A couple things - First, in extensions.con your context is [my-phone], but you're using my-phones in your dahdi and sip.conf files.
Second, you need an 's' extension somewhere in your receiving context in order for asterisk to answer the incoming analog call. Third, I think you've got some issues with your Dial statements, but I'm on my phone right now and can't really diagnose them. I'll take a look later when I'm back at a desk, if no one else has commented by then. Thanks, --Warren Selby, dCAP On Jun 28, 2011, at 12:30 PM, "motty.cruz" <[email protected]> wrote: > Hello, I have Asterisk 1.6 running on Centos, Also I have one analog > telephone line coming on > Wildcard TDM400P REV E/F Board 5 > > I can't get asterisk to dectect call coming from analog line. > Here is my /etc/dahdi/system.conf > fxsks=1 > > # global data > loadzone = us > defaultzone = us > > > /etc/asterisk/chan_dahdi.conf > [channels] > language=en > context=my-phones > switchtype=national > signalling=fxs_ks > channel => 1 > > > /etc/asterisk/extensions.conf > [globals] > CONSOLE=DAHDI/1 > TRUNK=DAHDI/4 > TRUNKMSD=1 > > [my-phone] > exten => 2000,1,Dial(DAHDI/1/116) > exten => 2000,2,cONGESTION > > exten => 2001,1,Dial(DAHDI/1/7608514114) ; analog phoneline > exten => 2001,2,HangUp() > > exten => 1001,1,Dial(DAHDI/1/7608514114) > exten => 1001,2,HangUp() > > exten => 1111,1,Dial(DAHDI/1/7608514114) > exten => l111,2,HangUp() > > > /etc/asterisk/sip.conf > [general] > port = 5060 > context = others > > [2000] > type=friend > context=my-phones > secret=1234 > host=dynamic > > [2001] > type=friend > context=my-phones > secret=1234 > host=dynamic > > > [1001] > type=friend > context=my-phones > secret=1234 > > [1111] > type=friend > context=my-phones > secret=1234 > > > [phonesys] > type=friend > username=user1 > secret=1234 > host=dynamic > context=my-phones > > > Any suggestions are welcome. > > Thanks, > motty > > > -- > _____________________________________________________________________ > -- 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
