[asterisk-users] dial in group

[Paul]

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
>>>      
>>>