[asterisk-users] dial in group

[Eric "ManxPower" Wieling]

As SIP is not Analog FXO, my comments do not apply to them.  I have no 
idea if or which analog adapters might detect line voltage or dialtone.

Paul wrote:
> Do the SIP-FXO gateway devices do any better?
> 
> Eric "ManxPower" Wieling wrote:
> 
>> Asterisk does not detect analog ports with no line plugged in.  It does 
>> not test for dialtone before dialing (this applies to all analog cards 
>> except the X100P).
>>
>> Rilawich Ango wrote:
>>  
>>
>>> It works if it specified the port exactly plugged to PSTN.  I want to
>>> clarify the dial command here.
>>>
>>> Dial(zap/g1/1234567)
>>>
>>> It will try channel 1, if it is busy, congested then it will try
>>> channel 2 and so on, right?
>>> I wonder if I don't plug the PSTN to channel 1, there should not be a
>>> dial tone on it.  Why it still try channel 1 and make call using it?
>>>
>>> On Nov 25, 2007 5:00 AM, Gordon Henderson <gordon+asterisk at drogon.net> wrote:
>>>    
>>>
>>>> On Sat, 24 Nov 2007, Rilawich Ango wrote:
>>>>
>>>>      
>>>>
>>>>> I have a TDM400 with all FXO module in it. Only one channel (say
>>>>> channel 3) is plugged to PSTN. In my understand, a dial command
>>>>> Dial(zap/g1/12345677) should search an available channel, which is 3,
>>>>> in group 1 to make a call. However, I found that it will still use
>>>>> channel 1 to make call even it hasn't plugged to the PSTN. Below are
>>>>> the conf files.
>>>>>
>>>>> --zapata.conf--
>>>>> group=1
>>>>> signalling=fxs_ks
>>>>> context=incoming
>>>>> channel => 1-8
>>>>>        
>>>>>
>>>> You really only want
>>>>
>>>>   channel => 3
>>>>      
>>>>
> 
> 
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