-- Jim Dickenson mailto:[email protected]
CfMC http://www.cfmc.com/ On Sep 29, 2010, at 10:20 AM, A J Stiles wrote: > On Wednesday 29 Sep 2010, Songtao Yu wrote: >> Hi All, >> >> When I tried to write my dial plan as below for my FXO port, which connects >> one PSTN line: >> >> [from-pstn] >> exten =>s,1,Answer() >> exten =>s,n,Wait(1) >> exten =>_X.,1,Dial(DAHDI/1) >> exten =>_X.,n,Hangup >> >> I got the following message: >> Connected to Asterisk 1.6.2.13 currently running on fax (pid = 8154) >> Verbosity was 0 and is now 4 >> -- Starting simple switch on 'DAHDI/1-1' >> -- Executing [...@from-pstn:1] Answer("DAHDI/1-1", "") in new stack >> -- Executing [...@from-pstn:2] Wait("DAHDI/1-1", "1") in new stack >> -- Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN' >> -- Hungup 'DAHDI/1-1' >> >> But if I changed the "_X." to "S" extension, I can get the whole thing to >> work well: [from-pstn] >> exten =>s,1,Answer() >> exten =>s,n,Wait(1) >> exten =>s,n,Dial(DAHDI/3) >> exten =>s,n,Hangup >> >> Would you please let me which casuses this issue? > > Extensions represent different numbers dialled by the calling party. > > An FXO port has only *one* number associated with it -- the number of the > POTS > line to which it is connected. It does not, therefore, have to be able to > differentiate between extensions. Incoming calls just go straight to the "s" > extension of the context associated with the channel. > > If for some reason you have more than one FXO port (ordinarily, you would > get > multiple lines by means of ISDN), then just bring each one in on a separate > context. > Either that or look at the channel in the dialplan and do what you want based on which channel the call was received on. > -- > AJS > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
