我在 CentOS 6.4 x64 上配置了 Asterisk 11.7.0，并具有以下sip.conf配置：
[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
并遵循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电话拨打我的手机号码时，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的意思，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？
我看不出VoIPProvider条目如何用于传出呼叫，因为它没有"主机"字段，因此Asterisk不知道SIP呼叫应该发送到哪里。
尝试在 sip.conf 中创建一个名为"VoIPProvider_Outgoing"或类似条目的新条目，并取消注释主机字段。然后在您的扩展.conf中将"VoIPProvider"替换为"VoIPProvider_Outgoing"。
请在拨号用户上下文中添加传出上下文。
如在内部下：
包含 = 传出或包含=传出
在星号中的意思相同