SPA 3102- STUN SERVER NOT REACHABLE

Hello,
Following are Line 1 settings of an SPA-3102
Nat mapping enabled=yes
Nat keep alive enabled=yes
Proxy server:correct proxy server specified
Outbound proxy server: Correct outbound proxy server specified
UserID and passwords are configured properly
SIP TAB:
RTP port min=16384
RTP port max=16390
Handle via received=yes
Insert via received=yes
Substitute via addr=yes
STUN enable=yes
STUN Server=aa.aa.com (correct STUN server configured)
Hadle via rport=yes
Inser via rport=yes
Send resp to SRC port=yes
STUN test enable=yes
TURN server=not configured
STUN test enable=yes
EXT IP=not configured
NAT keep alive INTVL=15
Problem:
On the REGISTER message I am receiving following warning
Warning: 399 spa "STUN Server Not Reachable"
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Would you please explain why I receive STUN not reachable warning? It is a reachable STUN server. SPA 3102 devices are placed behind ADSL modems of ITSP

I just noticed that you have Handle VIA received = yes, so if the proxy is responding with a 'received' tag, then the SPA would learn its external IP, without needing to contact the STUN server. I have a system that is behind NAT but does not need STUN or symmetric RTP, because it uses the IP from the received tag in the SDP. (In my case, I have the RTP ports forwarded to the SPA, so no port number translations are needed.)

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Message was edited by: Martin Thorpe -- hopefully removed the auto-'email address' tagging! (Argh, no, it didn't. Bodged a different way.)

Hi Lindsey,
Thanks for the quick response. Here's a complete SIP invite -- I've changed the telephone number and put spaces around @ signs again, but everything else is unmodified.
INVITE sip:spa-line1 @ 81.2.113.115:5060 SIP/2.0
Via: SIP/2.0/UDP 81.187.239.177:5060;branch=z9hG4bK4062e0e9;rport
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To:
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Date: Mon, 29 Oct 2012 19:51:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286
So there is a contact field in there as well.
That's from a slightly patched Asterisk server, which doesn't put a calling name in if it's blank -- by default if you didn't set a calling name, Asterisk will also set the calling name from the calling number and you'd get this instead:
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I've done product management myself so I know one customer asking for it to work a little differently (as opposed to it doing something wrong!) isn't going to make a change -- that's no problem at all. If it were to be changed, I'd rather the ATA didn't generate a calling name field in the CLID spill at all, rather than 'Unknown'. But hey, that's just my opinion!
For the avoidance of doubt, the ATA is always generating the calling *number* field in the CLID spill correctly.
Thanks again!
All the best,
Martin

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I am using SPA 3102 and D-Link wireless router together. Connection is Cable Modem to SPA 3102 to D-Link DIR-615.
Both the phone and wirless works fine in this set up. The main issue is when I use utorrent, SPA 3102 keeps resetting (loosing connection intermittently). All the three lights goes of for a second and then comes back. End result is, I have to keep resetting the routers (power off for 10 secs and switch on). This is really annoying, I have to keep doing it atleast 3 or 4 times a day. I guess the problem might be with the heavy downloading/uploading with utorrent. Any way out ? Help appreciated.
aj

Ooops, sorry My first post was a mistake. Here is my question:
I have modem connection to a remote site.This connection must go from wire to Ethernet. The modems are 2-wire, not dial-up, and are embedded in a device, so there is no access to the serial port of the device.
I have two SPA3102. On the remote site the modem is always connected to the FXS port. This port is configured with SAS Enable=Yes and SAS Inbound RTP Sink = $IP. On the local site when modem is connected the conection is automatically established. With Wireshark I can see that RTP audio stream is flowing in both directions but signal can not be heared on the server side. What should I do to hear the inbound stream? Is it possible with SPA3102?
First I tried with SPA112 as SAS on remote site. The signal passes in both directions and can be heared on SPA112 port. Same settings. But SPA112 causes delay of 140ms and timing is critical for my devices. SPA3102 has only 80ms delay with RTP Packet Size = 0.02 and modems work with two SPA3102 configured as hot line. The problem with hot line is that in case of connection fail I have to go to the remote site to disconnect the modem and connect it again. Not very convinient.
Thanks
Svilen

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