asterisk*CLI> sip debug SIP Debugging enabled Asterisk Ready. <-- SIP read from 192.168.1.3:5060: INVITE sip:14@192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000054452883e60000464c000000e1 Content-Length: 335 Contact: Call-ID: 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3 Content-Type: application/sdp CSeq: 1 INVITE From: "Pablo";tag=16954548418643 Max-Forwards: 70 To: User-Agent: SJphone/1.60.289a (SJ Labs) v=0 o=- 3369271910 3369271910 IN IP4 192.168.1.3 s=SJphone c=IN IP4 192.168.1.3 t=0 0 a=direction:active m=audio 49158 RTP/AVP 3 97 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (11 headers 15 lines)--- Using INVITE request as basis request - 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3 Sending to 192.168.1.3 : 5060 (NAT) Reliably Transmitting (no NAT) to 192.168.1.3:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000054452883e60000464c000000e1;received=192.168.1.3 From: "Pablo";tag=16954548418643 To: ;tag=as2f733647 Call-ID: 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6253989a" Content-Length: 0 --- Scheduling destruction of call '0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3' in 15000 ms Found user 'ivan' asterisk*CLI> <-- SIP read from 192.168.1.3:5060: ACK sip:14@192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000054452883e60000464c000000e1 Content-Length: 0 Call-ID: 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3 CSeq: 1 ACK From: "Pablo";tag=16954548418643 Max-Forwards: 70 To: ;tag=as2f733647 User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.1.3:5060: INVITE sip:14@192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000054452883e700004005000000e2 Content-Length: 335 Contact: Call-ID: 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3 Content-Type: application/sdp CSeq: 2 INVITE From: "Pablo";tag=16954548418643 Max-Forwards: 70 To: User-Agent: SJphone/1.60.289a (SJ Labs) Proxy-Authorization: Digest username="ivan",realm="asterisk",nonce="6253989a",uri="sip:14@192.168.1.2",response="d24aa324b832cabec5c14996cd7b55a5",algorithm="MD5" v=0 o=- 3369271910 3369271910 IN IP4 192.168.1.3 s=SJphone c=IN IP4 192.168.1.3 t=0 0 a=direction:active m=audio 49158 RTP/AVP 3 97 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (12 headers 15 lines)--- Using INVITE request as basis request - 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3 Sending to 192.168.1.3 : 5060 (NAT) Found user 'ivan' Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.3:49158 Found description format GSM Found description format iLBC Found description format iLBC Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 14 in outbound-local (domain 192.168.1.2) Reliably Transmitting (no NAT) to 192.168.1.3:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000054452883e700004005000000e2;received=192.168.1.3 From: "Pablo";tag=16954548418643 To: ;tag=as2f733647 Call-ID: 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk*CLI> <-- SIP read from 192.168.1.3:5060: ACK sip:14@192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000054452883e700004005000000e2 Content-Length: 0 Call-ID: 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3 CSeq: 2 ACK From: "Pablo";tag=16954548418643 Max-Forwards: 70 To: ;tag=as2f733647 User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- Destroying call '0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3' asterisk*CLI> <-- SIP read from 192.168.1.3:5060: OPTIONS sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000010452883f100007f0c000000e4 Content-Length: 0 Call-ID: 4320915F-9E26-4BD7-ACBB-6173AA90A5B4@192.168.1.3 CSeq: 56 OPTIONS From: ;tag=16955590619931 Max-Forwards: 70 To: --- (8 headers 0 lines)--- Looking for s in default (domain 192.168.1.2) Transmitting (no NAT) to 192.168.1.3:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000010452883f100007f0c000000e4;received=192.168.1.3 From: ;tag=16955590619931 To: ;tag=as4932bcc4 Call-ID: 4320915F-9E26-4BD7-ACBB-6173AA90A5B4@192.168.1.3 CSeq: 56 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '4320915F-9E26-4BD7-ACBB-6173AA90A5B4@192.168.1.3' asterisk*CLI> sip no debug SIP Debugging Disabled asterisk*CLI>