To Nat or not to nat?

Discussion in 'General' started by abnothing, Nov 19, 2009.

  1. abnothing

    Joined:
    Oct 6, 2009
    Messages:
    5
    Likes Received:
    0
    Ok,
    So I have a Elastix box setup, everything is working... extensions, PSTN Gateway, all of it. EXCEPT when I try to take a phone off site. The phone shows "No Service" both on my Aastra 57i as well as my Pollycom 560, Now I know the E-box is setup 1to1 Nat, and the Phones that are offsite are on static ip's on a T-1. I have configured sip_nat.conf for nat:

    nat=yes
    externip=(outside IP of server)
    fromdomain-(Server Name)
    localnet=192.168.0.110/255.255.255.0
    canreinvite=no

    and this is still not working. ANY SUGGESTIONS???????


    Thanks ahead of time.

    BTW The phones can make calls out but they can't recieve calls if you call the extension it comes back busy.
     
  2. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    When Off-site the phones need to REGISTER(qualify) with the external address of the Asterisk server to GET calls, UDP port 5060 connections must be be bidirectional for those phones through any router/firewall you might have. If indeed they all have individual public IP's then NAT is no on that extension (and on the phone also) and set the extension on the Eastix box to the individual IP of the Off-site phone, If they are behind a NAT router then NAT is yes and you must program your Remote router to pass bi-directionally connections on 5060 to those particular extensions so Asterisk and the routers can re-write the headers to suit. (further the rtp connection (audio) must similarly be routed/translated.)

    From the asterisk CLI sip show peers will identify known extensions, sip show peer <extension> will give more detrail about that extension, sip debug ip <address of extension> and and sip debug peer <extension>, will be diagnostic. as will rtp debug ip <address of extension>, when you get that far. It is usually less trouble to leave NAT on and let the phones/asterisk do the rewrites after you have you NAT boxes setup right.
     

Share This Page