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 callis 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 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos, Marco Mouta _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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