Hi Juan,
Thanks for your inputs, I tried with changes you suggested and find my observation. After adding context and extension able to make an outgoing call [Digium-fxs<3333> to X-lite<2000>]. But not able to make incoming call [X-lite<2000> to Digium-fxs<3333>]. Call failed with, (1) “*Call failed: 503 Service Unavailat 3333*” error message on X-lite (2) “CHANUNAVAIL” on asterisk CLI. **CLI> > Saved useragent "X-Lite release 1105d" for peer 2000* * == Using SIP RTP CoS mark 5* * -- Executing [3...@my-phones:1] Dial("SIP/2000-00000000", "Zap/1/3333") in new stack* *[May 6 13:02:44] WARNING[20496]: channel.c:4003 ast_request: No channel type registered for 'Zap'* *[May 6 13:02:44] WARNING[20496]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented)* * == Everyone is busy/congested at this time (1:0/0/1)* * -- Auto fallthrough, channel 'SIP/2000-00000000' status is 'CHANUNAVAIL'* Please find conf files below. chan_dahdi.conf ============ [channels] context=my-phones usecallerid=yes hidecallerid=no immediate=no signaling=fxo_ks echocancel=yes group=1 channel=1 sip.conf ====== [general] port=5060 bindaddr=0.0.0.0 context=my-phones [2000] type=friend context=my-phones secret=1234 host=dynamic extensions.conf =========== [my-phones] exten => 2000,1,Dial(SIP/2000) exten => 3333,1,Dial(Zap/1/3333) system.conf ======== fxoks=1 loadzone=us defaultzone=us Please let me know any other configuration needs to be done. On Fri, Apr 30, 2010 at 1:12 AM, Juan David Diaz <juanch...@gmail.com>wrote: > > > 2010/4/29 garge rama <garge.r...@gmail.com> > >> >> >> Hi, >> >> >> >> I am new to asterisk and trying to make calls with TDM400P asterisk digium >> card. >> >> >> >> I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and >> libpri-1.4.10.2 packages which are downloaded from asterisk website ( >> www.asterisk.org) >> >> and able to compile successfully. TDM400P Digium card (having only one FXS >> connected to J4) has installed successfully in PC. >> >> >> >> I would like to make calls across SIP [x-lite] to analog phone connected >> to TDM400P Digium card (fxs-j4). >> >> For this the following four conf files are modified as shown below. >> >> >> >> * chan_dahdi.conf* >> >> *==============* >> >> [channels] >> >> context=test >> >> usecallerid=yes >> >> hidecallerid=no >> >> immediate=no >> >> >> >> signaling=fxo_ks >> >> echocancel=yes >> >> group=1 >> >> channel=1 >> >> >> >> *extensions.conf*** >> >> *=============* >> >> [my-phones] ------------------->*EXTEN 3333 does not exists for your >> sip peer context* >> >> exten => 2000,1,Dial(SIP/2000) >> >> ; Should look like: >> > *exten => 3333,1,Asterisk_Application(Action)* ;Dial(Zap/1/${Phone_Number_you > want}) > >> [test] >> >> exten => 3333,1,Dial(Zap/1) >> >> exten => 3333,2,HangUp() >> >> >> >> *sip.conf*** >> >> *=======* >> >> [general] >> >> port = 5060 >> >> bindaddr = 0.0.0.0 >> >> context = others >> >> >> >> [2000] >> >> type=friend >> >> *context=**my-phones * >> >> secret=1234 >> >> host=dynamic >> >> >> >> *system.conf* >> >> *==========* >> >> fxoks=1 >> >> loadzone = be >> >> defaultzone = be >> >> >> >> With those changes x-lite getting registered with asterisk and analog >> device/phone is getting ring tone with off-hook and also getting debug >> prints on cli, but not able to make calls. >> >> >> >> Test Setup: >> >> ======== >> >> X-lite [configured as 2000, password… other info] running on asterisk PC >> à registered with asterisk. >> >> Analog phone connected to TDM400P Digium card - FXS-J4 running on same >> asterisk PC à getting ring tone >> >> >> >> Test Result: >> >> ========= >> >> Tried by calling 3333 from x-lite à getting message on CLI “call from >> ‘2000’ to ‘3333’ rejected because extension not found” >> >> Tried by calling 2000 from analog phone [Digium-FXS-J4] -> getting some >> engage/disconnected tone while pressing digts [2000] on phone itself. >> >> >> >> Welcome for your valuable suggestions and comments. Thank You in advance. >> >> >> >> Regards, >> >> Garge. >> >> >> >> -- >> >> _____________________________________________________________________ >> -- 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 >> > > > > -- > Juan. > Linux User #441131 > > -- > _____________________________________________________________________ > -- 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