[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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