[Asterisk-Users] setting SIP to dial PSTN with TDM400P

[Muhammad Haris]

to dear martijn,

i made every possible change i can make.... 
i have a TDM400P Zap card...
i had connected PSTN line to FXO Kewlstart at channel 1.
and analog phone to FXS Kewlstart at Channel 4.
i can hear continous ring tone when i hook up the receiver.
plz have a look at my confs.

my extension.conf is as follows;

[pstn-outbound]
exten => _.,1,Dial(Zap/1/${EXTEN})
exten => _.,2,Congestion

my zaptel.conf is as follows:

 [channels]
;
; Default language
;
language=en
musiconhold=default
usercallerid=yes
hidecallerid=no
callreturn=yes
callprogress=no

rxwink =300
echotraining=800
rxgain=0.0
txgain=0.0

busydetect=1
busycount=7

immediate=no

signalling=fxo_ks

;callerid=asreceived

context=pstn-outbound
channel=1

relaxdtmf=yes
callwaiting=yes
;
; Support three-way calling
;
threewaycalling=yes
;
; Support flash-hook call transfer (requires three way calling)
;
transfer=yes

********************************************************

when i dial a local number say (6998256) from analog phone set then
asterisk shows following messages.....................

*CLI>     -- Starting simple switch on 'Zap/1-1'
    -- Executing Dial("Zap/1-1", "Zap/1/6998256") in new stack
    -- Called 1/6998256
    -- Zap/1/6998256-busy-1013475805 is busy
    -- Hungup 'Zap/1/6998256-busy-1013475805'
  == Everyone is busy/congested at this time
    -- Timeout on Zap/1-1
  == CDR updated on Zap/1-1

******************************************************
please reply with your suggestions i always take care to run ztcfg
command whenever i made any changes to zaptel.conf.plz help me solving
this problem