Question

我在CentOS 6.4 x64上配置了Asterisk 11.7.0，配置如下sip.conf：

[general]
register =>mynumber:mypass@xxx.xxx.xxx.xxx
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.0.3
srvlookup=no
subscribecontext=from-sip

; The SIP provider
[VoIPProvider]
canreinvite=no
username=mynumber
fromuser=mynumber
secret=mypass
context=incoming
type=friend
fromdomain=xxx.xxx.xxx.xxx
;host=xxx.xxx.xxx.xxx
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=yes
insecure=very

; ext 100
[100]
type=friend
host=dynamic
secret=MyPass123
context=internal
mailbox=100@default
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
canreinvite=no

; ext 200
[200]
type=friend
host=dynamic
secret=MyPass123
context=internal
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
canreinvite=no

关注extensions.conf:

[incoming]
; Ring on extension 100, 200 and the mobile phone.
exten => s,1,Answer()
exten => s,n,Dial(SIP/100&SIP/200&SIP/VoIPProvider/*320423456789,150,r,t,)

; Pass unanswered call to a mobile phone
exten => s,n,Dial(SIP/VoIPProvider/*320423456789,150,r)

; Still not answered? Pass unanswered calls to voicemail
exten => s,n,Voicemail(100,u)
exten => s,n,Hangup

[outgoing]
exten => _XXXXXXXXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
exten => _XXXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
exten => _XXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})

[internal]
exten => _XXX,1,Dial(SIP/${EXTEN})

; Calls to ext 100
exten => 100,1,Dial(SIP/100,20)
exten => 100,n,VoiceMail(100,u)
exten => 100,n,Hangup

; Calls to ext 200
exten => 200,1,Dial(SIP/100,20)
exten => 200,n,Hangup

当我尝试从IP电话拨打手机号码时，我在Asterisk CLI中看到以下输出：

-- Executing [XXXXXXXXX@outgoing:1] Dial("SIP/XXX-00000002", "SIP/VoIPProvider/XXXXXXXXX") in new stack

WARNING[19884][C-00000003]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/XXX-00000002' status is 'CHANUNAVAIL'

现在，我知道cause 20 - Subscriber absent意味着什么，但我确信我的手机号码存在并且可以访问，因为当我从我的手机（来电）拨打IP电话的号码时，它可以工作。

有什么建议吗？

Answer 1

我无法看到VoIPProvider条目如何用于拨出呼叫，因为它没有“主机”字段，因此Asterisk不知道应该在哪里发送SIP呼叫。

尝试在sip.conf中创建名为“VoIPProvider_Outgoing”或类似内容的新条目，并取消注释主机字段。然后在您的extensions.conf中将“VoIPProvider”替换为“VoIPProvider_Outgoing”。

Answer 2

请在拨号用户上下文中添加外发上下文。

喜欢在内部：

include =传出要么 include =传出

在Asterisk中意味着相同

星号拨出呼叫失败

2 个答案: