Really destroying SIP dialog '366755007@192.168.3.28' Method: REGISTER <--- SIP read from UDP:192.168.2.68:5060 ---> INVITE sip:998022838200103@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK8036d0143fa8df11ba31001d600e4c12;rport From: ;tag=998740011 To: Call-ID: 8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68 CSeq: 24 INVITE Contact: Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite P-Preferred-Identity: Content-Length: 263 v=0 o=- 4257983104 0 IN IP4 192.168.2.68 s=SIPPER for PhonerLite c=IN IP4 192.168.2.68 t=0 0 m=audio 5062 RTP/AVP 8 0 2 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (14 headers 12 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.2.68 : 5060 (no NAT) Using INVITE request as basis request - 8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68 Found peer '868' for '868' from 192.168.2.68:5060 <--- Reliably Transmitting (no NAT) to 192.168.2.68:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK8036d0143fa8df11ba31001d600e4c12;received=192.168.2.68;rport=5060 From: ;tag=998740011 To: ;tag=as61edc761 Call-ID: 8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68 Seq: 24 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68341989" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.2.68:5060 ---> ACK sip:998022838200103@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK8036d0143fa8df11ba31001d600e4c12;rport From: ;tag=998740011 To: ;tag=as61edc761 Call-ID: 8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68 CSeq: 24 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:192.168.2.68:5060 ---> INVITE sip:998022838200103@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK8036d0143fa8df11ba32001d600e4c12;rport From: ;tag=998740011 To: Call-ID: 8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68 CSeq: 25 INVITE Contact: Authorization: Digest username="868", realm="asterisk", nonce="68341989", uri="sip:998022838200103@192.168.3.30", response="576f3b29855ae84eaa93154dd758743c", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite P-Preferred-Identity: Content-Length: 263 v=0 o=- 4257983104 0 IN IP4 192.168.2.68 s=SIPPER for PhonerLite c=IN IP4 192.168.2.68 t=0 0 m=audio 5062 RTP/AVP 8 0 2 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.2.68 : 5060 (no NAT) Using INVITE request as basis request - 8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68 Found peer '868' for '868' from 192.168.2.68:5060 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x80c (ulaw|alaw|g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.2.68:5062 Looking for 998022838200103 in interview (domain 192.168.3.30) list_route: hop: <--- Transmitting (no NAT) to 192.168.2.68:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK8036d0143fa8df11ba32001d600e4c12;received=192.168.2.68;rport=5060 From: ;tag=998740011 To: Call-ID: 8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68 CSeq: 25 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [998022838200103@interview:1] Answer("SIP/868-0000021e", "") in new stack Audio is at 192.168.3.30 port 12742 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 192.168.2.68:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK8036d0143fa8df11ba32001d600e4c12;received=192.168.2.68;rport=5060 From: ;tag=998740011 To: ;tag=as0504a185 Call-ID: 8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68 CSeq: 25 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 201 v=0 o=root 30990369 30990369 IN IP4 192.168.3.30 s=Asterisk PBX 1.6.2.11 c=IN IP4 192.168.3.30 t=0 0 m=audio 12742 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:192.168.2.68:5060 ---> ACK sip:998022838200103@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK00cd68153fa8df11ba32001d600e4c12;rport From: ;tag=998740011 To: ;tag=as0504a185 Call-ID: 8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68 CSeq: 25 ACK Contact: Authorization: Digest username="868", realm="asterisk", nonce="68341989", uri="sip:998022838200103@192.168.3.30", response="576f3b29855ae84eaa93154dd758743c", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Executing [998022838200103@interview:2] PlayTones("SIP/868-0000021e", "waitforring") in new stack -- Executing [998022838200103@interview:3] Dial("SIP/868-0000021e", "SIP/022838200103@gsmgate01:5060&Local/s@no-op,60") in new stack == Using SIP RTP CoS mark 5 Audio is at 192.168.3.30 port 13664 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 192.168.3.28:5060: INVITE sip:022838200103@192.168.3.28:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK3a445604;rport Max-Forwards: 70 From: "868" ;tag=as0751ff69 To: Contact: Call-ID: 04c6bad13eb96fb01d5dee773bdb7b81@192.168.3.30 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.11 Date: Tue, 17 Aug 2010 07:31:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 203 v=0 o=root 701501568 701501568 IN IP4 192.168.3.30 s=Asterisk PBX 1.6.2.11 c=IN IP4 192.168.3.30 t=0 0 m=audio 13664 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- -- Called 022838200103@gsmgate01:5060 -- Executing [s@no-op:1] Hangup("Local/s@no-op-150d;2", "") in new stack == Spawn extension (no-op, s, 1) exited non-zero on 'Local/s@no-op-150d;2' -- Called s@no-op <--- SIP read from UDP:192.168.3.28:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK3a445604;rport=5060 From: "868" ;tag=as0751ff69 To: Call-ID: 04c6bad13eb96fb01d5dee773bdb7b81@192.168.3.30 CSeq: 102 INVITE Server: SER at EcotelVoIP Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.3.28:5060 ---> SIP/2.0 100 Trying gateway Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK3a445604;rport=5060 From: "868" ;tag=as0751ff69 To: Call-ID: 04c6bad13eb96fb01d5dee773bdb7b81@192.168.3.30 CSeq: 102 INVITE Server: SER at EcotelVoIP Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.3.28:5060 ---> REGISTER sip:192.168.3.30:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bK8c7d.04acd966.0 Via: SIP/2.0/UDP 192.168.3.28:5061;branch=z9hG4bK.4f814dc5;rport=5061 From: sip:gsmgate01@192.168.3.30;tag=7c5743d6 To: sip:gsmgate01@192.168.3.30 Call-ID: 2086093782@192.168.3.28 CSeq: 1 REGISTER Content-Length: 0 Max-Forwards: 69 User-Agent: sipsak 0.9.1 Expires: 72 Contact: sip:gsmgate01@192.168.3.28 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.3.28 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.28:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bK8c7d.04acd966.0;received=192.168.3.28 Via: SIP/2.0/UDP 192.168.3.28:5061;branch=z9hG4bK.4f814dc5;rport=5061 From: sip:gsmgate01@192.168.3.30;tag=7c5743d6 To: sip:gsmgate01@192.168.3.30;tag=as68bf2b4e Call-ID: 2086093782@192.168.3.28 CSeq: 1 REGISTER Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="286df2bf" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2086093782@192.168.3.28' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.3.28:5060 ---> REGISTER sip:192.168.3.30:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bK8c7d.04acd966.1 Via: SIP/2.0/UDP 192.168.3.28:5061;branch=z9hG4bK.4f814dc5;rport=5061 From: sip:gsmgate01@192.168.3.30;tag=7c5743d6 To: sip:gsmgate01@192.168.3.30 Call-ID: 2086093782@192.168.3.28 CSeq: 2 REGISTER Content-Length: 0 Max-Forwards: 69 User-Agent: sipsak 0.9.1 Expires: 72 Contact: sip:gsmgate01@192.168.3.28 Authorization: Digest username="gsmgate01", realm="asterisk", nonce="286df2bf", uri="sip:192.168.3.30:5060", response="56edbf33dc45faba339418b482e3d373", algorithm="MD5" <-------------> --- (14 headers 0 lines) --- Sending to 192.168.3.28 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.28:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bK8c7d.04acd966.1;received=192.168.3.28 Via: SIP/2.0/UDP 192.168.3.28:5061;branch=z9hG4bK.4f814dc5;rport=5061 Record-Route: From: sip:gsmgate01@192.168.3.30;tag=7c5743d6 To: sip:gsmgate01@192.168.3.30;tag=as68bf2b4e Call-ID: 2086093782@192.168.3.28 CSeq: 2 REGISTER Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 72 Contact: ;expires=72 Date: Tue, 17 Aug 2010 07:31:18 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2086093782@192.168.3.28' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.3.28:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK3a445604;rport=5060 From: "868" ;tag=as0751ff69 To: ;tag=8e01d3189c3fea1a4c23e24fccc5da37-0f84 Call-ID: 04c6bad13eb96fb01d5dee773bdb7b81@192.168.3.30 CSeq: 102 INVITE Contact: Allow: INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Server: SER at EcotelVoIP Content-Length: 143 v=0 o=022838200103 168225 168225 IN IP4 192.168.3.28 s=SIP Call c=IN IP4 192.168.3.28 t=0 0 m=audio 5004 RTP/AVP 8 a=rtpmap:8 PCMA/8000 <-------------> --- (11 headers 7 lines) --- Found RTP audio format 8 Found audio description format PCMA for ID 8 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.3.28:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.28, port 5060 Transmitting (no NAT) to 192.168.3.28:5060: ACK sip:022838200103@192.168.3.28 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK160194a0;rport Max-Forwards: 70 From: "868" ;tag=as0751ff69 To: ;tag=8e01d3189c3fea1a4c23e24fccc5da37-0f84 Contact: Call-ID: 04c6bad13eb96fb01d5dee773bdb7b81@192.168.3.30 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.11 Content-Length: 0 --- -- SIP/gsmgate01:5060-0000021f answered SIP/868-0000021e -- Packet2Packet bridging SIP/868-0000021e and SIP/gsmgate01:5060-0000021f <--- SIP read from UDP:192.168.3.28:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK3a445604;rport=5060 From: "868" ;tag=as0751ff69 To: ;tag=8e01d3189c3fea1a4c23e24fccc5da37-0f84 Call-ID: 04c6bad13eb96fb01d5dee773bdb7b81@192.168.3.30 CSeq: 102 INVITE Contact: Allow: INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Server: SER at EcotelVoIP Content-Length: 143 v=0 o=022838200103 168225 168225 IN IP4 192.168.3.28 s=SIP Call c=IN IP4 192.168.3.28 t=0 0 m=audio 5004 RTP/AVP 8 a=rtpmap:8 PCMA/8000 <-------------> --- (11 headers 7 lines) --- <--- SIP read from UDP:192.168.2.38:5060 ---> INVITE sip:7868@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.38:5060;branch=z9hG4bK809e591e3fa8df11a7cd0019d1eadc3c;rport From: "441" ;tag=28276992 To: Call-ID: 809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38 CSeq: 25 INVITE Contact: Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite Content-Length: 263 v=0 o=- 3310221319 0 IN IP4 192.168.2.38 s=SIPPER for PhonerLite c=IN IP4 192.168.2.38 t=0 0 m=audio 5062 RTP/AVP 8 0 2 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (13 headers 12 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.2.38 : 5060 (no NAT) Using INVITE request as basis request - 809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38 Found peer '441' for '441' from 192.168.2.38:5060 <--- Reliably Transmitting (no NAT) to 192.168.2.38:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.38:5060;branch=z9hG4bK809e591e3fa8df11a7cd0019d1eadc3c;received=192.168.2.38;rport=5060 From: "441" ;tag=28276992 To: ;tag=as4eb32c37 Call-ID: 809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38 CSeq: 25 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a682338" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.2.38:5060 ---> ACK sip:7868@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.38:5060;branch=z9hG4bK809e591e3fa8df11a7cd0019d1eadc3c;rport From: "441" ;tag=28276992 To: ;tag=as4eb32c37 Call-ID: 809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38 CSeq: 25 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:192.168.2.38:5060 ---> INVITE sip:7868@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.38:5060;branch=z9hG4bK809e591e3fa8df11a7ce0019d1eadc3c;rport From: "441" ;tag=28276992 To: Call-ID: 809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38 CSeq: 26 INVITE Contact: Authorization: Digest username="441", realm="asterisk", nonce="1a682338", uri="sip:7868@192.168.3.30", response="c34c81a1f1a79b7608088b2053c9b07d", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite Content-Length: 263 v=0 o=- 3310221319 0 IN IP4 192.168.2.38 s=SIPPER for PhonerLite c=IN IP4 192.168.2.38 t=0 0 m=audio 5062 RTP/AVP 8 0 2 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (14 headers 12 lines) --- Sending to 192.168.2.38 : 5060 (no NAT) Using INVITE request as basis request - 809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38 Found peer '441' for '441' from 192.168.2.38:5060 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x80c (ulaw|alaw|g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.2.38:5062 Looking for 7868 in office (domain 192.168.3.30) list_route: hop: <--- Transmitting (no NAT) to 192.168.2.38:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.38:5060;branch=z9hG4bK809e591e3fa8df11a7ce0019d1eadc3c;received=192.168.2.38;rport=5060 From: "441" ;tag=28276992 To: Call-ID: 809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38 CSeq: 26 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [7868@office:1] ChanSpy("SIP/441-00000220", "SIP/868,qv(1)") in new stack Audio is at 192.168.3.30 port 11546 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 192.168.2.38:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.38:5060;branch=z9hG4bK809e591e3fa8df11a7ce0019d1eadc3c;received=192.168.2.38;rport=5060 From: "441" ;tag=28276992 To: ;tag=as047e7334 Call-ID: 809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38 CSeq: 26 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 201 v=0 o=root 29044870 29044870 IN IP4 192.168.3.30 s=Asterisk PBX 1.6.2.11 c=IN IP4 192.168.3.30 t=0 0 m=audio 11546 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:192.168.2.38:5060 ---> ACK sip:7868@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.38:5060;branch=z9hG4bK809e591e3fa8df11a7cf0019d1eadc3c;rport From: "441" ;tag=28276992 To: ;tag=as047e7334 Call-ID: 809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38 CSeq: 26 ACK Contact: Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- == Spying on channel SIP/868-0000021e [Aug 17 09:31:30] NOTICE[5794]: app_chanspy.c:414 start_spying: Attaching SIP/441-00000220 to SIP/868-0000021e [Aug 17 09:31:30] NOTICE[5794]: app_chanspy.c:414 start_spying: Attaching SIP/441-00000220 to SIP/868-0000021e [Aug 17 09:31:30] NOTICE[5794]: app_chanspy.c:414 start_spying: Attaching SIP/441-00000220 to SIP/gsmgate01:5060-0000021f <--- SIP read from UDP:192.168.3.28:5060 ---> BYE sip:868@192.168.3.30:5060 SIP/2.0 Max-Forwards: 70 Record-Route: Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bKcabe.688a8325.0 Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bKcabe.588a8325.0 To: "868" ;tag=as0751ff69 From: ;tag=8e01d3189c3fea1a4c23e24fccc5da37-0f84 CSeq: 1 BYE Call-ID: 04c6bad13eb96fb01d5dee773bdb7b81@192.168.3.30 Content-Length: 0 User-Agent: SER at EcotelVoIP <-------------> --- (11 headers 0 lines) --- Sending to 192.168.3.28 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.28:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bKcabe.688a8325.0;received=192.168.3.28 Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bKcabe.588a8325.0 Record-Route: From: ;tag=8e01d3189c3fea1a4c23e24fccc5da37-0f84 To: "868" ;tag=as0751ff69 Call-ID: 04c6bad13eb96fb01d5dee773bdb7b81@192.168.3.30 CSeq: 1 BYE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (interview, 998022838200103, 3) exited non-zero on 'SIP/868-0000021e' Scheduling destruction of SIP dialog '8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.68, port 5060 Reliably Transmitting (no NAT) to 192.168.2.68:5060: BYE sip:868@192.168.2.68:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK6d9019c6;rport Max-Forwards: 70 From: ;tag=as0504a185 To: ;tag=998740011 Call-ID: 8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.2.11 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Done Spying on channel SIP/868-0000021e <--- SIP read from UDP:192.168.2.68:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK6d9019c6;rport=5060 From: ;tag=as0504a185 To: ;tag=998740011 Call-ID: 8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68 CSeq: 102 BYE Contact: User-Agent: SIPPER for PhonerLite Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '8036D014-3FA8-DF11-BA30-001D600E4C12@192.168.2.68' Method: ACK Really destroying SIP dialog '04c6bad13eb96fb01d5dee773bdb7b81@192.168.3.30' Method: BYE <--- SIP read from UDP:192.168.2.68:5060 ---> INVITE sip:998022838200103@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK00704a273fa8df11ba33001d600e4c12;rport From: ;tag=3683519063 To: Call-ID: 00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68 CSeq: 26 INVITE Contact: Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite P-Preferred-Identity: Content-Length: 263 v=0 o=- 2548643813 0 IN IP4 192.168.2.68 s=SIPPER for PhonerLite c=IN IP4 192.168.2.68 t=0 0 m=audio 5062 RTP/AVP 8 0 2 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (14 headers 12 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.2.68 : 5060 (no NAT) Using INVITE request as basis request - 00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68 Found peer '868' for '868' from 192.168.2.68:5060 <--- Reliably Transmitting (no NAT) to 192.168.2.68:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK00704a273fa8df11ba33001d600e4c12;received=192.168.2.68;rport=5060 From: ;tag=3683519063 To: ;tag=as4ca005d4 Call-ID: 00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68 CSeq: 26 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="425364ef" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.2.68:5060 ---> ACK sip:998022838200103@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK00704a273fa8df11ba33001d600e4c12;rport From: ;tag=3683519063 To: ;tag=as4ca005d4 Call-ID: 00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68 CSeq: 26 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:192.168.2.68:5060 ---> INVITE sip:998022838200103@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK00704a273fa8df11ba34001d600e4c12;rport From: ;tag=3683519063 To: Call-ID: 00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68 CSeq: 27 INVITE Contact: Authorization: Digest username="868", realm="asterisk", nonce="425364ef", uri="sip:998022838200103@192.168.3.30", response="adc8c5410aa2d3d821d0c1648036e4e0", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite P-Preferred-Identity: Content-Length: 263 v=0 o=- 2548643813 0 IN IP4 192.168.2.68 s=SIPPER for PhonerLite c=IN IP4 192.168.2.68 t=0 0 m=audio 5062 RTP/AVP 8 0 2 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.2.68 : 5060 (no NAT) Using INVITE request as basis request - 00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68 Found peer '868' for '868' from 192.168.2.68:5060 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x80c (ulaw|alaw|g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.2.68:5062 Looking for 998022838200103 in interview (domain 192.168.3.30) list_route: hop: <--- Transmitting (no NAT) to 192.168.2.68:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK00704a273fa8df11ba34001d600e4c12;received=192.168.2.68;rport=5060 From: ;tag=3683519063 To: Call-ID: 00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68 CSeq: 27 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [998022838200103@interview:1] Answer("SIP/868-00000221", "") in new stack Audio is at 192.168.3.30 port 15258 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 192.168.2.68:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK00704a273fa8df11ba34001d600e4c12;received=192.168.2.68;rport=5060 From: ;tag=3683519063 To: ;tag=as74e51473 Call-ID: 00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68 CSeq: 27 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 205 v=0 o=root 1619452869 1619452869 IN IP4 192.168.3.30 s=Asterisk PBX 1.6.2.11 c=IN IP4 192.168.3.30 t=0 0 m=audio 15258 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv <------------> == Spying on channel SIP/868-00000221 [Aug 17 09:31:45] NOTICE[5794]: app_chanspy.c:414 start_spying: Attaching SIP/441-00000220 to SIP/868-00000221 [Aug 17 09:31:45] NOTICE[5794]: app_chanspy.c:414 start_spying: Attaching SIP/441-00000220 to SIP/868-00000221 <--- SIP read from UDP:192.168.2.68:5060 ---> ACK sip:998022838200103@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.68:5060;branch=z9hG4bK00704a273fa8df11ba35001d600e4c12;rport From: ;tag=3683519063 To: ;tag=as74e51473 Call-ID: 00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68 CSeq: 27 ACK Contact: Authorization: Digest username="868", realm="asterisk", nonce="425364ef", uri="sip:998022838200103@192.168.3.30", response="adc8c5410aa2d3d821d0c1648036e4e0", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Executing [998022838200103@interview:2] PlayTones("SIP/868-00000221", "waitforring") in new stack -- Executing [998022838200103@interview:3] Dial("SIP/868-00000221", "SIP/022838200103@gsmgate01:5060&Local/s@no-op,60") in new stack == Using SIP RTP CoS mark 5 Audio is at 192.168.3.30 port 12288 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 192.168.3.28:5060: INVITE sip:022838200103@192.168.3.28:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK2a429807;rport Max-Forwards: 70 From: "868" ;tag=as01967075 To: Contact: Call-ID: 50dfe95c7d11fbc10baf6598030d49a1@192.168.3.30 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.11 Date: Tue, 17 Aug 2010 07:31:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 205 v=0 o=root 1120136021 1120136021 IN IP4 192.168.3.30 s=Asterisk PBX 1.6.2.11 c=IN IP4 192.168.3.30 t=0 0 m=audio 12288 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- -- Called 022838200103@gsmgate01:5060 -- Executing [s@no-op:1] Hangup("Local/s@no-op-0bd3;2", "") in new stack == Spawn extension (no-op, s, 1) exited non-zero on 'Local/s@no-op-0bd3;2' -- Called s@no-op <--- SIP read from UDP:192.168.3.28:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK2a429807;rport=5060 From: "868" ;tag=as01967075 To: Call-ID: 50dfe95c7d11fbc10baf6598030d49a1@192.168.3.30 CSeq: 102 INVITE Server: SER at EcotelVoIP Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.3.28:5060 ---> SIP/2.0 100 Trying gateway Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK2a429807;rport=5060 From: "868" ;tag=as01967075 To: Call-ID: 50dfe95c7d11fbc10baf6598030d49a1@192.168.3.30 CSeq: 102 INVITE Server: SER at EcotelVoIP Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '2086093782@192.168.3.28' Method: REGISTER <--- SIP read from UDP:192.168.3.28:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK2a429807;rport=5060 From: "868" ;tag=as01967075 To: ;tag=8e01d3189c3fea1a4c23e24fccc5da37-3674 Call-ID: 50dfe95c7d11fbc10baf6598030d49a1@192.168.3.30 CSeq: 102 INVITE Contact: Allow: INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Server: SER at EcotelVoIP Content-Length: 143 v=0 o=022838200103 168256 168256 IN IP4 192.168.3.28 s=SIP Call c=IN IP4 192.168.3.28 t=0 0 m=audio 5008 RTP/AVP 8 a=rtpmap:8 PCMA/8000 <-------------> --- (11 headers 7 lines) --- Found RTP audio format 8 Found audio description format PCMA for ID 8 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.3.28:5008 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.3.28, port 5060 Transmitting (no NAT) to 192.168.3.28:5060: ACK sip:022838200103@192.168.3.28 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK42d80387;rport Max-Forwards: 70 From: "868" ;tag=as01967075 To: ;tag=8e01d3189c3fea1a4c23e24fccc5da37-3674 Contact: Call-ID: 50dfe95c7d11fbc10baf6598030d49a1@192.168.3.30 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.11 Content-Length: 0 --- -- SIP/gsmgate01:5060-00000222 answered SIP/868-00000221 <--- SIP read from UDP:192.168.3.28:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK2a429807;rport=5060 From: "868" ;tag=as01967075 To: ;tag=8e01d3189c3fea1a4c23e24fccc5da37-3674 Call-ID: 50dfe95c7d11fbc10baf6598030d49a1@192.168.3.30 CSeq: 102 INVITE Contact: Allow: INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Server: SER at EcotelVoIP Content-Length: 143 v=0 o=022838200103 168256 168256 IN IP4 192.168.3.28 s=SIP Call c=IN IP4 192.168.3.28 t=0 0 m=audio 5008 RTP/AVP 8 a=rtpmap:8 PCMA/8000 <-------------> --- (11 headers 7 lines) --- <--- SIP read from UDP:192.168.3.28:5060 ---> BYE sip:868@192.168.3.30:5060 SIP/2.0 Max-Forwards: 70 Record-Route: Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bK28ad.b29df9e4.0 Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bK28ad.a29df9e4.0 To: "868" ;tag=as01967075 From: ;tag=8e01d3189c3fea1a4c23e24fccc5da37-3674 CSeq: 1 BYE Call-ID: 50dfe95c7d11fbc10baf6598030d49a1@192.168.3.30 Content-Length: 0 User-Agent: SER at EcotelVoIP <-------------> --- (11 headers 0 lines) --- Sending to 192.168.3.28 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.28:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bK28ad.b29df9e4.0;received=192.168.3.28 Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bK28ad.a29df9e4.0 Record-Route: From: ;tag=8e01d3189c3fea1a4c23e24fccc5da37-3674 To: "868" ;tag=as01967075 Call-ID: 50dfe95c7d11fbc10baf6598030d49a1@192.168.3.30 CSeq: 1 BYE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (interview, 998022838200103, 3) exited non-zero on 'SIP/868-00000221' Scheduling destruction of SIP dialog '00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.68, port 5060 Reliably Transmitting (no NAT) to 192.168.2.68:5060: BYE sip:868@192.168.2.68:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK2ce30601;rport Max-Forwards: 70 From: ;tag=as74e51473 To: ;tag=3683519063 Call-ID: 00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.2.11 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Done Spying on channel SIP/868-00000221 <--- SIP read from UDP:192.168.2.68:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.30:5060;branch=z9hG4bK2ce30601;rport=5060 From: ;tag=as74e51473 To: ;tag=3683519063 Call-ID: 00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68 CSeq: 102 BYE Contact: User-Agent: SIPPER for PhonerLite Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '00704A27-3FA8-DF11-BA32-001D600E4C12@192.168.2.68' Method: ACK Really destroying SIP dialog '50dfe95c7d11fbc10baf6598030d49a1@192.168.3.30' Method: BYE <--- SIP read from UDP:192.168.2.38:5060 ---> BYE sip:7868@192.168.3.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.38:5060;branch=z9hG4bK00132c393fa8df11a7cf0019d1eadc3c;rport From: "441" ;tag=28276992 To: ;tag=as047e7334 Call-ID: 809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38 CSeq: 27 BYE Contact: Max-Forwards: 70 User-Agent: SIPPER for PhonerLite Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 192.168.2.38 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.2.38:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.38:5060;branch=z9hG4bK00132c393fa8df11a7cf0019d1eadc3c;received=192.168.2.38;rport=5060 From: "441" ;tag=28276992 To: ;tag=as047e7334 Call-ID: 809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38 CSeq: 27 BYE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (office, 7868, 1) exited non-zero on 'SIP/441-00000220' Really destroying SIP dialog '809E591E-3FA8-DF11-A7CC-0019D1EADC3C@192.168.2.38' Method: BYE <--- SIP read from UDP:192.168.3.28:5060 ---> REGISTER sip:192.168.3.30:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bK190a.dd239473.0 Via: SIP/2.0/UDP 192.168.3.28:5061;branch=z9hG4bK.2e1c5020;rport=5061 From: sip:gsmgate01@192.168.3.30;tag=3f4ce3f To: sip:gsmgate01@192.168.3.30 Call-ID: 66375231@192.168.3.28 CSeq: 1 REGISTER Content-Length: 0 Max-Forwards: 69 User-Agent: sipsak 0.9.1 Expires: 72 Contact: sip:gsmgate01@192.168.3.28 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.3.28 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.28:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bK190a.dd239473.0;received=192.168.3.28 Via: SIP/2.0/UDP 192.168.3.28:5061;branch=z9hG4bK.2e1c5020;rport=5061 From: sip:gsmgate01@192.168.3.30;tag=3f4ce3f To: sip:gsmgate01@192.168.3.30;tag=as5665a461 Call-ID: 66375231@192.168.3.28 CSeq: 1 REGISTER Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56fee8e7" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '66375231@192.168.3.28' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.3.28:5060 ---> REGISTER sip:192.168.3.30:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bK190a.dd239473.1 Via: SIP/2.0/UDP 192.168.3.28:5061;branch=z9hG4bK.2e1c5020;rport=5061 From: sip:gsmgate01@192.168.3.30;tag=3f4ce3f To: sip:gsmgate01@192.168.3.30 Call-ID: 66375231@192.168.3.28 CSeq: 2 REGISTER Content-Length: 0 Max-Forwards: 69 User-Agent: sipsak 0.9.1 Expires: 72 Contact: sip:gsmgate01@192.168.3.28 Authorization: Digest username="gsmgate01", realm="asterisk", nonce="56fee8e7", uri="sip:192.168.3.30:5060", response="6fb94575d93caed4f51f83fff31ef132", algorithm="MD5" <-------------> --- (14 headers 0 lines) --- Sending to 192.168.3.28 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.28:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.28;branch=z9hG4bK190a.dd239473.1;received=192.168.3.28 Via: SIP/2.0/UDP 192.168.3.28:5061;branch=z9hG4bK.2e1c5020;rport=5061 Record-Route: From: sip:gsmgate01@192.168.3.30;tag=3f4ce3f To: sip:gsmgate01@192.168.3.30;tag=as5665a461 Call-ID: 66375231@192.168.3.28 CSeq: 2 REGISTER Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 72 Contact: ;expires=72 Date: Tue, 17 Aug 2010 07:32:18 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '66375231@192.168.3.28' in 32000 ms (Method: REGISTER)