No, it's an Access Bank II SNMP.
Thanks,
James
C F wrote:
> 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.
>>>>>>
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