I had to set SETTINGS > SIP > LOCAL SETTINGS > Local SIP Port: 5062 (first nonzero value I tired).

I got logged in!!!! Can’t call or be called, but I got REGISTERED! The rest is usually much easier. I think I have NAT RTP issues now.

I will look at these next:

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

THANKS AGAIN.

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sean Devoy
Sent: Friday, December 07, 2012 3:15 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Polycom IP 335 - 401 Unauthorized .... NAT ISSUE?

First Brian THANKS, it clearly took some time to put together such a detailed response.

I reset to factory settings.

I have updated to a minor newer version than yours.

Yours

UC Software Version

4.0.2.11307

Mine

UC Software Version

4.0.3.7562

I set everything EXACTLY as you specified in your post except my server, extension, password, etc where it applies.

It still fails. There is an interesting difference in the SIP Messages though. From yours the phone sends CSeq: 1 Register, gets a response for CSeq: 1 Register (with the nonce), then your phone sends CSeq: 2 Register … From mine the phone sends CSeq: 1 Register again, and again, and again. I still think it is actually not receiving the 401 message with the nonce.

One other minor difference that may by important. On your FIRST 401 Unauthorized Message Via line says

Via: SIP/2.0/UDP 10.0.0.39;branch=z9hG4bK848ac3ba5589D827;received=76.238.166.184;rport=5060

Mine says:

Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa9b37440268F7B2B;received=71.127.152.57

It does not have an rport. I don’t know if that matters. It doesn’t it matter that other phones here are using port 5060, right? Are their other ports I can specify for rport? How can I tell if the phone is actually getting the 401?

The syslog from the phone says:

sip |4|03|Registration failed User: 228, Error Code:480 Temporarily not available 10.10.40.47 07/12 14:48:17.315

Thanks again for your help.

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Friday, December 07, 2012 2:11 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Polycom IP 335 - 401 Unauthorized .... NAT ISSUE?

This might help you:

http://pastebin.freeswitch.org/20301

Try comparing the SIP messages to yours. Notice that it tries to register twice, the first one is Unauthorized.

On Fri, Dec 7, 2012 at 1:54 PM, Brian Foster <bdfoster@endigotech.com> wrote:

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 Number	2345-12375-001 Rev:A
MAC Address	00:04:F2:37:3D:C0
IP Address	10.0.0.39
UC Software Version	4.0.2.11307
BootROM Software Version	5.0.2.12692

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

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	0
* Media Port Start	0
Keep-Alive Interval (s)	0

Settings > Lines

Identification

Display Name	Brian Foster
Address	2546@pbx.endigovoip.com
Authentication User ID	2546
Authentication Password	[ ]
Label	2546
Type	(X) Private ( ) Shared
Third Party Name
Number of Line Keys	2
Calls Per Line	4
Ring Type	[Low Trill \/]

Outbound Proxy

Address
Port	0
Transport	[DNSnaptr \/]

Server 1

Address
Port	0
Transport	[DNSnaptr \/]
Expires (s)	3600
Register	(X) Yes ( ) No
Retry Timeout (ms)	0
Retry Maximum Count	3
Line Seize Timeout (s)	30

Server 2

Address
Port	0
Transport	[DNSnaptr \/]
Expires (s)	3600
Register	(X) Yes ( ) No
Retry Timeout (ms)	0
Retry Maximum Count	3
Line Seize Timeout (s)	30

Call Diversion

* Always Forward	(X) Enable ( ) Disable
* Always Forward To Contact

* If Busy, Forward	(X) Enable ( ) Disable
* If Busy, Forward To Contact

* On No Answer, Forward	(X) Enable ( ) Disable
* On No Answer, Forward To Contact
* No Answer Timeout (seconds)	55

* On Do Not Disturb, Forward	( ) Enable (X) Disable
* On Do Not Disturb, Forward To Contact

* Disable Forward For Shared Lines	(X) Yes ( ) No

* Forward Specific Caller	(X) Enable ( ) Disable

Message Center

Subscription Address
Callback Mode	[Registration \/]
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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http://www.freeswitch.org

_________________________________________________________________________
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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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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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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