Looks like your endpoint don't dialing nothing. It can be a problem. I think you need to dial some number on your endpoint.
Also please post sip.conf here
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Mike_TK
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*CLI> sip debug
SIP Debugging Enabled
*CLI>
Sip read:
INVITE sip:10.0.0.50 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.198:5060;branch=z9hG4bK3696536193
From: <sip:anonymous@10.0.0.198>;tag=1408747987
To: test <sip:10.0.0.50>
Call-ID: 364657556@10.0.0.198
CSeq: 20 INVITE
Contact: <sip:anonymous@10.0.0.198>
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Type: application/sdp
Content-Length: 353
v=0
o=anonymous 123456 654321 IN IP4 10.0.0.198
s=A conversation
c=IN IP4 10.0.0.198
t=0 0
m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
b=AS:20
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:111 speex/16000/1
a=rtpmap:115 1015/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
11 headers, 15 lines
Using latest request as basis request
Sending to 10.0.0.198 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 115
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.198:7078
Found description format PCMU
Found description format GSM
Found description format PCMA
Found description format speex
Found description format speex
Found description format 1015
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Found no matching peer or user for '10.0.0.198:5060'
Looking for 10.0.0.50 in default
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.198:5060;branch=z9hG4bK3696536193
From: <sip:anonymous@10.0.0.198>;tag=1408747987
To: test <sip:10.0.0.50>;tag=as296ed96a
Call-ID: 364657556@10.0.0.198
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:10.0.0.50@10.0.0.50>
Content-Length: 0
to 10.0.0.198:5060
Sip read:
ACK sip:10.0.0.50 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.198:5060;branch=z9hG4bK3696536193
From: <sip:anonymous@10.0.0.198>;tag=1408747987
To: test <sip:10.0.0.50>;tag=as296ed96a
Call-ID: 364657556@10.0.0.198
CSeq: 20 ACK
Content-Length: 0
7 headers, 0 lines
Destroying call '364657556@10.0.0.198