Registration fails: no attempt at Reliably Transmitting (NAT

Asterisk Asterisk Support
1

A potential customer has phones deployed behind a Sonicwall NSA 3500. The on-site test phones we have deployed are not able to register. Sonicwalls are known to be a pain but customer already has VoIP service working through it with another provider so it should work.

We have phones in our lab registering successfully with this system. The extensions are all configured with NAT=yes. All phones use the same template.

The sip debug output below shows what is happening. Further below is sip debug output for a phone registering successfully from our lab.

What I don’t understand is: Why doesn’t Asterisk attempt to transmit with the "Reliably Transmitting (NAT)” method like it does with extensions registering from other sites?

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Failed registration dialog:

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<--- SIP read from UDP:x.x.x.x:25416 --->
REGISTER sip:provider.net:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.98:5060;branch=z9hG4bK75c58b2b6BD5EEA2
From: "test-office450" <sip:206@provider.net>;tag=8FA0FD5E-FCEE60D9
To: <sip:206@provider.net>
CSeq: 1 REGISTER
Call-ID: 2a4708cd-30d0e39c-a8c823f@10.0.3.98
Contact: <sip:206@10.0.3.98:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.4.2906
Accept-Language: en
Max-Forwards: 70
Expires: 3600
Content-Length: 0



<------------->
--- (12 headers 0 lines) ---
Sending to x.x.x.x:5060 (no NAT)
Sending to x.x.x.x:5060 (no NAT)



<--- Transmitting (no NAT) to x.x.x.x:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.3.98:5060;branch=z9hG4bK75c58b2b6BD5EEA2;received=x.x.x.x
From: "test-office450" <sip:206@provider.net>;tag=8FA0FD5E-FCEE60D9
To: <sip:206@provider.net>;tag=as5ed61db0
Call-ID: 2a4708cd-30d0e39c-a8c823f@10.0.3.98
CSeq: 1 REGISTER
Server: FPBX-2.11.0(11.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2a95f72e"
Content-Length: 0






<------------>
Scheduling destruction of SIP dialog '2a4708cd-30d0e39c-a8c823f@10.0.3.98' in 32000 ms (Method: REGISTER)



<--- SIP read from UDP:x.x.x.x:25416 --->
REGISTER sip:provider.net:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.98:5060;branch=z9hG4bK75c58b2b6BD5EEA2
From: "test-office450" <sip:206@provider.net>;tag=8FA0FD5E-FCEE60D9
To: <sip:206@provider.net>
CSeq: 1 REGISTER
Call-ID: 2a4708cd-30d0e39c-a8c823f@10.0.3.98
Contact: <sip:206@10.0.3.98:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.4.2906
Accept-Language: en
Max-Forwards: 70
Expires: 3600
Content-Length: 0



<------------->
--- (12 headers 0 lines) ---
Sending to x.x.x.x:5060 (no NAT)



<--- Transmitting (no NAT) to x.x.x.x:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.3.98:5060;branch=z9hG4bK75c58b2b6BD5EEA2;received=x.x.x.x
From: "test-office450" <sip:206@provider.net>;tag=8FA0FD5E-FCEE60D9
To: <sip:206@provider.net>;tag=as5ed61db0
Call-ID: 2a4708cd-30d0e39c-a8c823f@10.0.3.98
CSeq: 1 REGISTER
Server: FPBX-2.11.0(11.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2a95f72e"
Content-Length: 0


###################################
Successful registration dialog:
###################################



<--- SIP read from UDP:y.y.y.y:5060 --->
REGISTER sip:provider.net:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.200.113:5060;branch=z9hG4bK6a287d70CD455F51
From: "test-offsite" <sip:204@provider.net>;tag=B8ABDC3A-A673E963
To: <sip:204@provider.net>
CSeq: 25 REGISTER
Call-ID: 1c76d84d-6dbb5f32-97b0f1bb@172.16.200.113
Contact: <sip:204@172.16.200.113:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.2.11307
Accept-Language: en
Authorization: Digest username="204", realm="asterisk", nonce="6d9b59ba", uri="sip:provider.net:5060", response="c92d8f0e01672f1714e60532d78e6e6b", algorithm=MD5
Max-Forwards: 70
Expires: 60
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Sending to y.y.y.y:5060 (no NAT)
[2014-11-06 12:54:29] NOTICE[1950]: chan_sip.c:16417 check_auth: Correct auth, but based on stale nonce received from '"test-offsite" <sip:204@provider.net>;tag=B8ABDC3A-A673E963'


<--- Transmitting (no NAT) to y.y.y.y:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.200.113:5060;branch=z9hG4bK6a287d70CD455F51;received=y.y.y.y
From: "test-offsite" <sip:204@provider.net>;tag=B8ABDC3A-A673E963
To: <sip:204@provider.net>;tag=as0364bf42
Call-ID: 1c76d84d-6dbb5f32-97b0f1bb@172.16.200.113
CSeq: 25 REGISTER
Server: FPBX-2.11.0(11.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="095c2aca", stale=true
Content-Length: 0




<------------>
Scheduling destruction of SIP dialog '1c76d84d-6dbb5f32-97b0f1bb@172.16.200.113' in 32000 ms (Method: REGISTER)


<--- SIP read from UDP:y.y.y.y:5060 --->
REGISTER sip:provider.net:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.200.113:5060;branch=z9hG4bK9ca62d86A326759F
From: "test-offsite" <sip:204@provider.net>;tag=B8ABDC3A-A673E963
To: <sip:204@provider.net>
CSeq: 26 REGISTER
Call-ID: 1c76d84d-6dbb5f32-97b0f1bb@172.16.200.113
Contact: <sip:204@172.16.200.113:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.2.11307
Accept-Language: en
Authorization: Digest username="204", realm="asterisk", nonce="095c2aca", uri="sip:provider.net:5060", response="4a266e91fd2297441661de7ae1719fd2", algorithm=MD5
Max-Forwards: 70
Expires: 60
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Sending to y.y.y.y:5060 (no NAT)
Reliably Transmitting (NAT) to y.y.y.y:5060:
OPTIONS sip:204@172.16.200.113:5060 SIP/2.0
Via: SIP/2.0/UDP 174.140.227.99:5060;branch=z9hG4bK38d57175;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@174.140.227.99>;tag=as7b8d7907
To: <sip:204@172.16.200.113:5060>
Contact: <sip:Unknown@174.140.227.99:5060>
Call-ID: 7d3e11785e45650b6235f1412b305a51@174.140.227.99:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.5.1)
Date: Thu, 06 Nov 2014 20:54:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
2

The phones are not configured for NAT. They should he sending x.x.x.x in the Contact header.

I would try setting force_port explicitly (yes is deprecated).

There does seem to be something wrong the the port number that Asteirsk is replying to, assuming that x.x.x.x is not the 10/24 address in request.