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

james.texter at cox.net james.texter at cox.net
Wed Feb 1 10:21:54 MST 2006


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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