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 _____ Mike_TK ------------------------------------------------ *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: ;tag=1408747987 To: test Call-ID: 364657556@10.0.0.198 CSeq: 20 INVITE Contact: 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: ;tag=1408747987 To: test ;tag=as296ed96a Call-ID: 364657556@10.0.0.198 CSeq: 20 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: 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: ;tag=1408747987 To: test ;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