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