Hi Marco,

Thanks for your reply. Dial peer is working normal, but i heard horrible noise instead ring tone. Is my digium tdm04b card has problem? I have tested tdm04b using zttest. It seems is working normal. Would you give me advice?

Thanks for help

Here is some test using zttest

zttest
Opened pseudo zap interface, measuring accuracy...
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
--- Results after 14 passes ---
Best: 99.987793 -- Worst: 99.987793 -- Average: 99.987793


Heere is some debug message.


   -- Registered SIP '1000' at 192.168.0.12 port 4910 expires 3600
   -- Saved useragent "X-Lite release 1002tx stamp 29712" for peer 1000
   -- Executing Answer("SIP/1000-081ac318", "") in new stack
   -- Executing Dial("SIP/1000-081ac318", "Zap/g1/23") in new stack
   -- Called g1/23
   -- Zap/1-1 answered SIP/1000-081ac318
   -- Hungup 'Zap/1-1'
 == Spawn extension (home, 923, 2) exited non-zero on 'SIP/1000-081ac318'

Regards,

Ganbaa

----- Original Message ----- From: "Marco Mouta" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]>
Sent: Monday, July 10, 2006 10:31 PM
Subject: Re: [asterisk-users] outgoing call problem


I'm not a a guru, but

Check this line:

exten => _9.,2,Dial(Zap/g1/${EXTEN})

do you really want to dial digit 9 through your ZapLine? are you
connected to another pbx?

If you don't want do dial 9 to PSTN line , but you want your users to
dial 9 to place outgoing calls, try this:
exten => _9.,2,Dial(Zap/g1/${EXTEN:1})

Hope it helps.


Ps. Give me some feedback if you solved the problem



On 7/10/06, Ganbaa <[EMAIL PROTECTED]> wrote:


Hi,

I have configured digium tdm04b card with asterisk on debian. Incoming call
is ok. But outgoing call has problem. Would you give me advice ?

Here is my config files.

zaptel.conf

fxsks=1
fxsks=2
fxsks=3
fxsks=4
loadzone=us
defaultzone=us

zapata.conf

[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=1
txgain=4
group=1
callgroup=1
pickupgroup=1
immediate=no
musiconhold=default
busydetect=yes
callprogress=no
channel => 1-4

extension.conf

[general]
static=yes
writeprotect=no

[home]
exten => s,1,Answer
exten => s,3,Playback(thank-you-cooperation)
exten => s,4,WaitExten

exten => _1XXX,1,Playback(thank-you-cooperation)
exten => _1XXX,2,Answer
exten => _1XXX,3,Wait(1)
exten => _1XXX,4,Playback(thank-you-for-calling)
exten => _1XXX,5,Dial(SIP/${EXTEN},10)
exten => _1XXX,8,Voicemail(u${EXTEN})
exten => _1XXX,9,Hangup
exten => _1XXX,103,Voicemail(b${EXTEN})
exten => _1XXX,104,Hangup

exten => _9.,1,Answer
exten => _9.,1,Playback(thank-you-cooperation)
exten => _9.,2,Dial(Zap/g1/${EXTEN})

[incoming]
exten => s,1,Answer()
exten => s,2,Background(/tmp/greetings)
;exten => s,2,Background(enter-phone-number10)
exten => 1,1,Playback(digits/1)
exten => 1,2,Goto(sumiya,s,1)
exten => 2,1,Playback(digits/2)
exten => 2,2,Goto(ganbaa,s,1)
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(incoming,s,1)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup( )

[sumiya]
exten => s,1,Dial(SIP/1001,10)
exten => s,2,Hangup

[ganbaa]
exten => s,1,Dial(SIP/1000,10)
exten => s,2,Hangup


Regards,


Ganbaa
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--
Com os melhores cumprimentos,

Marco Mouta
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