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 <[EMAIL PROTECTED]> 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 >>>> >>>> > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users