Asterisk的呼出呼叫失败， '这里正忙' 的错误

Question

我有星号11.7.0在CentOS 6.4 64配置有以下sip.conf配置：

[general] 
register =>myuser:pass@voipproviderip 
registertimeout=20 
context=incoming 
allowoverlap=no 
bindport=5060 
bindaddr=192.168.0.3 
srvlookup=no 
subscribecontext=from-sip 


[VoIPProvider] 
canreinvite=yes 
username=myuser 
fromuser=myuser 
secret=mypass 
context=incoming 
type=friend 
;fromdomain=voipproviderip 
host=voipproviderip 
dtmfmode=rfc2833 
disallow=all 
allow=alaw 
allow=ulaw 
nat=yes 
insecure=very 




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

; ext 101 
[101] 
type=friend 
host=dynamic 
secret=MyPass123 
context=default 
mailbox=101@default 
callgroup=1 
pickupgroup=1 
dtmfmode=rfc2833 
canreinvite=no 

; ext 102 
[102] 
type=friend 
host=dynamic 
secret=MyPass123 
context=default 
mailbox=102@default 
callgroup=1 
pickupgroup=1 
dtmfmode=rfc2833 
canreinvite=no 

; ext 103 
[103] 
type=friend 
host=dynamic 
secret=MyPass123 
context=default 
mailbox=103@default 
callgroup=1 
pickupgroup=1 
dtmfmode=rfc2833 
canreinvite=no 

; ext 104 
[104] 
type=friend 
host=dynamic 
secret=MyPass123 
context=default 
mailbox=104@default 
callgroup=1 
pickupgroup=1 
dtmfmode=rfc2833 
canreinvite=no 

; ext 105 
[105] 
type=friend 
host=dynamic 
secret=MyPass123 
context=default 
mailbox=105@default 
callgroup=1 
pickupgroup=1 
dtmfmode=rfc2833 
canreinvite=no 

; ext 106 
[106] 
type=friend 
host=dynamic 
secret=MyPass123 
context=default 
mailbox=106@default 
callgroup=1 
pickupgroup=1 
dtmfmode=rfc2833 
canreinvite=no 

; ext 100 
[107] 
type=friend 
host=dynamic 
secret=MyPass123 
context=default 
mailbox=107@default 
callgroup=1 
pickupgroup=1 
dtmfmode=rfc2833 
canreinvite=no 

; ext 108 
[108] 
type=friend 
host=dynamic 
secret=MyPass123 
context=default 
mailbox=108@default 
callgroup=1 
pickupgroup=1 
dtmfmode=rfc2833 
canreinvite=no 

和以下的extensions.conf：

[default] 
include => internal 
include => incoming 
include => outgoing 


[incoming] 
; Ring on extension 100, 200 and the mobile phone. 
exten => s,1,Answer() 
exten => s,n,Dial(SIP/101&SIP/103,30,r,t,) 


; Pass unanswered call to a mobile phone 
exten => s,n,Dial(SIP/101&SIP/103/&SIP/100,30,r) 
    same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) 
    same => n(unavail),VoiceMail(100@default,u) 
    same => n,Hangup() 
    same => n(busy),VoiceMail(100@default,b) 
    same => n,Hangup() 


[outgoing] 
; Outbound calls can be routed based on the number of digits dialled (or the value of the first few digits) 
;exten=> _XXXXXXXXXXXX.,1,Dial(SIP/VoIPProvider/${EXTEN}) 
;exten=> _XXXXXXXXXXXX.,2,Hangup 

exten => _XXXXXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN}) 
exten => _XXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN}) 

exten => _XXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN}) 
exten => _XXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN}) 

[internal] 
; Calls between employees (between extensions) 
exten => _XXX,1,Dial(SIP/${EXTEN},60) 
include => outgoing 


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

; Calls to ext 101 
exten => 101,1,Dial(SIP/101,20) 
exten => 101,n,VoiceMail(101,u) 
exten => 101,n,Hangup 

; Calls to ext 102 
exten => 102,1,Dial(SIP/102,20) 
exten => 102,n,VoiceMail(102,u) 
exten => 102,n,Hangup 

; Calls to ext 103 
exten => 103,1,Dial(SIP/103,20) 
exten => 103,n,VoiceMail(103,u) 
exten => 103,n,Hangup 

; Calls to ext 104 
exten => 104,1,Dial(SIP/104,20) 
exten => 104,n,VoiceMail(104,u) 
exten => 104,n,Hangup 

; Calls to ext 105 
exten => 105,1,Dial(SIP/105,20) 
exten => 105,n,VoiceMail(105,u) 
exten => 105,n,Hangup 

; Calls to ext 106 
exten => 106,1,Dial(SIP/106,20) 
exten => 106,n,VoiceMail(106,u) 
exten => 106,n,Hangup 

; Calls to ext 107 
exten => 107,1,Dial(SIP/107,20) 
exten => 107,n,VoiceMail(107,u) 
exten => 107,n,Hangup 

; Calls to ext 108 
exten => 108,1,Dial(SIP/108,20) 
exten => 108,n,VoiceMail(108,u) 
exten => 108,n,Hangup 

每当我从CLI>console dial mynumber拨一个号码，我看到了如下因素输出：

-- Executing [mynumber@default:1] Dial("Console/dsp", "SIP/VoIPProvider/mynumber") in new stack 
    == Using SIP RTP CoS mark 5 
    -- Called SIP/VoIPProvider/mynumber 
    -- Got SIP response 486 "Busy Here" back from voipproviderip:5060 
    -- SIP/VoIPProvider-00000000 is busy 
    == Everyone is busy/congested at this time (1:1/0/0) 
    -- Auto fallthrough, channel 'Console/dsp' status is 'BUSY' 

我曾尝试所有可能的拨打号码的方式（包括/不包括国家代码等），但它总是与486 "Busy Here"重复播放，是否需要支付我的voip服务费用？

Answer 1

我会尝试拨号（SIP/mynumber @ VoIPProvider）; 另外，什么时候看你什么时候表演同辈？