SIP(Asterisk) 的 NAT 配置

发布于 2024-10-12 14:43:24 字数 463 浏览 7 评论 0原文

我安装了一个星号服务器，并且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]

请让我知道如何为特定 SIP 用户配置 NAT 设置，例如本例中 2000 年的 NAT 为空白，2005 年的 NAT 为 N。

原文

i have a asterisk server installed and have registered few SIP users when i try

*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]

Let me know how i can configure the NAT settings for a particular SIP user like for this case 2000 has NAT as blank and 2005 has NAT as N.

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挽你眉间 2024-10-19 14:43:25

您可以使用 CLI 编辑 sip*.conf（根据您的设置）。

到目前为止，Asterisk nat 支持已发展到以下选项：

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

不要忘记为 natted 用户设置 canreinvite=no。

下面我为用户 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

You can use CLI to edit sip*.conf (according to your settings).

By now Asterisk nat support has evolved to these options:

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

Don't forget to set canreinvite=no for natted users.

I have shown an example below for user 681.

[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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