SIP Debugging Enabled for IP: 10.1.7.167 *CLI> <--- SIP read from 10.1.7.167:5060 ---> INVITE sip:1000@10.1.7.17 SIP/2.0 Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-7de3e120 From: "Hello Worldz" ;tag=4ae444e3fb7597e0o0 To: Call-ID: ed5a4f5f-4684af94@10.1.7.167 CSeq: 101 INVITE Max-Forwards: 70 Contact: "Hello Worldz" Expires: 240 User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 387 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 7400 7400 IN IP4 10.1.7.167 s=- c=IN IP4 10.1.7.167 t=0 0 m=audio 16424 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (13 headers 18 lines) --- == Using TOS bits 0 == Using CoS mark 0 Sending to 10.1.7.167 : 5060 (no NAT) Using INVITE request as basis request - ed5a4f5f-4684af94@10.1.7.167 Found user '777' for '777' <--- Reliably Transmitting (no NAT) to 10.1.7.167:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-7de3e120;received=10.1.7.167 From: "Hello Worldz" ;tag=4ae444e3fb7597e0o0 To: ;tag=as61e1ebea Call-ID: ed5a4f5f-4684af94@10.1.7.167 CSeq: 101 INVITE User-Agent: Asterisk PBX SVN-trunk-r63330 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6efe7839" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'ed5a4f5f-4684af94@10.1.7.167' in 32000 ms (Method: INVITE) <--- SIP read from 10.1.7.167:5060 ---> ACK sip:1000@10.1.7.17 SIP/2.0 Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-7de3e120 From: "Hello Worldz" ;tag=4ae444e3fb7597e0o0 To: ;tag=as61e1ebea Call-ID: ed5a4f5f-4684af94@10.1.7.167 CSeq: 101 ACK Max-Forwards: 70 Contact: "Hello Worldz" User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.1.7.167:5060 ---> INVITE sip:1000@10.1.7.17 SIP/2.0 Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-395b3d35 From: "Hello Worldz" ;tag=4ae444e3fb7597e0o0 To: Call-ID: ed5a4f5f-4684af94@10.1.7.167 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="777",realm="asterisk",nonce="6efe7839",uri="sip:1000@10.1.7.17",algorithm=MD5,response="2057a3ffbe7a6e1e079b13e410801584" Contact: "Hello Worldz" Expires: 240 User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 387 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 7400 7400 IN IP4 10.1.7.167 s=- c=IN IP4 10.1.7.167 t=0 0 m=audio 16424 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (14 headers 18 lines) --- Sending to 10.1.7.167 : 5060 (no NAT) Using INVITE request as basis request - ed5a4f5f-4684af94@10.1.7.167 Found user '777' for '777' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port 10.1.7.167:16424 Found description format PCMU for ID 0 Found description format G726-32 for ID 2 Found description format G723 for ID 4 Found description format PCMA for ID 8 Found description format G729a for ID 18 Found description format G726-40 for ID 96 Found description format G726-24 for ID 97 Found description format G726-16 for ID 98 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.1.7.167:16424 Looking for 1000 in users (domain 10.1.7.17) <--- Reliably Transmitting (no NAT) to 10.1.7.167:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-395b3d35;received=10.1.7.167 From: "Hello Worldz" ;tag=4ae444e3fb7597e0o0 To: ;tag=as61e1ebea Call-ID: ed5a4f5f-4684af94@10.1.7.167 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r63330 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'ed5a4f5f-4684af94@10.1.7.167' in 32000 ms (Method: INVITE) <--- SIP read from 10.1.7.167:5060 ---> ACK sip:1000@10.1.7.17 SIP/2.0 Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-395b3d35 From: "Hello Worldz" ;tag=4ae444e3fb7597e0o0 To: ;tag=as61e1ebea Call-ID: ed5a4f5f-4684af94@10.1.7.167 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="777",realm="asterisk",nonce="6efe7839",uri="sip:1000@10.1.7.17",algorithm=MD5,response="a627bdd795a20a1dee5a3a2e1cea6286" Contact: "Hello Worldz" User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 0 <-------------> --- (11 headers 0 lines) ---