Asterisk & freePBX - asterisk

Question

I m beginner for asterisk, so I cannot transfer call from main line to asterisk line, can anyone help me??
I have Asterisk card which have 4 port, 2 for FXO and 2 for FXS and I attached 2 land-line on FXS port and plugged PSTN line in FXO port, I generated DAHDI extension for those two land-line one was 101 and second one is 102, I check both can call each-other successfully, using soft-phone also can call on 101 and 102 but problem is there when someone call on land-line they cannot ring and cannot attend the call, so please some give me dial plane.
I also configure
extension.conf
[incoming]
exten => s,1,Answer( )
exten => s,2,Background(enter-ext-of-person)
exten => 101,1,Dial(Dahdi/1,10)
exten => 101,2,Playback(vm-nobodyavail)
exten => 101,3,Hangup( )
exten => 101,102,Playback(tt-allbusy)
exten => 101,103,Hangup( )
exten => 102,1,Dial(SIP/Jane,10)
exten => 102,2,Playback(vm-nobodyavail)
exten => 102,3,Hangup( )
exten => 102,102,Playback(tt-allbusy)
exten => 102,103,Hangup( )
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup( )
[internal]
exten => 101,1,Dial(Dahdi/1,,r)
exten => tejas,1,Dial(Dahdi/1,,r)
exten => 102,1,Dial(Dahdi/chirag,,r)
exten => chirag,1,Dial(Dahdi/chirag,,r)
but still unsuccess....
so please help me....
for your more information I will paste some other .conf file
/etc/dahdi/system.conf
fxsks=1,2
fxoks=3,4
loadzone=in
defaultzone=in
As show in above file system.conf in this fxsks channels are 1 & 2 and fxoks channels are 3 & 4 but I also used freePBX for gui mode in this When I searched Connectivity => Dahdi then I got fxsks channels are 3 & 4 and fxoks channels are 1 & 2, which one is right???
/etc/asterisk/chan_dahdi.conf
[general]
#include chan_dahdi_general.conf
#include chan_dahdi_general_custome.conf
[channels]
language=en
busydetect=yes
busycount=10
usecallerid=yes
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=no
immediate=no
faxdetect=no
rxgain=0.0
txgain=0.0
#include chan_dahdi_channels_custem.conf
#include chan_dahdi_groups.conf
#include chan_dahdi_additional.conf
/etc/asterisk/dahdi-channels.conf
;line="1 WCTDM/4/0 FXSKS (in use) (EC:MG2-INACTIVE)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel=>1
callerid=
group=
context=default
;line="2 WCTDM/4/1 FXSKS (in use) (EC:MG2-INACTIVE)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel=>2
callerid=
group=
context=default
;line="3 WCTDM/4/2 FXOKS (in use) (EC:MG2-INACTIVE)"
signalling=fxo_ks
callerid="channel 3" <4003>
mailbox=4003
group=5
context=from-internal
channel=>3
callerid=
mailbox=
group=
context=default
;line="4 WCTDM/4/3 FXOKS (in use) (EC:MG2-INACTIVE)"
signalling=fxo_ks
callerid="channel 4" <4004>
mailbox=4004
group=5
context=from-internal
channel=>4
callerid=
mailbox=
group=
context=default
I got one more conf file which name is Zapata which I post bellow..
etc/asterisk/zapata.conf.template
[channels]
language=en
#include zapata_additional.conf
context=from-pstn
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
channel=1-2
and more thing I done change just in extension.conf which I mentioned in starting of discussion
I want share some more information, I install freePBX in server PC based on CentOS without gui interface, and I used freePBX in other pc using IP address of server.
And I made some extension based on SIP and Dahdi and its works successfully, If I call 101(Dahdi extension) from 105(SIP Extension) using soft-phone its work.
But when I try to call from my phone to landline then Dahdi extension line not get ring.
I also try to modify extension.conf file which I mentioned in above comment..
Tell one thing which way is better using freePBX or using modification in conf file??
Thanks....

Answer 1

Got it -- you don't have a context defined for from-pstn as specified in dahdi-channels.conf
Outside of freePBX the raw asterisk configuration would be, in your extensions.conf you'll need to add a section like this:
[from-pstn]
exten => _X.,1,Noop(Incoming call "from PSTN")
same => n,Answer()
same => n,Playback(demo-congrats)
same => n,Hangup()
In dahdi-channels.conf in the first two sections there's a definition of context=from-pstn which you'll need an accompanying context in your extensions.conf
The [bracketed] items are "contexts" in Asterisk, and specify a certain context in which the dialplan operates. More information can be found, especially, from the (free) book "Asterisk: The Future of Telephony"