--- (12 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 192.168.0.87:5060: INVITE sip:401@192.168.0.95 SIP/2.0 To: From: "402";tag=8d10bb3d Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-fc57b040ef2eb31f-1--d87543-;rport Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs CSeq: 1 INVITE Contact: Max-Forwards: 70 Content-Type: application/sdp User-Agent: Softphone2 Content-Length: 272 v=0 o=- 0 0 IN IP4 192.168.0.87 s= c=IN IP4 192.168.0.87 t=0 0 m=audio 5000 RTP/AVP 3 97 110 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 --- (11 headers 13 lines)--- Using INVITE request as basis request - 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs Sending to 192.168.0.87 : 5060 (NAT) Reliably Transmitting (no NAT) to 192.168.0.87:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-fc57b040ef2eb31f-1--d87543-;rport;received=192.168.0.87 From: "402";tag=8d10bb3d To: ;tag=as4607b9c8 Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs 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="4b51d48c" Content-Length: 0 --- Scheduling destruction of call '8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs' in 15000 ms Found user '402' asterisk1*CLI> <-- SIP read from 192.168.0.87:5060: ACK sip:401@192.168.0.95 SIP/2.0 To: ;tag=as4607b9c8 From: "402";tag=8d10bb3d Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-fc57b040ef2eb31f-1--d87543-;rport Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs CSeq: 1 ACK Content-Length: 0 --- (7 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 192.168.0.87:5060: INVITE sip:401@192.168.0.95 SIP/2.0 To: From: "402";tag=8d10bb3d Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-de6364197a166e66-1--d87543-;rport Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs CSeq: 2 INVITE Contact: Max-Forwards: 70 Content-Type: application/sdp Proxy-Authorization: Digest username="402",realm="asterisk",nonce="4b51d48c",uri="sip:401@192.168.0.95",response="06d693277c83b9599fba9001db9e339a",algorithm=MD5 User-Agent: Softphone2 Content-Length: 272 v=0 o=- 0 0 IN IP4 192.168.0.87 s= c=IN IP4 192.168.0.87 t=0 0 m=audio 5000 RTP/AVP 3 97 110 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 --- (12 headers 13 lines)--- Using INVITE request as basis request - 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs Sending to 192.168.0.87 : 5060 (NAT) Found user '402' Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 110 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.87:5000 Found description format GSM Found description format iLBC Found description format SPEEX Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 401 in from-internal (domain 192.168.0.95) list_route: hop: Transmitting (no NAT) to 192.168.0.87:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-de6364197a166e66-1--d87543-;rport;received=192.168.0.87 From: "402";tag=8d10bb3d To: Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- recordingcheck|20061212-112332|1165940612.50: Inbound recording not enabled dialparties.agi: Starting New Dialparties.agi dialparties.agi: Caller ID name is 'Eric Martindale (SIP)' number is '402' dialparties.agi: Methodology of ring is 'none' dialparties.agi: Extension 401 is not available to be called dialparties.agi: Extension 401 has call waiting disabled Transmitting (no NAT) to 192.168.0.87:5060: SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-de6364197a166e66-1--d87543-;rport;received=192.168.0.87 From: "402";tag=8d10bb3d To: ;tag=as119757a6 Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.0.87:5060: ACK sip:401@192.168.0.95 SIP/2.0 To: ;tag=as119757a6 From: "402";tag=8d10bb3d Via: SIP/2.0/UDP 192.168.0.87:5060;branch=z9hG4bK-d87543-de6364197a166e66-1--d87543-;rport Call-ID: 8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs CSeq: 2 ACK Content-Length: 0 --- (7 headers 0 lines)--- Destroying call '8e2404004e7c121e@Q0FMTENFTlRFUjAyLnRlY2hub2dnaW5zLmxvY2Fs'