Asterisk-1.4.4 Outbound Call Issues using Eyebeam 1.5.10.2

Have Asterisk-1.4.4 loaded am able to use SIP to check voice mail with Eyebeam 1.5.10.2, but unable to place outbound calls, have SIP trace from Asterisk.

Presently get “Call Failed,Not Found” again call does route to voice mail…

Newbie to Asterisk any help is appreciated…

— SIP read from 192.168.0.7:5060 —>
INVITE sip:917771234567@linux-robinsons SIP/2.0
Via: SIP/2.0/UDP 192.168.0.7:5060;branch=z9hG4bK-d87543-a1116638f1754b67-1–d87543-;rport
Max-Forwards: 70
Contact: sip:6000@192.168.0.7:5060
To: "917771234567"sip:917771234567@linux-robinsons
From: "6000"sip:6000@linux-robinsons;tag=8e5b7327
Call-ID: NGZlNDdlYmNkN2UzMTRiZWYxNTJhYTU5ZjRmOTViY2U.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1006e stamp 33793
Content-Length: 356

v=0
o=- 8 2 IN IP4 192.168.0.7
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.0.7
t=0 0
m=audio 5062 RTP/AVP 98 18 3 101
a=alt:1 1 : 7+QO+Nhq 0uhNpfi1 192.168.0.7 5062
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:98 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:93D4CF3C4FFD4EC98967FAB8E37A3582

<------------->
— (12 headers 14 lines) —
Sending to 192.168.0.7 : 5060 (NAT)
Using INVITE request as basis request - NGZlNDdlYmNkN2UzMTRiZWYxNTJhYTU5ZjRmOTViY2U.
Found no matching peer or user for '192.168.0.7:5060’
Found RTP audio format 98
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.7:5062
Found description format iLBC for ID 98
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x506 (gsm|ulaw|g729|ilbc), peer - audio=0x502 (gsm|g729|ilbc)/video=0x0 (nothing), combined - 0x502 (gsm|g729|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.7:5062
Looking for 917771234567 in default (domain linux-robinsons)

<— Reliably Transmitting (NAT) to 192.168.0.7:5060 —>
SIP/2.0 404 Not Found
v: SIP/2.0/UDP 192.168.0.7:5060;branch=z9hG4bK-d87543-a1116638f1754b67-1–d87543-;received=192.168.0.7;rport=5060
f: "6000"sip:6000@linux-robinsons;tag=8e5b7327
t: "917771234567"sip:917771234567@linux-robinsons;tag=as4e30fee2
i: NGZlNDdlYmNkN2UzMTRiZWYxNTJhYTU5ZjRmOTViY2U.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0

<------------>
Scheduling destruction of SIP dialog ‘NGZlNDdlYmNkN2UzMTRiZWYxNTJhYTU5ZjRmOTViY2U.’ in 32000 ms (Method: INVITE)
linux-robinsons*CLI>
<— SIP read from 192.168.0.7:5060 —>
ACK sip:917771234567@linux-robinsons SIP/2.0
Via: SIP/2.0/UDP 192.168.0.7:5060;branch=z9hG4bK-d87543-a1116638f1754b67-1–d87543-;rport
To: "917771234567"sip:917771234567@linux-robinsons;tag=as4e30fee2
From: "6000"sip:6000@linux-robinsons;tag=8e5b7327
Call-ID: NGZlNDdlYmNkN2UzMTRiZWYxNTJhYTU5ZjRmOTViY2U.
CSeq: 1 ACK
Content-Length: 0