[Asterisk-Users] SPA and NAT traversal

[Jim Sturtevant]

I appreciate everyone's help with setting up an external extension.


Here's a diagram

{SPA2000} - NAT1 - Internet - NAT2 - Asterisk - SIPPhone

SIPPhone is on the same internal subnet as *
NAT2 has a public/staic IP and ports are forwarded to Asterisk

I can successfully do the following:

1. call from SPA2000 - Asterisk VM
2. call from SPA2000 to SIPPhone
3. call from SPA2000 to outside PST as long as I'm using IAX to the ITSP

What I can't do is call from SPA2000 to an outside PSTN if the ITSP is SIP.
When I call the outside phone number I don't hear any ring back or when the
called party answers.   If I have RTP DEBUG on at the CLI I don't see any
RTP packets at all so it appears * is outside the media stream.

SIP-PSTN works great so long as the ITSP is being reached by IAX.

This situation exists regardless of the calue for canreinvite.

SIP PEER:
  * Name       : 202
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : sip
  Language     :
  AMA flags    : Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  LastMsgsSent : -1
  Inc. limit   : 0
  Outg. limit  : 0
  Dynamic      : Yes
  Callerid     : "" <>
  Expire       : 2749
  Expiry       : 900
  Insecure     : no
  Nat          : Always
  ACL          : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : 24.6.249.xxx Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 202
  Codecs       : 0x4 (ulaw)
  Codec Order  : (ulaw)
  Status       : UNKNOWN
  Useragent    : Sipura/SPA2000-2.0.10(c)
  Reg. Contact : sip:202 at 24.6.249.xxx:5060


[202]                   ;test ata
type=friend
username=202
secret=
host=dynamic
nat=no
reinvite=no             ;stay in the call
canreinvite=no          ;keeps asterisk in the media stream
disallow=all
allow=ulaw
context=sip

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nabeel
Jafferali
Sent: Saturday, April 09, 2005 1:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SPA and NAT traversal

> In your second option using a STUN server would I need to setup my
> own STUN server? 

No, use FWD or xten's STUN servers.

-- 
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698

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