[asterisk-users] problem making outgoing calls

[bikrish at aim.com]

Hi everybody

Here is my zapata.conf file and extension.conf file


zapata.conf

[channels]
;switchtype=national
;pridialplan=national
;signalling=pri_cpe
context=test
group=1
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
rxgain=0.0
txgain=0.0
callprogress=no
callerid=asreceived
pickupgroup=1
pri_dialplan=unknown
immediate=no
signalling=pri_cpe
switchtype=national
channel => 1-15,17-31



extension.conf

[test]
exten => s,1,Wait(1)??????????????????? ; Wait a second, just for fun
exten => s,n,Dial(SIP/2000)
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)
;exten => _X.,1,Dial(ZAP/g1/${EXTEN})
exten => _X.,1,Dial(ZAP/g1/${EXTEN},60)
exten => _X.,3,Hangup

I am from india. I am using centos 5 , the latest version of asterisk. With the above configuration i am able to make a call to my asterisk server and call to local mobile nos , but not able to call land line no.s and std no.s. Can anyone suggest where and what? i am missing. When i call to land line no or std no it gives me following log message

asterisk*CLI> 
??? -- Executing [32469868 at test:1] Dial("SIP/2002-097f63d0", "ZAP/g1/32469868|60") in new stack
??? -- Requested transfer capability: 0x00 - SPEECH
??? -- Called g1/32469868
??? -- Channel 0/1, span 1 got hangup request, cause 3
??? -- Hungup 'Zap/1-1'
? == Everyone is busy/congested at this time (1:0/0/1)
? == Auto fallthrough, channel 'SIP/2002-097f63d0' status is 'CHANUNAVAIL'


Thanks 

bikrish
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