asterisk users - Jan 2005 - Sipura Behind NAT howto

I am trying to get a SPA-3000 to work behind NAT - for the sake of the 
exercice.

The SPA is on the local network at the address 192.168.0.125 behind a 
NATted linux router.

The machine I am trying to work with is a friend's (let's call it 
lolo.dyndns.org) and I've installed Asterisk 1.0.3 on it.

I can see the SPA register but when I try to make an outbound call I get 
the message:

Jan 24 14:35:03 NOTICE[3184]: chan_sip.c:7295 handle_request: Unable to 
create/find channel
Jan 24 14:35:03 NOTICE[3184]: chan_sip.c:7295 handle_request: Unable to 
create/find channel
Jan 24 14:35:04 NOTICE[3184]: chan_sip.c:7295 handle_request: Unable to 
create/find channel

Also, when I comment out the 'secret=' line from sip.conf, everything 
seems to work just fine...

Any ideas what's going on?

Cheers,
Jean-Michel.

> I am trying to get a SPA-3000 to work behind NAT - for the
> sake of the exercice.
Post the relevant entries from sip.conf and extensions.conf, and the
relevant fields from the SPA-3000 Line 1 tab.

-- 
Nabeel Jafferali
Tel: +1 (416) 628-9342  Toronto
     +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeel<at>jafferali.net

Nabeel Jafferali wrote:
>>I am trying to get a SPA-3000 to work behind NAT - for the
>>sake of the exercice.
>>    
>>
>
>Post the relevant entries from sip.conf and extensions.conf, and the
>relevant fields from the SPA-3000 Line 1 tab.
>  
>
Sure!

 From sip.conf
========
[2001]
type=friend           ; This device takes and makes calls
username=2001         ; Username on device
; secret=2001           ; Password for device
host=dynamic          ; This host is not on the same IP addr every time
context=from-sip      ; Inbound calls from this host go here
; mailbox=2001          ; Activate the message waiting light if this
allow=all
dtmfmode=inband
nat=yes
qualify=yes


 From extensions.conf
=============
[from-sip]
include => full-access
include => record-messages
exten   => 9,1,Goto(ivr-dialout,s,1)


Sipura Line 1:
========Line Enable: yes

SAS Enable: no
SAS DLG Refresh Intvl: 30
SAS Inbound RTP Sink: <blank>

NAT Mapping Enable: no
NAT Keep Alive Enable: no
NAT Keep Alive Msg: yes

NAT Keep Alive Proxy: <blank>
SIP TOS/DiffServ Value:0x68
Network Jitter Level: high
RTP TOS/DiffServ Value:0xb8

SIP Port: 5060
SIP 100REL Enable: no
EXT SIP Port: <blank>
Auth Resync-Reboot: yes
SIP Proxy-Require: <blank>
SIP Remote-Party-ID: no
SIP Debug Option: none
RTP Log Intvl: 0

Blind Attn-Xfer Enable: no
MOH Server: <blank>
Xfer When Hangup Conf: yes

Proxy: lolo.dyndns.org
Use Outbound Proxy: no
Outbound Proxy: <blank>
Use OB Proxy In Dialog: yes
Register: yes
Make Call Without Reg: no
Register Expires: 3600
Ans Call Without Reg: no
Use DNS SRV: no
DNS SRV Auto Prefix: no
Proxy Fallback Intvl: 3600

Display Name: 2001
User ID: 2001
Password: 2001
Use Auth ID: yes
Auth ID: 2001
Mini Certificate: <blank>
SRTP Private Key: <blank>

<snip "Supplementary Service Subscription" and "Audio
Configuration">

Dial Plan: L:4,S:1,(0262xxxxxx|0692xxxxxx|0[1-9]xxxxxxxx|00x*|x*)
Enable IP Dialing: no

<snip "FXS Port Polarity Configuration">


There. Sorry for the lengthy post... SIPURAs have a million options :)

Thanks in advance for any tips you might share...

Cheers,
Jean-Michel.

> I am trying to get a SPA-3000 to work behind NAT - for the
> sake of the
> exercice.
I went through your sip.conf. With the SPA-3000, because it supports
STUN, I have found it unnecessary to use nat=yes. Instead, I prefer to
let the device handle the NAT mapping. Please try enabling everything
under NAT Support Parameters on the SIP tab and put in a STUN server
(stun.xten.net comes to mind). Then, enable NAT Mapping and NAT Keep
Alive on the Line 1 tab. Remove nat=yes from sip.conf and try again.

-- 
Nabeel Jafferali
Tel: +1 (416) 628-9342  Toronto
     +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeel<at>jafferali.net