vicidial.org

Hey guys having an issue with setting up incoming calls.
I have configured as per the sample manual but changed calls to go to a phone, regardless if I use exact same setup as manual I still have same issue. When I call the DID I get "All Circuits are Busy Now Please Try Your Call Again Later"
Any help would be appreciated.

Vicidial Version: 2.12
Asterisk: 1.4.39.1-vici
Running on CentOS6

This is trunk:
register => <username>:<password>@<HOST IP>

[test]
disallow=all
allow=ulaw
allow=alaw
type=friend
host=<HOST IP>
fromdomain=<HOST IP>
username=<username>
secret=<password>
dtmfmode=rfc2833
context=trunkinbound

TRUNK9 = SIP/test

exten => _0NNXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _0NNXXXXXXX.,2,Dial(${TRUNK9}/${EXTEN:0},55,tToR)
exten => _0NNXXXXXXX.,3,Hangup

exten => _NXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _NXXXXXXX,2,Dial(${TRUNK9}/07${EXTEN},55,tToR)
exten => _NXXXXXXX,3,Hangup



This is logs from debug:
<--- SIP read from <HOST IP>:5060 --->
INVITE sip:s@<server external IP> SIP/2.0
Via: SIP/2.0/UDP <HOST IP>:5060;branch=z9hG4bK-d8754z-e1ac007d592ddd65-1---d8754z-;rport
Via: SIP/2.0/UDP <HOST IP>:5061;branch=z9hG4bK-ag5uljnvt4mygcng;rport=5061
Max-Forwards: 69
Record-Route: <sip:<HOST IP>;lr>
Contact: "Anonymous"<sip:<HOST IP>:5061>
To: <sip:61284883252@<HOST IP>>
From: <sip:0280739749@<HOST IP>>;tag=kvtkyc6po3m4q6fm.o
Call-ID: 248a85fc41bed5cf20f31f1c54459768@<server external IP>:5060
CSeq: 363 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
Content-Length: 252
cisco-GUID: 3413759056-1545867749-2387673119-688564542
h323-conf-id: 3413759056-1545867749-2387673119-688564542

v=0
o=Sippy 207013132 0 IN IP4 <HOST IP>
s=Asterisk PBX 11.12.0
t=0 0
m=audio 45840 RTP/AVP 8 0 101
c=IN IP4 110.93.22.41
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
--- (17 headers 12 lines) ---
Sending to <HOST IP> : 5060 (NAT)
Using INVITE request as basis request - 248a85fc41bed5cf20f31f1c54459768@<server external IP>:5060
Found peer 'test'

<--- Reliably Transmitting (NAT) to <HOST IP>:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP <HOST IP>:5060;branch=z9hG4bK-d8754z-e1ac007d592ddd65-1---d8754z-;received=<HOST IP>;rport=5060
Via: SIP/2.0/UDP <HOST IP>:5061;branch=z9hG4bK-ag5uljnvt4mygcng;rport=5061
From: <sip:0280739749@<HOST IP>>;tag=kvtkyc6po3m4q6fm.o
To: <sip:61284883252@<HOST IP>>;tag=as723d5f9b
Call-ID: 248a85fc41bed5cf20f31f1c54459768@<server external IP>:5060
CSeq: 363 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4339c35b"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '248a85fc41bed5cf20f31f1c54459768@<server external IP>:5060' in 6400 ms (Method: INVITE)

<--- SIP read from <HOST IP>:5060 --->
ACK sip:s@<server external IP> SIP/2.0
Via: SIP/2.0/UDP <HOST IP>:5060;branch=z9hG4bK-d8754z-e1ac007d592ddd65-1---d8754z-;rport
Max-Forwards: 70
To: <sip:61284883252@<HOST IP>>;tag=as723d5f9b
From: <sip:0280739749@<HOST IP>>;tag=kvtkyc6po3m4q6fm.o
Call-ID: 248a85fc41bed5cf20f31f1c54459768@<CALLER IP>:5060
CSeq: 363 ACK
Content-Length: 0

Hello,
Have you setup this DID in vicidial->Inbound->DIDs? if yes, post here the settings