Red5phone在呼叫建立之前忙或被拒绝?

时间:2011-06-05 07:52:07

标签: asterisk sip voip phone-call red5

我是这样的: 1.安装red5,星号 2.设置环境变量,如java,apache-ant 3.在星号处配置sip.conf 4.运行Red5phone

它已成功注册,但当我在两个客户端之间呼叫时,Red5phone结束呼叫并显示忙或被拒绝?

我对此一无所知。我该怎么办?

提前感谢。

我的sip.conf配置:

[general]
enabled=yes
bindaddr =0.0.0.0
context=lain-lain
allowoverlap=no
srvlookup=yes

[1000]
username=1000
secret=1234
host=dynamic
disallow=all
qualify=yes
type=peer
context=digium
allow=alaw
allow=ulaw

[1001]
username=1001
secret=1234
host=dynamic
disallow=all
qualify=yes
type=peer
context=digium
allow=alaw
allow=ulaw

我的extensions.conf:

[globals]
[general]
autofallthrough=yes
[lain-lain]
[digium]
exten => 1000,1,Dial(SIP/1000)
exten => 1001,1,Dial(SIP/1001)
include => internal
include => remote

[internal]
# This is how we get to our voicemail. Dial 123 from any SIP connected phone.
exten => 123,1,Answer()
exten => 123,2,VoiceMailMain(0203123456)
exten => 123,3,Hangup()
# If we’re trying to call any extension that starts with the number 2 and has 4 digits only, assume internal.
exten => _2XXX,1,NoOp()
exten => _2XXX,n,Dial(SIP/${EXTEN},30)
exten => _2XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _2XXX,n,Hangup()

[remote]
# Anything that isn’t internal we send to the PSTN.
exten => _X!,1,NoOp()
exten => _X!,n,Dial(SIP/siptrunk/${EXTEN})
exten => _X!,n,Hangup()

[incoming]
# This is where calls coming in from the PSTN are directed – see context setting in sip.conf
exten => _X.,1,NoOp()
# Try and call the desktop and mobile. If this fails, direct to voicemail.
exten => _X.,n,Dial(SIP/jamesdesktop)
exten => _X.,n,Dial(SIP/jamesmobile)
exten => _X.,n,VoiceMail(0203123456,u)
exten => _X.,n,Hangup()

1 个答案:

答案 0 :(得分:1)

首先,感谢@ gnxtech3的关注

最后,我做到了。解决方案是关于module.conf的配置

我把这行代码放在我的module.conf:

  

[modules]

     


  

     

load = app_dial.so

不要忘记重新加载星号。