Let's do this step by step. First of all, my server is off site. We are going through a NAT in order to get to FreeSWITCH. Looks like this is the same setup you have.

I have the same phone as you do:
Phone Information
Phone Model SoundPoint IP 335
Part Number2345-12375-001 Rev:A
MAC Address00:04:F2:37:3D:C0
IP Address10.0.0.39
UC Software Version 4.0.2.11307
BootROM Software Version5.0.2.12692

Alright so now that we have that squared away, the next step is to set up the phone.

Settings > SIP

Local Settings:

Local SIP Port: 0
Calls Per Line Key: 4
New SDP Type: Disable
Live Communication Server Support: Disable
Non-Standard Line Seize: Enable
Digitmap: Not relevent
Digitmap Timeout: 3|3|3|3|3|3
Remove End-of-Dial Marker: Enable
Digit Impossible Match: 0

Outbound Proxy:

Address: <Blank>
Port: 0
Transport: DNSnaptr

Server 1:

Address: pbx.endigovoip.com
Port: 0
Transport: DNSnaptr (You shouldn't have issues with UDPonly, might be worth trying though.
Espires (s): 3600
Register: Yes
Retry Timeout (ms): 0
Retry Maximum Count: 3
Line Seize Timeout: 30

I do not have a second server.

Settings > Network > NAT

NAT

* IP Address
* Signalling Port
* Media Port Start
Keep-Alive Interval (s)

Settings > Lines

 Identification

Display Name
Address
Authentication User ID
Authentication Password
Label
Type  Private  Shared
Third Party Name
Number of Line Keys
Calls Per Line
Ring Type

 Outbound Proxy

Address
Port
Transport

 Server 1

Address
Port
Transport
Expires (s)
Register Yes  No
Retry Timeout (ms)
Retry Maximum Count
Line Seize Timeout (s)

 Server 2

Address
Port
Transport
Expires (s)
Register Yes  No
Retry Timeout (ms)
Retry Maximum Count
Line Seize Timeout (s)

 Call Diversion

* Always Forward  Enable  Disable
* Always Forward To Contact
* If Busy, Forward Enable  Disable
* If Busy, Forward To Contact
* On No Answer, Forward  Enable  Disable
* On No Answer, Forward To Contact
* No Answer Timeout (seconds)
* On Do Not Disturb, Forward  Enable  Disable
* On Do Not Disturb, Forward To Contact
* Disable Forward For Shared Lines  Yes  No
* Forward Specific Caller  Enable  Disable

 Message Center

Subscription Address
Callback Mode
Callback Contact


Check those and let us know where you stand after that.

-BDF

On Fri, Dec 7, 2012 at 1:20 PM, Steven Ayre <steveayre@gmail.com> wrote:
Try this parameter:
http://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport

or if that fails
http://wiki.freeswitch.org/wiki/NDLB#NDLB-connectile-dysfunction


On 7 December 2012 16:39, Sean Devoy <sdevoy@bizfocused.com> wrote:

HI All,

 

I am still banging my head against the wall here try to get a Polycom 335 to register w/ FS.  I have checked all the SERVER and USER/AUTH fields like 1000 times and 900 variations.  I think my problem may be NAT related.  I know on my CISCO 504G I had to enable several NAT features to work behind our firewall.  I am totally new to Polycom, so some very basic help is needed.

 

The server is remote but not behind a NAT there.  The phones are NAT’ed to the internet.  In the sofia sip trace I see this over and over:

   ------------------------------------------------------------------------

recv 552 bytes from udp/[71.127.152.57]:1026 at 16:26:07.358892:

   ------------------------------------------------------------------------

   REGISTER sip:fs_bfis.bizfocused.com SIP/2.0

   Via: SIP/2.0/UDP 10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5

   From: "228 Sean" <sip:228@fs_bfis.bizfocused.com>;tag=3F42C046-B61A297

   To: <sip:228@fs_bfis.bizfocused.com>

   CSeq: 1 REGISTER

   Call-ID: 2f482c2-2599cc43-1fb1a78@10.10.40.47

   Contact: <sip:228@10.10.40.47:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

   User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069

   Accept-Language: en

   Max-Forwards: 70

   Expires: 600

   Content-Length: 0

 

   ------------------------------------------------------------------------

send 710 bytes to udp/[71.127.152.57]:5060 at 16:26:07.359067:

   ------------------------------------------------------------------------

   SIP/2.0 401 Unauthorized

   Via: SIP/2.0/UDP 10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5;received=71.127.152.57

  From: "228 Sean" <sip:228@fs_bfis.bizfocused.com>;tag=3F42C046-B61A297

   To: <sip:228@fs_bfis.bizfocused.com>;tag=t232me1NSD02S

   Call-ID: 2f482c2-2599cc43-1fb1a78@10.10.40.47

   CSeq: 1 REGISTER

   User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120712T101002Z

   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

   Supported: precondition, path, replaces

   WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="b9583359-0163-4bf2-9818-788f64c34207", algorithm=MD5, qop="auth"

   Content-Length: 0

If I understand correctly, the server should be sending back this 401 message with the nonce so the phone can re-attempt the registration with an encrypted password.  If NAT is failing, the phone is never seeing the 401 w/ the nonce.

 

So what do I do in the WEB config interface to enable NAT on this phone?

 

Thanks,

Sean


_________________________________________________________________________
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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--
Brian D. Foster
Endigo Computer LLC
Email: bdfoster@endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

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