Inbound help on Connecting Vicidial to FreePBX

I have setup a connection between FreePBX and Vicidial, I can call from FreePBX to Vicidial with not but I can’t get inbound calls from Vicidial to FreePBX. I have included my SIP Debug from both Vicidial and FreePBX. Any help on getting the inbound issue resolved out be appreciated.

User Agent: FPBX-2.8.1(1.8.5.0)
SDP Session Name: Asterisk PBX 1.8.5.0

VICIDIAL TRUNK

[FREEPBX]
disallow=all
allow=ulaw
type=friend
host=192.168.3.204
dtmfmode=rfc2833
insecure=very
context=trunkinbound

GLOBAL STRING
FREEPBXTRUNK = SIP/FREEPBX

DIALPLAN ENTRY
exten => _5NXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _5NXX,2,Dial(${FREEPBXTRUNK}/${EXTEN:1},tTor)
exten => _5NXX,3,Hangup

FREEPBX TRUNK

TRUNK NAME: vici1-fpbx

PEER DETAILS
host=192.168.3.10
type=friend
dtmfmode=rfc2833
allow=ulaw
qualify=yes

USER SETTINGS
host=192.168.3.10

SIP DEBUG INFORMATION FROM FREEPBX

<— SIP read from UDP:192.168.3.10:5060 —>
INVITE sip:[email protected];cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK5b720ce7;rport
From: “Remote 1” sip:[email protected];tag=as778dcf84
To: sip:[email protected];cpd=on
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “Remote 1” sip:[email protected];privacy=off;screen=no
Date: Tue, 20 Dec 2011 03:23:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 6331 6331 IN IP4 192.168.3.10
s=session
c=IN IP4 192.168.3.10
t=0 0
m=audio 17940 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (15 headers 12 lines) —
Sending to 192.168.3.10:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘1001’ for ‘1001’ from 192.168.3.10:5060

<— Reliably Transmitting (NAT) to 192.168.3.10:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK5b720ce7;received=192.168.3.10;rport=5060
From: “Remote 1” sip:[email protected];tag=as778dcf84
To: sip:[email protected];cpd=on;tag=as69dc501f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="1e500d2e"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.3.10:5060 —>
ACK sip:[email protected];cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.3.10:5060;branch=z9hG4bK5b720ce7;rport
From: “Remote 1” sip:[email protected];tag=as778dcf84
To: sip:[email protected];cpd=on;tag=as69dc501f
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “Remote 1” sip:[email protected];privacy=off;screen=no
Content-Length: 0

SIP DEBUG FROM VICIDIAL

<------------->
[Dec 19 22:24:42] — (13 headers 14 lines) —
[Dec 19 22:24:42] Sending to 10.248.33.9 : 22706 (NAT)
[Dec 19 22:24:42] Using INVITE request as basis request - MTk5YTkxODg4NDc5NTAwYmUxOThkZDE5OGE2ZGU2ZTY.
[Dec 19 22:24:42] Found user ‘1001’
[Dec 19 22:24:42] Found RTP audio format 107
[Dec 19 22:24:42] Found RTP audio format 0
[Dec 19 22:24:42] Found RTP audio format 8
[Dec 19 22:24:42] Found RTP audio format 101
[Dec 19 22:24:42] Peer audio RTP is at port 10.248.33.9:9848
[Dec 19 22:24:42] Found unknown media description format BV32 for ID 107
[Dec 19 22:24:42] Found audio description format telephone-event for ID 101
[Dec 19 22:24:42] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Dec 19 22:24:42] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Dec 19 22:24:42] Peer audio RTP is at port 10.248.33.9:9848
[Dec 19 22:24:42] Looking for 5399 in default (domain 192.168.3.10)
[Dec 19 22:24:42] list_route: hop: sip:[email protected]:22706
[Dec 19 22:24:42]
<— Transmitting (NAT) to 10.248.33.9:22706 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.248.33.9:22706;branch=z9hG4bK-d8754z-db347500d300552b-1—d8754z-;received=10.248.33.9;rport=22706
From: "Derrick Shand"sip:[email protected];tag=c44d6067
To: "5399"sip:[email protected]
Call-ID: MTk5YTkxODg4NDc5NTAwYmUxOThkZDE5OGE2ZGU2ZTY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
[Dec 19 22:24:42] – Executing [[email protected]:1] AGI(“SIP/1001-b2618c10”, “agi://127.0.0.1:4577/call_log”) in new stack
[Dec 19 22:24:42] – AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 19 22:24:42] – Executing [[email protected]:2] Dial(“SIP/1001-b2618c10”, “SIP/FREEPBX/399||tTor”) in new stack
[Dec 19 22:24:42] – Called FREEPBX/399
[Dec 19 22:24:42]
<— Transmitting (NAT) to 10.248.33.9:22706 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.248.33.9:22706;branch=z9hG4bK-d8754z-db347500d300552b-1—d8754z-;received=10.248.33.9;rport=22706
From: "Derrick Shand"sip:[email protected];tag=c44d6067
To: "5399"sip:[email protected];tag=as58b70786
Call-ID: MTk5YTkxODg4NDc5NTAwYmUxOThkZDE5OGE2ZGU2ZTY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
[Dec 19 22:24:42] NOTICE[6402]: chan_sip.c:12358 handle_response_invite: Failed to authenticate on INVITE to ‘“Remote 1” sip:[email protected];tag=as57f6a81a’
[Dec 19 22:24:42] – SIP/FREEPBX-0832aa30 is circuit-busy
[Dec 19 22:24:42] == Everyone is busy/congested at this time (1:0/1/0)
[Dec 19 22:24:42] – Executing [[email protected]:3] Hangup(“SIP/1001-b2618c10”, “”) in new stack
[Dec 19 22:24:42] == Spawn extension (default, 5399, 3) exited non-zero on ‘SIP/1001-b2618c10’
[Dec 19 22:24:42] – Executing [[email protected]:1] DeadAGI(“SIP/1001-b2618c10”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----21-----CONGESTION----------”) in new stack
[Dec 19 22:24:42] – AGI Script agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----21-----CONGESTION---------- completed, returning 0
[Dec 19 22:24:42] Scheduling destruction of SIP dialog ‘MTk5YTkxODg4NDc5NTAwYmUxOThkZDE5OGE2ZGU2ZTY.’ in 32000 ms (Method: INVITE)
[Dec 19 22:24:42]
<— Reliably Transmitting (NAT) to 10.248.33.9:22706 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.248.33.9:22706;branch=z9hG4bK-d8754z-db347500d300552b-1—d8754z-;received=10.248.33.9;rport=22706
From: "Derrick Shand"sip:[email protected];tag=c44d6067
To: "5399"sip:[email protected];tag=as58b70786
Call-ID: MTk5YTkxODg4NDc5NTAwYmUxOThkZDE5OGE2ZGU2ZTY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
[Dec 19 22:24:42]
<— SIP read from 10.248.33.9:22706 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.248.33.9:22706;branch=z9hG4bK-d8754z-db347500d300552b-1—d8754z-;rport
To: "5399"sip:[email protected];tag=as58b70786
From: "Derrick Shand"sip:[email protected];tag=c44d6067
Call-ID: MTk5YTkxODg4NDc5NTAwYmUxOThkZDE5OGE2ZGU2ZTY.
CSeq: 2 ACK
Content-Length: 0

<------------->
[Dec 19 22:24:42] — (7 headers 0 lines) —