406 SIP Error - when dialing between phones

Okay, so I’m using (4) Linksys SPA-941’s within the office.
I can dial between 3 of the phones. However, it seems one of the phones I’m not able to dial directly. I can receive calls that come in the queue, on that phone. Just can’t get direct dial calls.

extensions.conf:

[quote][internal]
exten => 100,1,Dial(SIP/100)
exten => 105,1,Dial(SIP/105)
exten => 110,1,Dial(SIP/110)
exten => 115,1,Dial(SIP/115)
exten => 120,1,Dial(SIP/120)
exten => 125,1,Dial(SIP/125)
exten => 200,1,Dial(SIP/200)
exten => 205,1,Dial(SIP/205)
exten => 210,1,Dial(SIP/210)
exten => 215,1,Dial(SIP/215)
exten => 220,1,Dial(SIP/220)[/quote]

sip.conf:

[quote]
[105]
type=friend
context=internal
callerid=105
host=dynamic
secret=105password
canreinvite=no
insecure=port,invite
allow=all
nat=yes
expire=120

[110]
type=friend
context=internal
callerid=110
host=dynamic
secret=110password
canreinvite=no
insecure=port,invite
allow=all
nat=yes
expire=120[/quote]

Here’s the SIP Debug for one of these such calls:

[quote]<— SIP read from office.hostname.com:1027 —>
ACK sip:105@pbx.hostname.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.168:5060;branch=z9hG4bK-66121742
From: “Anonymous” sip:115@pbx.hostname.com:5060;tag=308d60fa42ea01c8o0
To: “Tech Support” sip:105@pbx.hostname.com:5060;tag=as2837a6a2
Call-ID: c97a4e62-f84fb380@localhost
CSeq: 101 ACK
Max-Forwards: 70
Contact: “Anonymous” sip:115@192.168.15.168:5060
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 0

<------------->
— (10 headers 0 lines) —
voip*CLI>
<— SIP read from office.hostname.com:1027 —>
INVITE sip:105@pbx.hostname.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.168:5060;branch=z9hG4bK-481c4b6
From: “Anonymous” sip:115@pbx.hostname.com:5060;tag=308d60fa42ea01c8o0
To: “Tech Support” sip:105@pbx.hostname.com:5060
Call-ID: c97a4e62-f84fb380@localhost
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username=“115”,realm=“asterisk”,nonce=“123f0139”,uri="sip:105@pbx.hostname.com",algorithm=MD5,response="f5cf541d701eac94e4153e148ffa1faf"
Contact: “Anonymous” sip:115@192.168.15.168:5060
Expires: 240
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 393
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 19451 19451 IN IP4 office.hostname.com
s=-
c=IN IP4 office.hostname.com
t=0 0
m=audio 16410 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (15 headers 18 lines) —
Sending to office.hostname.com : 1027 (NAT)
Using INVITE request as basis request - c97a4e62-f84fb380@localhost
Found user '115’
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port office.hostname.com:16410
Found description format PCMU for ID 0
Found description format G726-32 for ID 2
Found description format G723 for ID 4
Found description format PCMA for ID 8
Found description format G729a for ID 18
Found description format G726-40 for ID 96
Found description format G726-24 for ID 97
Found description format G726-16 for ID 98
Found description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xd0d (g723|ulaw|alaw|g726|g729|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port office.hostname.com:16410
Looking for 105 in internal (domain pbx.hostname.com)
list_route: hop: sip:115@192.168.15.168:5060

<— Transmitting (NAT) to office.hostname.com:1027 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.15.168:5060;branch=z9hG4bK-481c4b6;received=office.hostname.com
From: “Anonymous” sip:115@pbx.hostname.com:5060;tag=308d60fa42ea01c8o0
To: “Tech Support” sip:105@pbx.hostname.com:5060
Call-ID: c97a4e62-f84fb380@localhost
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:105@pbx.hostname.com
Content-Length: 0

<------------>
– Executing [105@internal:1] Dial(“SIP/115-091074e0”, “SIP/105”) in new stack
Audio is at pbx.hostname.com port 14002
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to office.hostname.com:1025:
INVITE sip:105@192.168.15.165:5060 SIP/2.0
Via: SIP/2.0/UDP pbx.hostname.com:5060;branch=z9hG4bK0ef46a03;rport
From: “Anonymous” sip:115@pbx.hostname.com;tag=as27f0b393
To: sip:105@192.168.15.165:5060
Contact: sip:115@pbx.hostname.com
Call-ID: 74c7ad1d04ad76ea5ce3a4717e26be39@pbx.hostname.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 02 Jan 2008 16:50:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 471

v=0
o=root 12967 12967 IN IP4 pbx.hostname.com
s=session
c=IN IP4 pbx.hostname.com
t=0 0
m=audio 14002 RTP/AVP 0 3 8 112 5 10 7 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


-- Called 105

voip*CLI>
<— SIP read from office.hostname.com:1025 —>
SIP/2.0 100 Trying
To: sip:105@192.168.15.165:5060
From: “Anonymous” sip:115@pbx.hostname.com:5060;tag=as27f0b393
Call-ID: 74c7ad1d04ad76ea5ce3a4717e26be39@pbx.hostname.com
CSeq: 102 INVITE
Via: SIP/2.0/UDP pbx.hostname.com:5060;branch=z9hG4bK0ef46a03
Server: Linksys/SPA941-5.1.8
Content-Length: 0

<------------->
— (8 headers 0 lines) —
voip*CLI>
<— SIP read from office.hostname.com:1025 —>
SIP/2.0 406 Not Acceptable
To: sip:105@192.168.15.165:5060;tag=6ef5db2e5ce6633ci0
From: “Anonymous” sip:115@pbx.hostname.com:5060;tag=as27f0b393
Call-ID: 74c7ad1d04ad76ea5ce3a4717e26be39@pbx.hostname.com
CSeq: 102 INVITE
Via: SIP/2.0/UDP pbx.hostname.com:5060;branch=z9hG4bK0ef46a03
Server: Linksys/SPA941-5.1.8
Content-Length: 0

<------------->
— (8 headers 0 lines) —
– Got SIP response 406 “Not Acceptable” back from office.hostname.com
Transmitting (NAT) to office.hostname.com:1025:
ACK sip:105@192.168.15.165:5060 SIP/2.0
Via: SIP/2.0/UDP pbx.hostname.com:5060;branch=z9hG4bK0ef46a03;rport
From: “Anonymous” sip:115@pbx.hostname.com;tag=as27f0b393
To: sip:105@192.168.15.165:5060;tag=6ef5db2e5ce6633ci0
Contact: sip:115@pbx.hostname.com
Call-ID: 74c7ad1d04ad76ea5ce3a4717e26be39@pbx.hostname.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


-- No one is available to answer at this time (1:0/0/0)

== Auto fallthrough, channel ‘SIP/115-091074e0’ status is 'NOANSWER’
Scheduling destruction of SIP dialog ‘c97a4e62-f84fb380@localhost’ in 32000 ms (Method: INVITE)
voip*CLI>
<— Reliably Transmitting (NAT) to office.hostname.com:1027 —>
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.15.168:5060;branch=z9hG4bK-481c4b6;received=office.hostname.com
From: “Anonymous” sip:115@pbx.hostname.com:5060;tag=308d60fa42ea01c8o0
To: “Tech Support” sip:105@pbx.hostname.com:5060;tag=as171f283b
Call-ID: c97a4e62-f84fb380@localhost
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:105@pbx.hostname.com
Content-Length: 0

<------------>
voip*CLI>
<— SIP read from office.hostname.com:1027 —>
ACK sip:105@pbx.hostname.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.168:5060;branch=z9hG4bK-481c4b6
From: “Anonymous” sip:115@pbx.hostname.com:5060;tag=308d60fa42ea01c8o0
To: “Tech Support” sip:105@pbx.hostname.com:5060;tag=as171f283b
Call-ID: c97a4e62-f84fb380@localhost
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username=“115”,realm=“asterisk”,nonce=“123f0139”,uri="sip:105@pbx.hostname.com",algorithm=MD5,response="f5cf541d701eac94e4153e148ffa1faf"
ontact: “Anonymous” sip:115@192.168.15.168:5060
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog '74c7ad1d04ad76ea5ce3a4717e26be39@pbx.hostname.com’ Method: INVITE
voip*CLI>
<— SIP read from office.hostname.com:1026 —>
NOTIFY sip:pbx.hostname.com SIP/2.0
Via: SIP/2.0/UDP 192.168.15.167:5060;branch=z9hG4bK-ffd9f051
From: “Tech Support” sip:110@pbx.hostname.com:5060;tag=c5dc1c96a14b693o0
To: sip:pbx.hostname.com
Call-ID: d79967c1-474088fb@192.168.15.167
CSeq: 21494 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— Transmitting (no NAT) to office.hostname.com:1026 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.15.167:5060;branch=z9hG4bK-ffd9f051;received=office.hostname.com
From: “Tech Support” sip:110@pbx.hostname.com:5060;tag=c5dc1c96a14b693o0
To: sip:pbx.hostname.com;tag=as5d4581d6
Call-ID: d79967c1-474088fb@192.168.15.167
CSeq: 21494 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>[/quote]

My head is spinning on this one. Thanks for your help!!