 600?

On 2/1/06, james.texter at cox.net <james.texter at cox.net> 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, james.texter at cox.net <james.texter at cox.net> 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, james.texter at cox.net <james.texter at cox.net> 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.
> >>>>
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