【发布时间】:2011-06-05 07:52:06
【问题描述】:
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通话时,Red5phone 结束通话并显示忙或被拒绝?
我对此一无所知。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()
【问题讨论】:
-
您是否尝试过登录 Asterisk 控制台并执行“SIP SET DEBUG”并观看对话?两个扩展都正确注册了吗?我也没有在您的 sip.conf 中看到您的 SIP 提供商的出站分支。
-
它在星号处显示警告“没有用于分机的应用程序拨号(telpun,2002,1)”
-
您在拨分机 2002 吗?因为这就是它的样子。
-
对不起,我弄错了。我的意思是“没有申请分机拨号(telpun,1001,1)”
标签: asterisk sip voip phone-call red5