Strange Registration behavior

Sorry for posting so much lately - my problem is evolving and changing all at the same time…

I previously posted regarding an error I was getting (SIP 603), now that has changed.

Let me start from the beginning:

  1. Server is FPBX 2.10, fresh install
  2. Server is
  3. Asterisk version is
  4. EXT 201 is Cisco 7962G,
  5. EXT 202 is Cisco 6941,
  6. Phones are flashed with SIP firmware- 7962 is SIP42.9-3-1-1S 6941 is SIP69xx.9-3-1-5
  7. Phones both pull valid SEPMAC.conf.xml files
  8. EXT 202 registers fine
  9. EXT 201 fails registration (but still sortof works…?)
  10. I can call from 201 to 202, and talk both ways
  11. I can NOT call from 202 to 201, I get a fast busy

Here is a sip trace with core verbose = 255

<--- SIP read from TCP: ---> INVITE sip:*[email protected];user=phone SIP/2.0 Via: SIP/2.0/TCP;branch=z9hG4bKccea608c From: "201" ;tag=0024c4be6f06000261fae804-021a4d46 To: Call-ID: [email protected] Max-Forwards: 70 Date: Tue, 11 Sep 2012 05:32:16 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7962G/9.3.1 Contact: Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1 Allow-Events: kpml,dialog Content-Length: 350 Content-Type: application/sdp Content-Disposition: session;handling=optional

o=Cisco-SIPUA 21743 0 IN IP4
s=SIP Call
t=0 0
m=audio 19938 RTP/AVP 0 8 18 102 116 101
c=IN IP4
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
— (18 headers 16 lines) —
Sending to (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘201’ for ‘201’ from

<— Reliably Transmitting (NAT) to —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP;branch=z9hG4bKccea608c;received=;rport=52004
From: “201” sip:[email protected];tag=0024c4be6f06000261fae804-021a4d46
To: sip:*[email protected];tag=as478ced6e
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-2.10.1(
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6e0412d1"
Content-Length: 0

I have set NAT=Never on both extensions, and TCP is enabled. This verifies because the 6941 (EXT 202) works fine.

Can someone help me out here? If more info or debug is needed just tell me what to get and how to get it.

I am completely baffled by this!