<--- SIP read from 10.0.1.41:5060 ---> INVITE sip:3102@10.0.1.21 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-4d779f52 From: ;tag=be9c545d28cb6c1ao0 To: Call-ID: a50a6fd9-a559a6ce@10.0.1.41 CSeq: 101 INVITE Max-Forwards: 70 Contact: 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=- 19028 19028 IN IP4 10.0.1.41 s=- c=IN IP4 10.0.1.41 t=0 0 m=audio 16456 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:30 a=sendrecv <-------------> --- (13 headers 18 lines) --- Sending to 10.0.1.41 : 5060 (no NAT) Using INVITE request as basis request - a50a6fd9-a559a6ce@10.0.1.41 <--- Reliably Transmitting (no NAT) to 10.0.1.41:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-4d779f52;received=10.0.1.41 From: ;tag=be9c545d28cb6c1ao0 To: ;tag=as25711cfe Call-ID: a50a6fd9-a559a6ce@10.0.1.41 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e7017fd" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'a50a6fd9-a559a6ce@10.0.1.41' in 32000 ms (Method: INVITE) Found user '3101' akira*CLI> <--- SIP read from 10.0.1.41:5060 ---> ACK sip:3102@10.0.1.21 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-4d779f52 From: ;tag=be9c545d28cb6c1ao0 To: ;tag=as25711cfe Call-ID: a50a6fd9-a559a6ce@10.0.1.41 CSeq: 101 ACK Max-Forwards: 70 Contact: User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- akira*CLI> <--- SIP read from 10.0.1.41:5060 ---> INVITE sip:3102@10.0.1.21 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-bfdeb27 From: ;tag=be9c545d28cb6c1ao0 To: Call-ID: a50a6fd9-a559a6ce@10.0.1.41 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="3101",realm="asterisk",nonce="5e7017fd",uri="sip:3102@10.0.1.21",algorithm=MD5,response="f803cbb941ef4649a92794ca5cd36433" Contact: 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=- 19028 19028 IN IP4 10.0.1.41 s=- c=IN IP4 10.0.1.41 t=0 0 m=audio 16456 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:30 a=sendrecv <-------------> --- (14 headers 18 lines) --- Sending to 10.0.1.41 : 5060 (no NAT) Using INVITE request as basis request - a50a6fd9-a559a6ce@10.0.1.41 Found user '3101' 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 10.0.1.41:16456 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 - 0x8 (alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.1.41:16456 Looking for 3102 in handsets (domain 10.0.1.21) <--- Reliably Transmitting (no NAT) to 10.0.1.41:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-bfdeb27;received=10.0.1.41 From: ;tag=be9c545d28cb6c1ao0 To: ;tag=as25711cfe Call-ID: a50a6fd9-a559a6ce@10.0.1.41 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'a50a6fd9-a559a6ce@10.0.1.41' in 32000 ms (Method: INVITE) akira*CLI> <--- SIP read from 10.0.1.41:5060 ---> ACK sip:3102@10.0.1.21 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-bfdeb27 From: ;tag=be9c545d28cb6c1ao0 To: ;tag=as25711cfe Call-ID: a50a6fd9-a559a6ce@10.0.1.41 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="3101",realm="asterisk",nonce="5e7017fd",uri="sip:3102@10.0.1.21",algorithm=MD5,response="c6b9190d5973410f96f33b5ef23f87df" Contact: User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog 'a50a6fd9-a559a6ce@10.0.1.41' Method: ACK akira*CLI>