[Asterisk-Users] A@H 2.6 : problem connecting call from PSTN

[WALTER LOH]

hi,

i have a pronlem connecting call from pstn with the following confuguration, 
please advice

extensions.conf

[from-trunk]
include => from-pstn

[from-pstn]
include => from-pstn-custom
include => ext-did
include => from-pstn-timecheck
exten => fax,1,Goto(ext-fax,in_fax,1)

extensions_custom.conf

[from-pstn-custom]

exten => s,1,Answer
exten => s,2,Background(demo-echodone)
exten => s,3,WaitExten(60)
exten => 1,1,Dial(SIP/201,10) ; as soon as i key 1 or 2, call drop when i'm 
calling from pstn but works on lan
exten => 2,1,Dial(SIP/203,10)

logs (asterisk -r -vvvv)

(from pstn; does not work properly)
--Executing GoTo("SIP/PSTN-6b31", "s") in new stack
== spawn extension (from-pstn,192.168.0.5, 1) exited non-zero on 
'SIP/PSTN-6b31'
--Executing GoTo("SIP/PSTN-6b31", "s|1") in new stack
--Goto (from-pstn,s,1)
--Executing Answer("SIP/PSTN-6b31", "") in new stack
--Executing BackGround("SIP?PSTN-6b31", "demo-echodone") in new stack
--Playong 'demo-echodone' (language 'en')
--Executing WaitExten("SIP/PSTN-6b31", "15") in new stack
==Spawn extension (from-pstn, s, 3) exited non-zero on 'SIP/PSTN-6b31'

(from lan; work fine)
--Executing GoTo("SIP/203-841a", "from-pstn|s|1") in new stack
--Goto (from-pstn,s,1)
--Executing Answer("SIP/203-841a", "") in new stack
--Executing BackGround("SIP/203-841a", "demo-echodone") in new stack
--Playing 'demo-echodone' (language 'en')
--Executing WaitExten("SIP/203-841a", "15") in new stack
== CDR updated on SIP/203-841a
--Executing Dial("SIP/203-841a", "SIP/201|10") in new stack
-- Called 201
--SIP/201-030c is ringing
--SIP/201-030c answered SIP/203-841a
--Attempting native bridge of SIP/204-841a and SIP/201-030c
==Spawn extension (from-pstn, 1, 1) exited non-zero on 'SIP/203-841a'
--Executing GoTo("SIP/203-841a", "S|1") in new stack
--Goto (from-pstn,s,1)
--Executing Answer("SIP/203-841a", "") in new stack
==Spawn extension (from-pstn, s, 1) exited non-zero on 'SIP/203-841a'

i found out that when using calling from lan(with x-lite), after the
exten => s,3,WaitExten(60) ; in extensions_custom.conf of [from-pstn-custom] 
there is this line that i notice "Oooh, got something to jump out with 
('1')!" that i cant find it when i'm with calling from PSTN, can anyone 
advice me on this?

(fro log file)
Apr 24 04:09:55 VERBOSE[3118] logger.c: -- Executing 
WaitExten("SIP/203-de7f", "60") in new stack
Apr 24 04:10:00 DEBUG[3118] pbx.c: Oooh, got something to jump out with 
('1')!

by the way i'm using spa-3000 connecting pstn to asterisk box through LAN,
and i have try changing the configuration

under pstn line-> audio configuration -> dtmf tx menthod : with various 
option inband,avt,info,auto,inband+info, avt+info (is this where spa-3000 
handle DTMF tones)

even with all the option here it is still not working, please advice

thanks

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