[Mar 21 17:25:29] Asterisk SVN-branch-12-r410933 built by root @ newtonr-laptop on a x86_64 running Linux on 2014-03-19 19:33:12 UTC [Mar 21 17:25:29] VERBOSE[10291] config.c: == Parsing '/etc/asterisk/logger.conf': Found [Mar 21 17:25:29] VERBOSE[10291] logger.c: Asterisk Queue Logger restarted [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙INVITE sip:6002@10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjXRsRFc2sbwwGdA5xdpGgSWdXYdjo0fbI ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙To: ˙Contact: "RustyONE" ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 7927 INVITE ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Session-Expires: 1800 ˙Min-SE: 90 ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Type: application/sdp ˙Content-Length: 288 ˙ ˙v=0 ˙o=- 90812036 90812036 IN IP4 10.24.18.16 ˙s=digphn ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4022 RTP/AVP 0 8 9 96 ˙a=rtcp:4023 IN IP4 10.24.18.16 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:8 PCMA/8000 ˙a=rtpmap:9 G722/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: --- (15 headers 14 lines) --- [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Using INVITE request as basis request - XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found peer '6001' for '6001' from 10.24.18.16:5060 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: ˙<--- Reliably Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjXRsRFc2sbwwGdA5xdpGgSWdXYdjo0fbI;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙To: ;tag=as682e9890 ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 7927 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a784bd1" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog 'XgkXgNogOMUJe2kUonThRDjEg.90Gq9d' in 32000 ms (Method: INVITE) [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙ACK sip:6002@10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjXRsRFc2sbwwGdA5xdpGgSWdXYdjo0fbI ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙To: ;tag=as682e9890 ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 7927 ACK ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙INVITE sip:6002@10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPj-.JektU.a2wEiWHuisdY7h0.RAMzb4cM ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙To: ˙Contact: "RustyONE" ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 7928 INVITE ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Session-Expires: 1800 ˙Min-SE: 90 ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Authorization: Digest username="6001", realm="asterisk", nonce="7a784bd1", uri="sip:6002@10.24.18.124", response="99788125481debec343b73c3101c1986", algorithm=MD5 ˙Content-Type: application/sdp ˙Content-Length: 288 ˙ ˙v=0 ˙o=- 90812036 90812036 IN IP4 10.24.18.16 ˙s=digphn ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4022 RTP/AVP 0 8 9 96 ˙a=rtcp:4023 IN IP4 10.24.18.16 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:8 PCMA/8000 ˙a=rtpmap:9 G722/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: --- (16 headers 14 lines) --- [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Using INVITE request as basis request - XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found peer '6001' for '6001' from 10.24.18.16:5060 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 8 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 9 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format PCMA for ID 8 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format G722 for ID 9 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Peer audio RTP is at port 10.24.18.16:4022 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Looking for 6002 in from-internal (domain 10.24.18.124) [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: list_route: route/path hop: [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 100 Trying ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPj-.JektU.a2wEiWHuisdY7h0.RAMzb4cM;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙To: ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 7928 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:25:43] VERBOSE[10357][C-00000002] pbx.c: -- Executing [6002@from-internal:1] Dial("SIP/6001-00000004", "SIP/6002,15") in new stack [Mar 21 17:25:43] VERBOSE[10357][C-00000002] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:25:43] VERBOSE[10357][C-00000002] chan_sip.c: Audio is at 14986 [Mar 21 17:25:43] VERBOSE[10357][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:43] VERBOSE[10357][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:43] VERBOSE[10357][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK592fb047 ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ˙Contact: ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 102 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Date: Fri, 21 Mar 2014 22:25:43 GMT ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Type: application/sdp ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1905655011 1905655011 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 14986 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:25:43] VERBOSE[10357][C-00000002] app_dial.c: -- Called SIP/6002 [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 100 Trying ˙Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK592fb047 ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ˙CSeq: 102 INVITE ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: --- (7 headers 0 lines) --- [Mar 21 17:25:44] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 180 ringing ˙Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK592fb047 ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙CSeq: 102 INVITE ˙Contact: "RustyTWO" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:44] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:25:44] VERBOSE[10274][C-00000002] chan_sip.c: list_route: route/path hop: [Mar 21 17:25:44] VERBOSE[10357][C-00000002] app_dial.c: -- SIP/6002-00000005 is ringing [Mar 21 17:25:44] VERBOSE[10357][C-00000002] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 180 Ringing ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPj-.JektU.a2wEiWHuisdY7h0.RAMzb4cM;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙To: ;tag=as4a9b6f65 ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 7928 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:25:45] VERBOSE[10357][C-00000002] res_rtp_asterisk.c: > 0x7f5dc800f9d0 -- Probation passed - setting RTP source address to 10.24.18.138:4014 [Mar 21 17:25:45] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK592fb047 ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙CSeq: 102 INVITE ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Contact: "RustyTWO" ˙Supported: replaces, 100rel, timer, norefersub ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Type: application/sdp ˙Content-Length: 243 ˙ ˙v=0 ˙o=- 90812036 90812037 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4014 RTP/AVP 0 96 ˙a=rtcp:4015 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:25:45] VERBOSE[10274] chan_sip.c: --- (13 headers 12 lines) --- [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4014 [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: list_route: route/path hop: [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Transmitting (no NAT) to 10.24.18.138:5060: ˙ACK sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK67f25251 ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙Contact: ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 102 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:45] VERBOSE[10357][C-00000002] app_dial.c: -- SIP/6002-00000005 answered SIP/6001-00000004 [Mar 21 17:25:45] VERBOSE[10357][C-00000002] chan_sip.c: Audio is at 10448 [Mar 21 17:25:45] VERBOSE[10357][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:45] VERBOSE[10357][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:45] VERBOSE[10357][C-00000002] chan_sip.c: ˙<--- Reliably Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPj-.JektU.a2wEiWHuisdY7h0.RAMzb4cM;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙To: ;tag=as4a9b6f65 ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 7928 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Type: application/sdp ˙Require: timer ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1923129567 1923129567 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 10448 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙<------------> [Mar 21 17:25:45] VERBOSE[10357][C-00000002] bridge_channel.c: -- Channel SIP/6001-00000004 joined 'simple_bridge' basic-bridge [Mar 21 17:25:45] VERBOSE[10358][C-00000002] bridge_channel.c: -- Channel SIP/6002-00000005 joined 'simple_bridge' basic-bridge [Mar 21 17:25:45] VERBOSE[10358][C-00000002] bridge.c: > Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: switching from simple_bridge technology to native_rtp [Mar 21 17:25:45] VERBOSE[10358][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:45] VERBOSE[10358][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:25:45] VERBOSE[10358][C-00000002] chan_sip.c: Audio is at 14986 [Mar 21 17:25:45] VERBOSE[10358][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:45] VERBOSE[10358][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:45] VERBOSE[10358][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK15650d62 ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙Contact: ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 103 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 287 ˙ ˙v=0 ˙o=root 1905655011 1905655012 IN IP4 10.24.18.16 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙m=audio 4022 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:25:45] VERBOSE[10358][C-00000002] res_rtp_asterisk.c: > 0x7f5dc800f9d0 -- Probation passed - setting RTP source address to 10.24.18.138:4014 [Mar 21 17:25:45] VERBOSE[10357][C-00000002] res_rtp_asterisk.c: > 0x7f5d68005f40 -- Probation passed - setting RTP source address to 10.24.18.16:4022 [Mar 21 17:25:45] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙ACK sip:6002@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjeUPk8jKfWRBFLK0ilXBPsbA0iwkfwkU0 ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙To: ;tag=as4a9b6f65 ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 7928 ACK ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:45] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Audio is at 10448 [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.16:5060: ˙INVITE sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK55f0b6f7;rport ˙Max-Forwards: 70 ˙From: ;tag=as4a9b6f65 ˙To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙Contact: ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 102 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Session-Expires: 1800;refresher=uac ˙Min-SE: 90 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 289 ˙ ˙v=0 ˙o=root 1923129567 1923129568 IN IP4 10.24.18.138 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙m=audio 4014 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:25:46] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK15650d62 ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙CSeq: 103 INVITE ˙Contact: "RustyTWO" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 243 ˙ ˙v=0 ˙o=- 90812036 90812038 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4014 RTP/AVP 0 96 ˙a=rtcp:4015 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:25:46] VERBOSE[10274] chan_sip.c: --- (11 headers 12 lines) --- [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4014 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Transmitting (no NAT) to 10.24.18.138:5060: ˙ACK sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK5c2b9d45 ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙Contact: ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 103 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:46] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK55f0b6f7 ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙From: ;tag=as4a9b6f65 ˙To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙CSeq: 102 INVITE ˙Session-Expires: 1800;refresher=uac ˙Contact: "RustyONE" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 240 ˙ ˙v=0 ˙o=- 90812036 90812037 IN IP4 10.24.18.16 ˙s=digphn ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4022 RTP/AVP 0 96 ˙a=rtcp:4023 IN IP4 10.24.18.16 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:25:46] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Peer audio RTP is at port 10.24.18.16:4022 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Transmitting (no NAT) to 10.24.18.16:5060: ˙ACK sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK3039c450;rport ˙Max-Forwards: 70 ˙From: ;tag=as4a9b6f65 ˙To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙Contact: ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 102 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:46] VERBOSE[10358][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:46] VERBOSE[10358][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:25:46] VERBOSE[10358][C-00000002] chan_sip.c: Audio is at 14986 [Mar 21 17:25:46] VERBOSE[10358][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:46] VERBOSE[10358][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:46] VERBOSE[10358][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK2a4e65fa ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙Contact: ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 104 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 287 ˙ ˙v=0 ˙o=root 1905655011 1905655013 IN IP4 10.24.18.16 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙m=audio 4022 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:25:46] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK2a4e65fa ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙CSeq: 104 INVITE ˙Contact: "RustyTWO" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 243 ˙ ˙v=0 ˙o=- 90812036 90812039 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4014 RTP/AVP 0 96 ˙a=rtcp:4015 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:25:46] VERBOSE[10274] chan_sip.c: --- (11 headers 12 lines) --- [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4014 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Transmitting (no NAT) to 10.24.18.138:5060: ˙ACK sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK5b1895f9 ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙Contact: ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 104 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:51] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙INVITE sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPj7Z8t4PoWlL9WQ15erJWpeBg8f5csmsGe ˙Max-Forwards: 70 ˙From: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙To: "Alice" ;tag=as5656d5d0 ˙Contact: "RustyTWO" ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 24145 INVITE ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Session-Expires: 1800 ˙Min-SE: 90 ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Type: application/sdp ˙Content-Length: 303 ˙ ˙v=0 ˙o=- 90812036 90812040 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙a=sendonly ˙m=audio 4014 RTP/AVP 0 8 9 96 ˙a=rtcp:4015 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:8 PCMA/8000 ˙a=rtpmap:9 G722/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙a=sendonly ˙<-------------> [Mar 21 17:25:51] VERBOSE[10274] chan_sip.c: --- (15 headers 15 lines) --- [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 8 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 9 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format PCMA for ID 8 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format G722 for ID 9 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4014 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 100 Trying ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPj7Z8t4PoWlL9WQ15erJWpeBg8f5csmsGe;received=10.24.18.138;rport=5060 ˙From: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙To: "Alice" ;tag=as5656d5d0 ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 24145 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Audio is at 14986 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: ˙<--- Reliably Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPj7Z8t4PoWlL9WQ15erJWpeBg8f5csmsGe;received=10.24.18.138;rport=5060 ˙From: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙To: "Alice" ;tag=as5656d5d0 ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 24145 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Contact: ˙Content-Type: application/sdp ˙Content-Length: 287 ˙ ˙v=0 ˙o=root 1905655011 1905655014 IN IP4 10.24.18.16 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙m=audio 4022 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=recvonly ˙ ˙<------------> [Mar 21 17:25:51] VERBOSE[10357][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:51] VERBOSE[10357][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:25:51] VERBOSE[10357][C-00000002] chan_sip.c: Audio is at 10448 [Mar 21 17:25:51] VERBOSE[10357][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:51] VERBOSE[10357][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:51] VERBOSE[10357][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.16:5060: ˙INVITE sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK0a49f001;rport ˙Max-Forwards: 70 ˙From: ;tag=as4a9b6f65 ˙To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙Contact: ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 103 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Session-Expires: 1800;refresher=uac ˙Min-SE: 90 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1923129567 1923129569 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 10448 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:25:51] VERBOSE[10357][C-00000002] res_musiconhold.c: -- Started music on hold, class 'default', on channel 'SIP/6001-00000004' [Mar 21 17:25:51] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙ACK sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjjCl9j7osTxzrobciXFOKiVfVcdWdjxaE ˙Max-Forwards: 70 ˙From: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙To: "Alice" ;tag=as5656d5d0 ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 24145 ACK ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:51] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:25:52] VERBOSE[10357][C-00000002] res_rtp_asterisk.c: > 0x7f5d68005f40 -- Probation passed - setting RTP source address to 10.24.18.16:4022 [Mar 21 17:25:52] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK0a49f001 ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙From: ;tag=as4a9b6f65 ˙To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙CSeq: 103 INVITE ˙Session-Expires: 1800;refresher=uac ˙Contact: "RustyONE" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 240 ˙ ˙v=0 ˙o=- 90812036 90812038 IN IP4 10.24.18.16 ˙s=digphn ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4022 RTP/AVP 0 96 ˙a=rtcp:4023 IN IP4 10.24.18.16 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:25:52] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Peer audio RTP is at port 10.24.18.16:4022 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Transmitting (no NAT) to 10.24.18.16:5060: ˙ACK sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK1c50b65e;rport ˙Max-Forwards: 70 ˙From: ;tag=as4a9b6f65 ˙To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙Contact: ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 103 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Audio is at 10448 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.16:5060: ˙INVITE sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK029e8025;rport ˙Max-Forwards: 70 ˙From: ;tag=as4a9b6f65 ˙To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙Contact: ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 104 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Session-Expires: 1800;refresher=uac ˙Min-SE: 90 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1923129567 1923129570 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 10448 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:25:52] VERBOSE[10357][C-00000002] res_rtp_asterisk.c: > 0x7f5d68005f40 -- Probation passed - setting RTP source address to 10.24.18.16:4022 [Mar 21 17:25:52] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK029e8025 ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙From: ;tag=as4a9b6f65 ˙To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙CSeq: 104 INVITE ˙Session-Expires: 1800;refresher=uac ˙Contact: "RustyONE" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 240 ˙ ˙v=0 ˙o=- 90812036 90812039 IN IP4 10.24.18.16 ˙s=digphn ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4022 RTP/AVP 0 96 ˙a=rtcp:4023 IN IP4 10.24.18.16 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:25:52] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Peer audio RTP is at port 10.24.18.16:4022 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Transmitting (no NAT) to 10.24.18.16:5060: ˙ACK sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK342451cb;rport ˙Max-Forwards: 70 ˙From: ;tag=as4a9b6f65 ˙To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙Contact: ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 104 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:52] VERBOSE[10357][C-00000002] res_rtp_asterisk.c: > 0x7f5d68005f40 -- Probation passed - setting RTP source address to 10.24.18.16:4022 [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙INVITE sip:6003@10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjw7pegsyVY-B9abW1XjD7JCR0oIGAFMWw ˙Max-Forwards: 70 ˙From: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙To: ˙Contact: "RustyTWO" ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 28049 INVITE ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Session-Expires: 1800 ˙Min-SE: 90 ˙User-Agent: Digium D40 1_4_0_0_57389 ˙X-Digium-Call-Hint: potentialTransfer ˙Content-Type: application/sdp ˙Content-Length: 291 ˙ ˙v=0 ˙o=- 90812046 90812046 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4016 RTP/AVP 0 8 9 96 ˙a=rtcp:4017 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:8 PCMA/8000 ˙a=rtpmap:9 G722/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: --- (16 headers 14 lines) --- [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Using INVITE request as basis request - lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found peer '6002' for '6002' from 10.24.18.138:5060 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: ˙<--- Reliably Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjw7pegsyVY-B9abW1XjD7JCR0oIGAFMWw;received=10.24.18.138;rport=5060 ˙From: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙To: ;tag=as45476b2b ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 28049 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16cf0b93" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Scheduling destruction of SIP dialog 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' in 32000 ms (Method: INVITE) [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙ACK sip:6003@10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjw7pegsyVY-B9abW1XjD7JCR0oIGAFMWw ˙Max-Forwards: 70 ˙From: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙To: ;tag=as45476b2b ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 28049 ACK ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙INVITE sip:6003@10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjc8qRbpNocXKgP-.mIxJwQTqPH28TiDV1 ˙Max-Forwards: 70 ˙From: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙To: ˙Contact: "RustyTWO" ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 28050 INVITE ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Session-Expires: 1800 ˙Min-SE: 90 ˙User-Agent: Digium D40 1_4_0_0_57389 ˙X-Digium-Call-Hint: potentialTransfer ˙Authorization: Digest username="6002", realm="asterisk", nonce="16cf0b93", uri="sip:6003@10.24.18.124", response="bff28b032e6436f2ab63614ad6b00c81", algorithm=MD5 ˙Content-Type: application/sdp ˙Content-Length: 291 ˙ ˙v=0 ˙o=- 90812046 90812046 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4016 RTP/AVP 0 8 9 96 ˙a=rtcp:4017 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:8 PCMA/8000 ˙a=rtpmap:9 G722/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: --- (17 headers 14 lines) --- [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Using INVITE request as basis request - lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found peer '6002' for '6002' from 10.24.18.138:5060 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 8 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 9 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format PCMA for ID 8 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format G722 for ID 9 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4016 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Looking for 6003 in from-internal (domain 10.24.18.124) [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: list_route: route/path hop: [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 100 Trying ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjc8qRbpNocXKgP-.mIxJwQTqPH28TiDV1;received=10.24.18.138;rport=5060 ˙From: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙To: ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 28050 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:25:54] VERBOSE[10359][C-00000003] pbx.c: -- Executing [6003@from-internal:1] Dial("SIP/6002-00000006", "SIP/6003,15") in new stack [Mar 21 17:25:54] VERBOSE[10359][C-00000003] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:25:54] VERBOSE[10359][C-00000003] chan_sip.c: Audio is at 19596 [Mar 21 17:25:54] VERBOSE[10359][C-00000003] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:54] VERBOSE[10359][C-00000003] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:54] VERBOSE[10359][C-00000003] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.180:5060: ˙INVITE sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK54a6425b ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as1e36869b ˙To: ˙Contact: ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙CSeq: 102 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Date: Fri, 21 Mar 2014 22:25:54 GMT ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Type: application/sdp ˙Content-Length: 288 ˙ ˙v=0 ˙o=root 274004456 274004456 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 19596 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:25:54] VERBOSE[10359][C-00000003] app_dial.c: -- Called SIP/6003 [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 100 Trying ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK54a6425b ˙From: "Bob" ;tag=as1e36869b ˙To: "6003" ;tag=E3364A2C-61B601E7 ˙CSeq: 102 INVITE ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙Contact: ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '21608363712124-1345167932089@10.24.18.166' Method: REGISTER [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 180 Ringing ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK54a6425b ˙From: "Bob" ;tag=as1e36869b ˙To: "6003" ;tag=E3364A2C-61B601E7 ˙CSeq: 102 INVITE ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙Contact: ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Allow-Events: talk,hold,conference ˙Accept-Language: en ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: --- (11 headers 0 lines) --- [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: list_route: route/path hop: [Mar 21 17:25:54] VERBOSE[10359][C-00000003] app_dial.c: -- SIP/6003-00000007 is ringing [Mar 21 17:25:54] VERBOSE[10359][C-00000003] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 180 Ringing ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjc8qRbpNocXKgP-.mIxJwQTqPH28TiDV1;received=10.24.18.138;rport=5060 ˙From: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙To: ;tag=as5b57c5ec ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 28050 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:25:55] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK54a6425b ˙From: "Bob" ;tag=as1e36869b ˙To: "6003" ;tag=E3364A2C-61B601E7 ˙CSeq: 102 INVITE ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙Contact: ˙Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER ˙Supported: 100rel,replaces ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Content-Type: application/sdp ˙Content-Length: 197 ˙ ˙v=0 ˙o=- 1395440752 1395440752 IN IP4 10.24.18.180 ˙s=Polycom IP Phone ˙c=IN IP4 10.24.18.180 ˙t=0 0 ˙m=audio 2226 RTP/AVP 0 96 ˙a=sendrecv ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙<-------------> [Mar 21 17:25:55] VERBOSE[10274] chan_sip.c: --- (13 headers 9 lines) --- [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: Peer audio RTP is at port 10.24.18.180:2226 [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: list_route: route/path hop: [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: Transmitting (no NAT) to 10.24.18.180:5060: ˙ACK sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK4d33139a ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as1e36869b ˙To: ;tag=E3364A2C-61B601E7 ˙Contact: ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙CSeq: 102 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:55] VERBOSE[10359][C-00000003] app_dial.c: -- SIP/6003-00000007 answered SIP/6002-00000006 [Mar 21 17:25:55] VERBOSE[10359][C-00000003] chan_sip.c: Audio is at 12276 [Mar 21 17:25:55] VERBOSE[10359][C-00000003] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:55] VERBOSE[10359][C-00000003] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:55] VERBOSE[10359][C-00000003] chan_sip.c: ˙<--- Reliably Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjc8qRbpNocXKgP-.mIxJwQTqPH28TiDV1;received=10.24.18.138;rport=5060 ˙From: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙To: ;tag=as5b57c5ec ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 28050 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Type: application/sdp ˙Require: timer ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1786014922 1786014922 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 12276 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙<------------> [Mar 21 17:25:55] VERBOSE[10359][C-00000003] bridge_channel.c: -- Channel SIP/6002-00000006 joined 'simple_bridge' basic-bridge [Mar 21 17:25:55] VERBOSE[10360][C-00000003] bridge_channel.c: -- Channel SIP/6003-00000007 joined 'simple_bridge' basic-bridge [Mar 21 17:25:55] VERBOSE[10360][C-00000003] bridge.c: > Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: switching from simple_bridge technology to native_rtp [Mar 21 17:25:55] VERBOSE[10360][C-00000003] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:55] VERBOSE[10360][C-00000003] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:25:55] VERBOSE[10360][C-00000003] chan_sip.c: Audio is at 19596 [Mar 21 17:25:55] VERBOSE[10360][C-00000003] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:55] VERBOSE[10360][C-00000003] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:55] VERBOSE[10360][C-00000003] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.180:5060: ˙INVITE sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK2e718d62 ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as1e36869b ˙To: ;tag=E3364A2C-61B601E7 ˙Contact: ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙CSeq: 103 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 287 ˙ ˙v=0 ˙o=root 274004456 274004457 IN IP4 10.24.18.138 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙m=audio 4016 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:25:56] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK2e718d62 ˙From: "Bob" ;tag=as1e36869b ˙To: "6003" ;tag=E3364A2C-61B601E7 ˙CSeq: 103 INVITE ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙Contact: ˙Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER ˙Supported: 100rel,replaces ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Content-Type: application/sdp ˙Content-Length: 197 ˙ ˙v=0 ˙o=- 1395440752 1395440753 IN IP4 10.24.18.180 ˙s=Polycom IP Phone ˙c=IN IP4 10.24.18.180 ˙t=0 0 ˙m=audio 2226 RTP/AVP 0 96 ˙a=sendrecv ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙<-------------> [Mar 21 17:25:56] VERBOSE[10274] chan_sip.c: --- (13 headers 9 lines) --- [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Peer audio RTP is at port 10.24.18.180:2226 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Transmitting (no NAT) to 10.24.18.180:5060: ˙ACK sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK096cce79 ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as1e36869b ˙To: ;tag=E3364A2C-61B601E7 ˙Contact: ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙CSeq: 103 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:56] VERBOSE[10360][C-00000003] res_rtp_asterisk.c: > 0x7f5dd400f9d0 -- Probation passed - setting RTP source address to 10.24.18.180:2226 [Mar 21 17:25:56] VERBOSE[10359][C-00000003] res_rtp_asterisk.c: > 0x7f5d6803b500 -- Probation passed - setting RTP source address to 10.24.18.138:4016 [Mar 21 17:25:56] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙ACK sip:6003@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjOCWM3Y.gaYpoWCl45HulIPIpGurFHf9g ˙Max-Forwards: 70 ˙From: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙To: ;tag=as5b57c5ec ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 28050 ACK ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:56] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Audio is at 12276 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK756728b5;rport ˙Max-Forwards: 70 ˙From: ;tag=as5b57c5ec ˙To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙Contact: ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 102 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Session-Expires: 1800;refresher=uac ˙Min-SE: 90 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 289 ˙ ˙v=0 ˙o=root 1786014922 1786014923 IN IP4 10.24.18.180 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.180 ˙t=0 0 ˙m=audio 2226 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:25:56] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK756728b5 ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙From: ;tag=as5b57c5ec ˙To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙CSeq: 102 INVITE ˙Session-Expires: 1800;refresher=uac ˙Contact: "RustyTWO" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 243 ˙ ˙v=0 ˙o=- 90812046 90812047 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4016 RTP/AVP 0 96 ˙a=rtcp:4017 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:25:56] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4016 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Transmitting (no NAT) to 10.24.18.138:5060: ˙ACK sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK0b70f06e;rport ˙Max-Forwards: 70 ˙From: ;tag=as5b57c5ec ˙To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙Contact: ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 102 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:56] VERBOSE[10360][C-00000003] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:56] VERBOSE[10360][C-00000003] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:25:56] VERBOSE[10360][C-00000003] chan_sip.c: Audio is at 19596 [Mar 21 17:25:56] VERBOSE[10360][C-00000003] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:56] VERBOSE[10360][C-00000003] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:56] VERBOSE[10360][C-00000003] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.180:5060: ˙INVITE sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK28f62ab6 ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as1e36869b ˙To: ;tag=E3364A2C-61B601E7 ˙Contact: ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙CSeq: 104 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 287 ˙ ˙v=0 ˙o=root 274004456 274004458 IN IP4 10.24.18.138 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙m=audio 4016 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:25:56] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK28f62ab6 ˙From: "Bob" ;tag=as1e36869b ˙To: "6003" ;tag=E3364A2C-61B601E7 ˙CSeq: 104 INVITE ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙Contact: ˙Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER ˙Supported: 100rel,replaces ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Content-Type: application/sdp ˙Content-Length: 197 ˙ ˙v=0 ˙o=- 1395440752 1395440754 IN IP4 10.24.18.180 ˙s=Polycom IP Phone ˙c=IN IP4 10.24.18.180 ˙t=0 0 ˙m=audio 2226 RTP/AVP 0 96 ˙a=sendrecv ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙<-------------> [Mar 21 17:25:56] VERBOSE[10274] chan_sip.c: --- (13 headers 9 lines) --- [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Peer audio RTP is at port 10.24.18.180:2226 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Transmitting (no NAT) to 10.24.18.180:5060: ˙ACK sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK5a14a562 ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as1e36869b ˙To: ;tag=E3364A2C-61B601E7 ˙Contact: ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙CSeq: 104 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:58] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙REFER sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjZh9QmnwlqxhyOPbxAgRml.0BZG6R6FyQ ˙Max-Forwards: 70 ˙From: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙To: "Alice" ;tag=as5656d5d0 ˙Contact: "RustyTWO" ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 24146 REFER ˙Event: refer ˙Expires: 600 ˙Supported: replaces, 100rel, timer, norefersub ˙Accept: message/sipfrag;version=2.0 ˙Allow-Events: presence, message-summary, refer ˙Refer-To: ˙Referred-By: ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:58] VERBOSE[10274] chan_sip.c: --- (17 headers 0 lines) --- [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: Call 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 got a SIP call transfer from caller: (REFER)! [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: SIP transfer to extension 6003@from-internal by 6002@10.24.18.138 [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 202 Accepted ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjZh9QmnwlqxhyOPbxAgRml.0BZG6R6FyQ;received=10.24.18.138;rport=5060 ˙From: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙To: "Alice" ;tag=as5656d5d0 ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 24146 REFER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:25:58] VERBOSE[10357][C-00000002] res_musiconhold.c: -- Stopped music on hold on SIP/6001-00000004 [Mar 21 17:25:58] VERBOSE[10274][C-00000002] bridge_channel.c: -- Channel SIP/6001-00000004 left 'native_rtp' basic-bridge [Mar 21 17:25:58] VERBOSE[10274][C-00000002] bridge_channel.c: -- Channel SIP/6001-00000004 swapped with SIP/6002-00000006 into 'native_rtp' basic-bridge [Mar 21 17:25:58] VERBOSE[10274][C-00000002] bridge_channel.c: -- Channel SIP/6002-00000006 left 'native_rtp' basic-bridge [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: Audio is at 12276 [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK411ea1b4;rport ˙Max-Forwards: 70 ˙From: ;tag=as5b57c5ec ˙To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙Contact: ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 103 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Session-Expires: 1800;refresher=uac ˙Min-SE: 90 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1786014922 1786014924 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 12276 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:25:58] VERBOSE[10274][C-00000002] bridge.c: > Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: switching from native_rtp technology to simple_bridge [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: Audio is at 19596 [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.180:5060: ˙INVITE sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK0e356a61 ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as1e36869b ˙To: ;tag=E3364A2C-61B601E7 ˙Contact: ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙CSeq: 105 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 288 ˙ ˙v=0 ˙o=root 274004456 274004459 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 19596 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:25:58] VERBOSE[10274][C-00000002] bridge.c: > Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: switching from native_rtp technology to simple_bridge [Mar 21 17:25:58] VERBOSE[10358][C-00000002] bridge_channel.c: -- Channel SIP/6002-00000005 left 'simple_bridge' basic-bridge [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:25:58] VERBOSE[10274][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙NOTIFY sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK28040381;rport ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙Contact: ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 105 NOTIFY ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Event: refer;id=24146 ˙Subscription-state: terminated;reason=noresource ˙Content-Type: message/sipfrag;version=2.0 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Length: 16 ˙ ˙SIP/2.0 200 OK ˙ ˙--- [Mar 21 17:25:58] VERBOSE[10358][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' in 32000 ms (Method: REFER) [Mar 21 17:25:58] VERBOSE[10359][C-00000003] pbx.c: == Spawn extension (from-internal, 6003, 1) exited non-zero on 'SIP/6002-00000006' [Mar 21 17:25:58] VERBOSE[10359][C-00000003] chan_sip.c: Scheduling destruction of SIP dialog 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' in 32000 ms (Method: ACK) [Mar 21 17:25:58] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK0e356a61 ˙From: "Bob" ;tag=as1e36869b ˙To: "6003" ;tag=E3364A2C-61B601E7 ˙CSeq: 105 INVITE ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙Contact: ˙Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER ˙Supported: 100rel,replaces ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Content-Type: application/sdp ˙Content-Length: 197 ˙ ˙v=0 ˙o=- 1395440752 1395440755 IN IP4 10.24.18.180 ˙s=Polycom IP Phone ˙c=IN IP4 10.24.18.180 ˙t=0 0 ˙m=audio 2226 RTP/AVP 0 96 ˙a=sendrecv ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙<-------------> [Mar 21 17:25:58] VERBOSE[10274] chan_sip.c: --- (13 headers 9 lines) --- [Mar 21 17:25:58] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:58] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:58] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:58] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:58] VERBOSE[10274][C-00000003] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:58] VERBOSE[10274][C-00000003] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:58] VERBOSE[10274][C-00000003] chan_sip.c: Peer audio RTP is at port 10.24.18.180:2226 [Mar 21 17:25:58] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:58] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:25:58] VERBOSE[10274][C-00000003] chan_sip.c: Transmitting (no NAT) to 10.24.18.180:5060: ˙ACK sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK61ae212c ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as1e36869b ˙To: ;tag=E3364A2C-61B601E7 ˙Contact: ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙CSeq: 105 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:58] VERBOSE[10360][C-00000003] res_rtp_asterisk.c: > 0x7f5dd400f9d0 -- Probation passed - setting RTP source address to 10.24.18.180:2226 [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK411ea1b4 ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙From: ;tag=as5b57c5ec ˙To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙CSeq: 103 INVITE ˙Session-Expires: 1800;refresher=uac ˙Contact: "RustyTWO" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 243 ˙ ˙v=0 ˙o=- 90812046 90812048 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4016 RTP/AVP 0 96 ˙a=rtcp:4017 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4016 [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: Transmitting (no NAT) to 10.24.18.138:5060: ˙ACK sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK69f494d4;rport ˙Max-Forwards: 70 ˙From: ;tag=as5b57c5ec ˙To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙Contact: ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 103 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙BYE sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK439668d5;rport ˙Max-Forwards: 70 ˙From: ;tag=as5b57c5ec ˙To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙CSeq: 104 BYE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Proxy-Authorization: Digest username="6002", realm="asterisk", algorithm=MD5, uri="sip:10.24.18.124", nonce="16cf0b93", response="220008f1624e3ca0ac9c46a449e13b61" ˙X-Asterisk-HangupCause: Normal Clearing ˙X-Asterisk-HangupCauseCode: 16 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: Scheduling destruction of SIP dialog 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' in 32000 ms (Method: ACK) [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK28040381 ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙From: "Alice" ;tag=as5656d5d0 ˙To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙CSeq: 105 NOTIFY ˙Contact: "RustyTWO" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK439668d5 ˙Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 ˙From: ;tag=as5b57c5ec ˙To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 ˙CSeq: 104 BYE ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: --- (7 headers 0 lines) --- [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' Method: ACK [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙BYE sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjnnOrdpCiAGdquReHJdV0idt0IQDzx9eW ˙Max-Forwards: 70 ˙From: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙To: "Alice" ;tag=as5656d5d0 ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 24147 BYE ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: --- (9 headers 0 lines) --- [Mar 21 17:25:59] VERBOSE[10274][C-00000002] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:25:59] VERBOSE[10274][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' in 32000 ms (Method: BYE) [Mar 21 17:25:59] VERBOSE[10274][C-00000002] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjnnOrdpCiAGdquReHJdV0idt0IQDzx9eW;received=10.24.18.138;rport=5060 ˙From: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt ˙To: "Alice" ;tag=as5656d5d0 ˙Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 ˙CSeq: 24147 BYE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙REGISTER sip:10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPja.yRq.JQv446PTtoOhOggpyoEP8Lc1Uc ˙Max-Forwards: 70 ˙From: "RustyTWO" ;tag=VsHseI8XNiWkjZdDFg8CSgK4EBTFISyt ˙To: "RustyTWO" ˙Call-ID: 8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL ˙CSeq: 44933 REGISTER ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Contact: "RustyTWO" ˙Expires: 300 ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: --- (12 headers 0 lines) --- [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPja.yRq.JQv446PTtoOhOggpyoEP8Lc1Uc;received=10.24.18.138;rport=5060 ˙From: "RustyTWO" ;tag=VsHseI8XNiWkjZdDFg8CSgK4EBTFISyt ˙To: "RustyTWO" ;tag=as3eed8d32 ˙Call-ID: 8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL ˙CSeq: 44933 REGISTER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="52d40800" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog '8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL' in 32000 ms (Method: REGISTER) [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙REGISTER sip:10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjL-O9UP8sAhyzpEmVJV0Z2betbx9LskCh ˙Max-Forwards: 70 ˙From: "RustyTWO" ;tag=VsHseI8XNiWkjZdDFg8CSgK4EBTFISyt ˙To: "RustyTWO" ˙Call-ID: 8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL ˙CSeq: 44934 REGISTER ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Contact: "RustyTWO" ˙Expires: 300 ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Authorization: Digest username="6002", realm="asterisk", nonce="52d40800", uri="sip:10.24.18.124:5060", response="1e4f93a32344e5fb73676e4d87574b3d", algorithm=MD5 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: --- (13 headers 0 lines) --- [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjL-O9UP8sAhyzpEmVJV0Z2betbx9LskCh;received=10.24.18.138;rport=5060 ˙From: "RustyTWO" ;tag=VsHseI8XNiWkjZdDFg8CSgK4EBTFISyt ˙To: "RustyTWO" ;tag=as3eed8d32 ˙Call-ID: 8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL ˙CSeq: 44934 REGISTER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Expires: 300 ˙Contact: ;expires=300 ˙Date: Fri, 21 Mar 2014 22:26:06 GMT ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog '8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL' in 32000 ms (Method: REGISTER) [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SUBSCRIBE sip:6002@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjykOCVgt-CMcCD7Lw90xKJn1ku690zP1c ˙Max-Forwards: 70 ˙From: "RustyTWO" ;tag=BMMI-ojYAS5hOUEjieRW3xGdrVs6C07o ˙To: "RustyTWO" ˙Contact: "RustyTWO" ˙Call-ID: edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx ˙CSeq: 21461 SUBSCRIBE ˙Event: message-summary ˙Expires: 3600 ˙Supported: replaces, 100rel, timer, norefersub ˙Accept: application/simple-message-summary ˙Allow-Events: presence, message-summary, refer ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: --- (15 headers 0 lines) --- [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: list_route: route/path hop: [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Found peer '6002' for '6002' from 10.24.18.138:5060 [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjykOCVgt-CMcCD7Lw90xKJn1ku690zP1c;received=10.24.18.138;rport=5060 ˙From: "RustyTWO" ;tag=BMMI-ojYAS5hOUEjieRW3xGdrVs6C07o ˙To: "RustyTWO" ;tag=as27ed9052 ˙Call-ID: edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx ˙CSeq: 21461 SUBSCRIBE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="25d12960" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog 'edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx' in 32000 ms (Method: SUBSCRIBE) [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SUBSCRIBE sip:6002@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjpZYiiiQViK1Aq9cAoEVg8ZDMxRTONXDe ˙Max-Forwards: 70 ˙From: "RustyTWO" ;tag=BMMI-ojYAS5hOUEjieRW3xGdrVs6C07o ˙To: "RustyTWO" ˙Contact: "RustyTWO" ˙Call-ID: edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx ˙CSeq: 21462 SUBSCRIBE ˙Event: message-summary ˙Expires: 3600 ˙Supported: replaces, 100rel, timer, norefersub ˙Accept: application/simple-message-summary ˙Allow-Events: presence, message-summary, refer ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Authorization: Digest username="6002", realm="asterisk", nonce="25d12960", uri="sip:6002@10.24.18.124:5060", response="3b75fdab09732f3a23112e532be7bad0", algorithm=MD5 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: --- (16 headers 0 lines) --- [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Found peer '6002' for '6002' from 10.24.18.138:5060 [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 404 Not found (no mailbox) ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjpZYiiiQViK1Aq9cAoEVg8ZDMxRTONXDe;received=10.24.18.138;rport=5060 ˙From: "RustyTWO" ;tag=BMMI-ojYAS5hOUEjieRW3xGdrVs6C07o ˙To: "RustyTWO" ;tag=as27ed9052 ˙Call-ID: edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx ˙CSeq: 21462 SUBSCRIBE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:26:06] NOTICE[10274] chan_sip.c: Received SIP subscribe for peer without mailbox: 6002 [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog 'edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx' Method: SUBSCRIBE [Mar 21 17:26:07] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙BYE sip:6002@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPj21jT3x58vRsqYsakH78W0nN1byCGPtKc ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙To: ;tag=as4a9b6f65 ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 7929 BYE ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:26:07] VERBOSE[10274] chan_sip.c: --- (9 headers 0 lines) --- [Mar 21 17:26:07] VERBOSE[10274][C-00000002] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:07] VERBOSE[10274][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog 'XgkXgNogOMUJe2kUonThRDjEg.90Gq9d' in 32000 ms (Method: BYE) [Mar 21 17:26:07] VERBOSE[10274][C-00000002] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPj21jT3x58vRsqYsakH78W0nN1byCGPtKc;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF ˙To: ;tag=as4a9b6f65 ˙Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d ˙CSeq: 7929 BYE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:26:07] VERBOSE[10357][C-00000002] bridge_channel.c: -- Channel SIP/6001-00000004 left 'simple_bridge' basic-bridge [Mar 21 17:26:07] VERBOSE[10360][C-00000003] bridge_channel.c: -- Channel SIP/6003-00000007 left 'simple_bridge' basic-bridge [Mar 21 17:26:07] VERBOSE[10357][C-00000002] pbx.c: == Spawn extension (from-internal, 6002, 1) exited non-zero on 'SIP/6001-00000004' [Mar 21 17:26:07] VERBOSE[10360][C-00000003] chan_sip.c: Scheduling destruction of SIP dialog '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' in 32000 ms (Method: INVITE) [Mar 21 17:26:07] VERBOSE[10360][C-00000003] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:26:07] VERBOSE[10360][C-00000003] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:26:07] VERBOSE[10360][C-00000003] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.180:5060: ˙BYE sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK35830e09 ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as1e36869b ˙To: ;tag=E3364A2C-61B601E7 ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙CSeq: 106 BYE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙X-Asterisk-HangupCause: Normal Clearing ˙X-Asterisk-HangupCauseCode: 16 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:26:07] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK35830e09 ˙From: "Bob" ;tag=as1e36869b ˙To: "6003" ;tag=E3364A2C-61B601E7 ˙CSeq: 106 BYE ˙Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 ˙Contact: ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:26:07] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:26:07] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' Method: INVITE [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.166:5060 ---> ˙REGISTER sip:10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.166:5060;branch=z9hG4bK76673914174292040;rport ˙From: 6004 ;tag=3167523177 ˙To: 6004 ˙Call-ID: 21608363712124-1345167932089@10.24.18.166 ˙CSeq: 7 REGISTER ˙Contact: ˙Max-Forwards: 70 ˙Expires: 60 ˙Supported: path ˙User-Agent: Voip Phone 1.0 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: --- (12 headers 0 lines) --- [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.166:5060 (no NAT) [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.166:5060 (no NAT) [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.166:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.166:5060;branch=z9hG4bK76673914174292040;received=10.24.18.166;rport=5060 ˙From: 6004 ;tag=3167523177 ˙To: 6004 ;tag=as4cfa8f54 ˙Call-ID: 21608363712124-1345167932089@10.24.18.166 ˙CSeq: 7 REGISTER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="444e890e" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog '21608363712124-1345167932089@10.24.18.166' in 32000 ms (Method: REGISTER) [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.166:5060 ---> ˙REGISTER sip:10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.166:5060;branch=z9hG4bK176871677511483793;rport ˙From: 6004 ;tag=3167523177 ˙To: 6004 ˙Call-ID: 21608363712124-1345167932089@10.24.18.166 ˙CSeq: 8 REGISTER ˙Contact: ˙Authorization: Digest username="6004", realm="asterisk", nonce="444e890e", uri="sip:10.24.18.124", response="8916eb4c088b2e8df0e28c81657d440d", algorithm=MD5 ˙Max-Forwards: 70 ˙Expires: 60 ˙Supported: path ˙User-Agent: Voip Phone 1.0 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: --- (13 headers 0 lines) --- [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.166:5060 (no NAT) [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.166:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.166:5060;branch=z9hG4bK176871677511483793;received=10.24.18.166;rport=5060 ˙From: 6004 ;tag=3167523177 ˙To: 6004 ;tag=as4cfa8f54 ˙Call-ID: 21608363712124-1345167932089@10.24.18.166 ˙CSeq: 8 REGISTER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Expires: 60 ˙Contact: ;expires=60 ˙Date: Fri, 21 Mar 2014 22:26:20 GMT ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog '21608363712124-1345167932089@10.24.18.166' in 32000 ms (Method: REGISTER) [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙REGISTER sip:10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjwkdzAsZj7EnG8DXiEaJT-1TrPTXVU3Qi ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=7AQZq4QPsiEBNYkZv6mZ8eR..PSXYKmZ ˙To: "RustyONE" ˙Call-ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV ˙CSeq: 35644 REGISTER ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Contact: "RustyONE" ˙Expires: 300 ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: --- (12 headers 0 lines) --- [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjwkdzAsZj7EnG8DXiEaJT-1TrPTXVU3Qi;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=7AQZq4QPsiEBNYkZv6mZ8eR..PSXYKmZ ˙To: "RustyONE" ;tag=as2aaf4732 ˙Call-ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV ˙CSeq: 35644 REGISTER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="659a9f74" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog 'm6qybWzN.pQsFuItAEreYfCYc7g34OiV' in 32000 ms (Method: REGISTER) [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙REGISTER sip:10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjrBZR4phaqmKI1ZzATLfe6N4V3-hzviii ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=7AQZq4QPsiEBNYkZv6mZ8eR..PSXYKmZ ˙To: "RustyONE" ˙Call-ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV ˙CSeq: 35645 REGISTER ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Contact: "RustyONE" ˙Expires: 300 ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Authorization: Digest username="6001", realm="asterisk", nonce="659a9f74", uri="sip:10.24.18.124:5060", response="702a63e8cba541d825290c5fb233f6ae", algorithm=MD5 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: --- (13 headers 0 lines) --- [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjrBZR4phaqmKI1ZzATLfe6N4V3-hzviii;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=7AQZq4QPsiEBNYkZv6mZ8eR..PSXYKmZ ˙To: "RustyONE" ;tag=as2aaf4732 ˙Call-ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV ˙CSeq: 35645 REGISTER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Expires: 300 ˙Contact: ;expires=300 ˙Date: Fri, 21 Mar 2014 22:26:23 GMT ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog 'm6qybWzN.pQsFuItAEreYfCYc7g34OiV' in 32000 ms (Method: REGISTER) [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SUBSCRIBE sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjhCgIKsh0gD7VgD888leVafuC3k5Vb8Q. ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=yeuK.RmIBLmDywGlApTuWD1lqecjeUJt ˙To: "RustyONE" ˙Contact: "RustyONE" ˙Call-ID: hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 ˙CSeq: 5665 SUBSCRIBE ˙Event: message-summary ˙Expires: 3600 ˙Supported: replaces, 100rel, timer, norefersub ˙Accept: application/simple-message-summary ˙Allow-Events: presence, message-summary, refer ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: --- (15 headers 0 lines) --- [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: list_route: route/path hop: [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Found peer '6001' for '6001' from 10.24.18.16:5060 [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjhCgIKsh0gD7VgD888leVafuC3k5Vb8Q.;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=yeuK.RmIBLmDywGlApTuWD1lqecjeUJt ˙To: "RustyONE" ;tag=as0a53d884 ˙Call-ID: hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 ˙CSeq: 5665 SUBSCRIBE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="557ca6ae" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog 'hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3' in 32000 ms (Method: SUBSCRIBE) [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SUBSCRIBE sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjvWDQ4j20ZP.DyjFl8iiZu3irVS8078Zh ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=yeuK.RmIBLmDywGlApTuWD1lqecjeUJt ˙To: "RustyONE" ˙Contact: "RustyONE" ˙Call-ID: hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 ˙CSeq: 5666 SUBSCRIBE ˙Event: message-summary ˙Expires: 3600 ˙Supported: replaces, 100rel, timer, norefersub ˙Accept: application/simple-message-summary ˙Allow-Events: presence, message-summary, refer ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Authorization: Digest username="6001", realm="asterisk", nonce="557ca6ae", uri="sip:6001@10.24.18.124:5060", response="9abd4ad6331288d6d8410a3a2eea7cd9", algorithm=MD5 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: --- (16 headers 0 lines) --- [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Found peer '6001' for '6001' from 10.24.18.16:5060 [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 404 Not found (no mailbox) ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjvWDQ4j20ZP.DyjFl8iiZu3irVS8078Zh;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=yeuK.RmIBLmDywGlApTuWD1lqecjeUJt ˙To: "RustyONE" ;tag=as0a53d884 ˙Call-ID: hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 ˙CSeq: 5666 SUBSCRIBE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:26:23] NOTICE[10274] chan_sip.c: Received SIP subscribe for peer without mailbox: 6001 [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog 'hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3' Method: SUBSCRIBE [Mar 21 17:26:31] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' Method: BYE