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SIP(星号)的NAT配置

[英]NAT configuration for SIP(Asterisk)

我安装了星号服务器,并在尝试时注册了几个SIP用户

*CLI> sip show peers

Name/username          Host            Dyn Nat ACL Port     Status     

2000/2000              (Unspecified)   D           5060     Unmonitored 


2005/2005              (Unspecified)   D  *N   *   0        Unmonitored 

6 sip peers [Monitored: 0 online, 0 offline Unmonitored: 5 online, 1 offline]

让我知道如何为特定SIP用户配置NAT设置,例如在这种情况下2000的NAT为空白,而2005的NAT为N。

您可以使用CLI编辑sip * .conf(根据您的设置)。

到目前为止,Asterisk nat支持已演变为以下选项:

nat = no                ; Do no special NAT handling other than RFC3581
nat = force_rport       ; Pretend there was an rport parameter even if there wasn't
nat = comedia           ; Send media to the port Asterisk received it from regardless of where the SDP says to send it.
nat = auto_force_rport  ; Set the force_rport option if Asterisk detects NAT (default)
nat = auto_comedia      ; Set the comedia option if Asterisk detects NAT

不要忘了为有格调的用户设置canreinvite = no。

我在下面为用户681显示了一个示例。

[681]
deny=0.0.0.0/0.0.0.0
type=friend
secret=123456
qualify=yes
port=5060
nat=yes
dtmfmode=rfc2833
dial=SIP/681
context=from-internal
canreinvite=no
callgroup=
callerid=device <681>
accountcode=
call-limit=50

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