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 InformationPhone ModelSoundPoint IP 335Part Number2345-12375-001 Rev:AMAC Address00:04:F2:37:3D:C0IP Address10.0.0.39UC Software Version4.0.2.11307BootROM 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 StartKeep-Alive Interval (s) Settings > Lines *Identification* Display NameAddressAuthentication User IDAuthentication PasswordLabelType Private SharedThird Party NameNumber of Line KeysCalls Per LineRing TypeLow TrillLow Double TrillMedium TrillMedium Double TrillHigh TrillHigh Double TrillHighest TrillHighest Double TrillBeebleTripletRingback-styleLow Trill PrecedenceRing Splash *Outbound Proxy* AddressPortTransport UDPOnly TCPpreferred DNSnaptr TCPonly TLS *Server 1* AddressPortTransport UDPOnly TCPpreferred DNSnaptr TCPonly TLS Expires (s)Register Yes NoRetry Timeout (ms)Retry Maximum CountLine Seize Timeout (s) *Server 2* AddressPortTransport UDPOnly TCPpreferred DNSnaptr TCPonly TLS Expires (s)Register Yes NoRetry Timeout (ms)Retry Maximum CountLine 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 AddressCallback Mode Registration Contact Disabled 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.****
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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.****
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So what do I do in the WEB config interface to enable NAT on this phone?* ***
** **
Thanks,****
Sean ****
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