NAT configuration for SIP(Asterisk)

Question

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.

Pujan Pujan · Accepted Answer · 2011-01-27T04:57:47

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

NAT configuration for SIP(Asterisk)

1 Answers