Red5phone 在呼叫建立之前占线或被拒绝?

发布于 2024-11-13 23:32:23 字数 1772 浏览 3 评论 0原文

我确实喜欢这样: 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

注册成功,但是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?

我对此没有任何想法。我必须做什么?

提前致谢。

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()

I did like this :
1. install red5,asterisk
2. setting environment variable, such as java, apache-ant
3. configure sip.conf at asterisk
4. Running Red5phone

It's successfully registered, but when I call between two client the Red5phone ended calling and shows that Busy or Rejected?

I don't have any idea about it. What must I do?

thanks in advance.

My sip.conf configuration :

[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

my 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()

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漆黑的白昼 2024-11-20 23:32:23

首先,感谢@gnxtech3的关注

,我终于成功了。解决方案是关于 module.conf 中的配置

我将这行代码放在我的 module.conf 中:

[模块]

.
.

加载=app_dial.so

不要忘记重新加载星号。

First, thanks to @gnxtech3 for your attention

finally, I make it. the solution is about configuration at module.conf

i put this line of code at my module.conf :

[modules]

.
.

load = app_dial.so

dont forget to reload asterisk.

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