No extension found, called # is targeted as extension

Update: I found the link below that solved the problem in case anyone needs this. I had to create the extension_custom.conf file.
sysadminman.net/blog/2008/callce … reepbx-210

I can call out but not in, no ring. The trace shows it looks like it want to target a phone number instead of a extension 600. I noticed that for another trunk/ITSP it never shows the “Looking for” message. The odd thing is that they are setup the same way. I have an inbond trunk setup for each separately, one rings and one does not. Here is the peer settings and trace. I already got some help on another problem still in progress on this forum and hope someone can set me on the right path for this one since it will help with the second problem. Any help is greatly appreciated. Thanks!

I think these are the important lines (see more below):
[Feb 8 14:59:39] VERBOSE[2835] logger.c: Looking for 4081234567 in from-voipyourphone (domain 99.31.165.5)
[Feb 8 14:59:39] VERBOSE[2835] logger.c: <— Reliably Transmitting (NAT) to 216.143.130.36:5060 —>
SIP/2.0 404 Not Found

host=sip.voipyourphone.com
defaultip=sip.voipyourphone.com
fromdomain=sip.voipyourphone.com
defaultuser=myid
secret=xxxxxx
type=peer
authname=myid
fromuser=myid
allow=ulaw
canrenvite=no
context=from-voipyourphone
insecure=very
nat=yes
qualify=yes

[Feb 8 15:11:10] VERBOSE[2835] logger.c: — (14 headers 0 lines) —
[Feb 8 15:11:10] VERBOSE[2835] logger.c: Really destroying SIP dialog ‘6f1f69886573ed393a82af571fd12649@10.246.1.24’ Method: OPTIONS
[Feb 8 15:11:21] VERBOSE[2835] logger.c:
<— SIP read from UDP://216.143.130.36:5060 —>
INVITE sip:4081234567@99.31.165.5:5060 SIP/2.0
Record-Route: sip:216.143.130.36;lr=on;ftag=as3ceb0510
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKb33f.575f79b1.0
Via: SIP/2.0/UDP 216.143.130.112:5060;received=216.143.130.112;branch=z9hG4bK7dc78602;rport=5060
From: “San Jose CA” sip:4087654321@216.143.130.112;tag=as3ceb0510
To: sip:myid@216.143.130.36
Contact: sip:4087654321@216.143.130.112
Call-ID: 28087ea12edb900b74effcba58742e31@216.143.130.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Mon, 08 Feb 2010 21:49:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 26673 26673 IN IP4 216.143.130.112
s=session
c=IN IP4 216.143.130.112
b=CT:384
t=0 0
m=audio 11794 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 52646 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=sendrecv

<------------->
[Feb 8 15:11:21] VERBOSE[2835] logger.c: — (16 headers 16 lines) —
[Feb 8 15:11:21] VERBOSE[2835] logger.c: == Using SIP RTP TOS bits 184
[Feb 8 15:11:21] VERBOSE[2835] logger.c: == Using SIP RTP CoS mark 5
[Feb 8 15:11:21] VERBOSE[2835] logger.c: == Using SIP VRTP TOS bits 136
[Feb 8 15:11:21] VERBOSE[2835] logger.c: == Using SIP VRTP CoS mark 6
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Sending to 216.143.130.36 : 5060 (NAT)
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Using INVITE request as basis request - 28087ea12edb900b74effcba58742e31@216.143.130.112
[Feb 8 15:11:21] VERBOSE[2835] logger.c: No user ‘4087654321’ in SIP users list
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found peer ‘myid’ for ‘4087654321’ from 216.143.130.36:5060
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found RTP audio format 0
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found RTP audio format 8
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found audio description format PCMU for ID 0
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found audio description format PCMA for ID 8
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found RTP video format 31
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found RTP video format 34
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found video description format H261 for ID 31
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found video description format H263 for ID 34
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0xc0000 (h261|h263)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Peer audio RTP is at port 216.143.130.112:11794
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Looking for 4081234567 in from-voipyourphone (domain 99.31.165.5)
[Feb 8 15:11:21] VERBOSE[2835] logger.c:
<— Reliably Transmitting (NAT) to 216.143.130.36:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKb33f.575f79b1.0;received=216.143.130.36
Via: SIP/2.0/UDP 216.143.130.112:5060;received=216.143.130.112;branch=z9hG4bK7dc78602;rport=5060
From: “San Jose CA” sip:4087654321@216.143.130.112;tag=as3ceb0510
To: sip:myid@216.143.130.36;tag=as622c0618
Call-ID: 28087ea12edb900b74effcba58742e31@216.143.130.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
[Feb 8 15:11:21] NOTICE[2835] chan_sip.c: Call from ‘myid’ to extension ‘4081234567’ rejected because extension not found.
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Scheduling destruction of SIP dialog ‘28087ea12edb900b74effcba58742e31@216.143.130.112’ in 6400 ms (Method: INVITE)
[Feb 8 15:11:21] VERBOSE[2835] logger.c:
<— SIP read from UDP://216.143.130.36:5060 —>
ACK sip:4081234567@99.31.165.5:5060 SIP/2.0
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKb33f.575f79b1.0
From: “San Jose CA” sip:4087654321@216.143.130.112;tag=as3ceb0510
Call-ID: 28087ea12edb900b74effcba58742e31@216.143.130.112
To: sip:myid@216.143.130.36;tag=as622c0618
CSeq: 102 ACK
Max-Forwards: 70
User-Agent: OpenSIPS (1.6.1-notls (i386/freebsd))
Content-Length: 0