thanks very much for your reply,Grygoriy,you are so warm-heart! thank you 
advance!
Here is my extensions.conf and meetme.conf. I don't use the digium card so I 
just use the ztdummy modules.
[meetme]
exten => 4105,1,Answer()
exten => 4105,n,meetme(99008664105|Ap)
exten => 4105,n,Hangup()
meetm.conf
[rooms]
conf =>4105


I have compare my two different manchines,(one work OK,and another is failed):
when  use "zap show channels" to see the channels status:
 Chan Extension  Context         Language   MOH Interpret       
 pseudo            default                    default    

then i dial the 4105 and channels show 
 Chan Extension  Context         Language   MOH Interpret       
 pseudo            default                    default    
pseudo            default                    default    

then i hangup,but the channels still have two pseudo:
 Chan Extension  Context         Language   MOH Interpret       
 pseudo            default                    default    
pseudo            default                    default    


then i try again,the Meetme didn't ctreat room anymore.

and i found a strange thing :
after i install the zaptel ,my asterisk didn't play any voice.
i use the Playback(Nomoney):
Executing [4...@4105:1] Answer("SIP/22238-08211340", "") in new stack
    -- Executing [4...@4105:2] Playback("SIP/22238-08211340", "NoMoney") in new 
stack
    -- <SIP/22238-08211340> Playing 'NoMoney' (language 'en')
It show well but no voice!!

Is it wrong in my system? thanks

2009-03-05 



邱磊 



发件人: Grygoriy Dobrovolskyy 
发送时间: 2009-03-04  16:30:06 
收件人: Asterisk Users Mailing List - Non-Commercial Discussion 
抄送: 
主题: Re: [asterisk-users] after install the zaptel but the rtp failed 
 



2009/3/4 邱磊 <qiulei...@163.com>

hi Grygoriy :
appreciate your reply ,
that's my cli command:
CLI> zap show status 
Description                              Alarms     IRQ        bpviol     CRC4  
    
ZTDUMMY/1 1                              UNCONFIGUR 0          0          0   

Is't all right? forward your echo .
thanks 

Yes normally you should have meetme working. Paste your extensions.conf here 
(only the context with the conference) Also the config of the sip peer who is 
trying to join the conference and more cli output during that join. 
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to