[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] DEBUG[10291] config.c: Parsing /etc/asterisk/logger.conf [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] DEBUG[10274] chan_sip.c: Header 0 [ 36]: INVITE sip:6002@10.24.18.124 SIP/2.0 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjXRsRFc2sbwwGdA5xdpGgSWdXYdjo0fbI [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 4 [ 27]: To: [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 5 [ 50]: Contact: "RustyONE" [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 7 [ 17]: CSeq: 7927 INVITE [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 12 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 14 [ 19]: Content-Length: 288 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 15 [ 0]: [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 1 [ 40]: o=- 90812036 90812036 IN IP4 10.24.18.16 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 6 [ 29]: m=audio 4022 RTP/AVP 0 8 9 96 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 7 [ 30]: a=rtcp:4023 IN IP4 10.24.18.16 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 10 [ 20]: a=rtpmap:9 G722/8000 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 11 [ 10]: a=sendrecv [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 12 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 13 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: --- (15 headers 14 lines) --- [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: = Looking for Call ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d (Checking From) --From tag Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF --To-tag [Mar 21 17:25:43] DEBUG[10274] acl.c: For destination '10.24.18.16', our source address is '10.24.18.124'. [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:25:43] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:25:43] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for XgkXgNogOMUJe2kUonThRDjEg.90Gq9d - INVITE (No RTP) [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 21 17:25:43] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, timer, norefersub" [Mar 21 17:25:43] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Found SIP option: -replaces- [Mar 21 17:25:43] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Matched SIP option: replaces [Mar 21 17:25:43] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Found SIP option: -100rel- [Mar 21 17:25:43] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Matched SIP option: 100rel [Mar 21 17:25:43] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Found SIP option: -timer- [Mar 21 17:25:43] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Matched SIP option: timer [Mar 21 17:25:43] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Found SIP option: -norefersub- [Mar 21 17:25:43] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Matched SIP option: norefersub [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: Initializing initreq for method INVITE - callid XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [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] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #77 [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.16:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 33]: ACK sip:6002@10.24.18.124 SIP/2.0 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjXRsRFc2sbwwGdA5xdpGgSWdXYdjo0fbI [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 4 [ 42]: To: ;tag=as682e9890 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 6 [ 14]: CSeq: 7927 ACK [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: = Looking for Call ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d (Checking From) --From tag Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF --To-tag as682e9890 [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #77 [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Stopping retransmission on 'XgkXgNogOMUJe2kUonThRDjEg.90Gq9d' of Response 7927: Match Found [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] DEBUG[10274] chan_sip.c: Header 0 [ 36]: INVITE sip:6002@10.24.18.124 SIP/2.0 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPj-.JektU.a2wEiWHuisdY7h0.RAMzb4cM [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 4 [ 27]: To: [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 5 [ 50]: Contact: "RustyONE" [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 7 [ 17]: CSeq: 7928 INVITE [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 12 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 13 [162]: Authorization: Digest username="6001", realm="asterisk", nonce="7a784bd1", uri="sip:6002@10.24.18.124", response="99788125481debec343b73c3101c1986", algorithm=MD5 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 15 [ 19]: Content-Length: 288 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 16 [ 0]: [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 1 [ 40]: o=- 90812036 90812036 IN IP4 10.24.18.16 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 6 [ 29]: m=audio 4022 RTP/AVP 0 8 9 96 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 7 [ 30]: a=rtcp:4023 IN IP4 10.24.18.16 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 10 [ 20]: a=rtpmap:9 G722/8000 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 11 [ 10]: a=sendrecv [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 12 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Body 13 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: --- (16 headers 14 lines) --- [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: = Looking for Call ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d (Checking From) --From tag Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF --To-tag [Mar 21 17:25:43] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:43] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:25:43] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:43] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: Initializing initreq for method INVITE - callid XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [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] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274][C-00000002] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f5d6800bc28' [Mar 21 17:25:43] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Allocated port 10448 for RTP instance '0x7f5d6800bc28' [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:25:43] DEBUG[10274][C-00000002] rtp_engine.c: RTP instance '0x7f5d6800bc28' is setup and ready to go [Mar 21 17:25:43] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6800bc28' [Mar 21 17:25:43] VERBOSE[10274][C-00000002] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Setting NAT on RTP to Off [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP o=- 90812036 90812036 IN IP4 10.24.18.16... OK. [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16' into... [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port ''. [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.16... OK. [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:43] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f8590 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 8 [Mar 21 17:25:43] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 8 based on m type on 0x7f5d818f8590 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 9 [Mar 21 17:25:43] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 9 based on m type on 0x7f5d818f8590 [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:43] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f8590 [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4023 IN IP4 10.24.18.16... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5d6800bc28' [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] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 0 from 0x7f5d818f8590 to 0x7f5d6800bd68 [Mar 21 17:25:43] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 8 from 0x7f5d818f8590 to 0x7f5d6800bd68 [Mar 21 17:25:43] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 9 from 0x7f5d818f8590 to 0x7f5d6800bd68 [Mar 21 17:25:43] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 96 from 0x7f5d818f8590 to 0x7f5d6800bd68 [Mar 21 17:25:43] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5d6800bc28' [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Checking SIP call limits for device 6001 [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Updating call counter for incoming call [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:43] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274][C-00000002] chan_sip.c: Incoming INVITE with 'timer' option supported [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: INVITE also has "Session-Expires" header. [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Session-Expires: 1800 [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: INVITE also has "Min-SE" header. [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Received Min-SE: 90 [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: *** Our native formats are (ulaw) [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: *** Joint capabilities are (ulaw) [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: *** Our capabilities are (ulaw) [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: This channel will not be able to handle video. [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: build_route: Contact hop: "RustyONE" [Mar 21 17:25:43] VERBOSE[10274][C-00000002] chan_sip.c: list_route: route/path hop: [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: Session timer started: 79 - XgkXgNogOMUJe2kUonThRDjEg.90Gq9d 900000ms [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: SIP/6001-00000004: New call is still down.... Trying... [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] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:25:43] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6001 [Mar 21 17:25:43] DEBUG[10240] chan_sip.c: Checking device state for peer 6001 [Mar 21 17:25:43] DEBUG[10240] devicestate.c: Changing state for SIP/6001 - state 1 (Not in use) [Mar 21 17:25:43] DEBUG[10357][C-00000002] pbx.c: Result of 'EXTEN' is '6002' [Mar 21 17:25:43] DEBUG[10357][C-00000002] pbx.c: Launching 'Dial' [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] DEBUG[10357][C-00000002] chan_sip.c: Asked to create a SIP channel with formats: (ulaw) [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Allocating new SIP dialog for 1941a5817591a3691ef776535bc91b03@127.0.1.1:5060 - INVITE (No RTP) [Mar 21 17:25:43] DEBUG[10357][C-00000002] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f5dc800b488' [Mar 21 17:25:43] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Allocated port 14986 for RTP instance '0x7f5dc800b488' [Mar 21 17:25:43] DEBUG[10357][C-00000002] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:43] DEBUG[10357][C-00000002] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:25:43] DEBUG[10357][C-00000002] rtp_engine.c: RTP instance '0x7f5dc800b488' is setup and ready to go [Mar 21 17:25:43] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dc800b488' [Mar 21 17:25:43] VERBOSE[10357][C-00000002] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Setting NAT on RTP to Off [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Mar 21 17:25:43] DEBUG[10357][C-00000002] acl.c: For destination '10.24.18.138', our source address is '10.24.18.124'. [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Setting NAT on RTP to Off [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: SIP call-id changed from '1941a5817591a3691ef776535bc91b03@127.0.1.1:5060' to '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: *** Our native formats are (ulaw) [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: *** Joint capabilities are (ulaw) [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: *** Our capabilities are (ulaw) [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: *** Our preferred formats from the incoming channel are (ulaw) [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: This channel will not be able to handle video. [Mar 21 17:25:43] DEBUG[10357][C-00000002] channel_internal_api.c: Channel Call ID changing from [C-00000002] to [C-00000002] [Mar 21 17:25:43] DEBUG[10357][C-00000002] rtp_engine.c: Copying payload 0 from 0x7f5d6800bd68 to 0x7f5dc800b5c8 [Mar 21 17:25:43] DEBUG[10357][C-00000002] rtp_engine.c: Copying payload 8 from 0x7f5d6800bd68 to 0x7f5dc800b5c8 [Mar 21 17:25:43] DEBUG[10357][C-00000002] rtp_engine.c: Copying payload 9 from 0x7f5d6800bd68 to 0x7f5dc800b5c8 [Mar 21 17:25:43] DEBUG[10357][C-00000002] rtp_engine.c: Copying payload 96 from 0x7f5d6800bd68 to 0x7f5dc800b5c8 [Mar 21 17:25:43] DEBUG[10357][C-00000002] rtp_engine.c: Seeded SDP of 'SIP/6002-00000005' with that of 'SIP/6001-00000004' [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Outgoing Call for 6002 [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: ** Our capability: (ulaw) Video flag: False Text flag: False [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: ** Our prefcodec: (ulaw) [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] DEBUG[10357][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Initializing initreq for method INVITE - callid 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 0 [ 44]: INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK592fb047 [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 4 [ 35]: To: [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 6 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 9 [ 35]: Date: Fri, 21 Mar 2014 22:25:43 GMT [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10357][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #81 [Mar 21 17:25:43] DEBUG[10357][C-00000002] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:43] VERBOSE[10357][C-00000002] app_dial.c: -- Called SIP/6002 [Mar 21 17:25:43] DEBUG[10357][C-00000002] channel.c: SIP/6001-00000004: Dropping redundant connected line update "Bob" <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] DEBUG[10274] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK592fb047 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 2 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 4 [ 30]: To: [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Mar 21 17:25:43] VERBOSE[10274] chan_sip.c: --- (7 headers 0 lines) --- [Mar 21 17:25:43] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 (Checking To) --From tag as5656d5d0 --To-tag [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: *** SIP TIMER: Cancelling retransmission #81 - INVITE (got response) [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' Request 102: Found [Mar 21 17:25:43] DEBUG[10274][C-00000002] chan_sip.c: SIP response 100 to standard invite [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] DEBUG[10274] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 ringing [Mar 21 17:25:44] DEBUG[10274] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK592fb047 [Mar 21 17:25:44] DEBUG[10274] chan_sip.c: Header 2 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:44] DEBUG[10274] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:44] DEBUG[10274] chan_sip.c: Header 4 [ 67]: To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:44] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 21 17:25:44] DEBUG[10274] chan_sip.c: Header 6 [ 51]: Contact: "RustyTWO" [Mar 21 17:25:44] DEBUG[10274] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:44] DEBUG[10274] chan_sip.c: Header 8 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:25:44] DEBUG[10274] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Mar 21 17:25:44] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:25:44] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 (Checking To) --From tag as5656d5d0 --To-tag HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:44] DEBUG[10274][C-00000002] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' Request 102: Found [Mar 21 17:25:44] DEBUG[10274][C-00000002] chan_sip.c: SIP response 180 to standard invite [Mar 21 17:25:44] DEBUG[10274][C-00000002] chan_sip.c: build_route: Contact hop: "RustyTWO" [Mar 21 17:25:44] VERBOSE[10274][C-00000002] chan_sip.c: list_route: route/path hop: [Mar 21 17:25:44] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:25:44] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:25:44] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:25:44] VERBOSE[10357][C-00000002] app_dial.c: -- SIP/6002-00000005 is ringing [Mar 21 17:25:44] DEBUG[10357][C-00000002] rtp_engine.c: Setting early bridge SDP of 'SIP/6001-00000004' with that of 'SIP/6002-00000005' [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 17:25:44] DEBUG[10357][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:25:45] DEBUG[10357][C-00000002] res_rtp_asterisk.c: 0x7f5dc800f9d0 -- Probation learning mode pass with source address 10.24.18.138:4014 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK592fb047 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 2 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 4 [ 67]: To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 7 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 8 [ 51]: Contact: "RustyTWO" [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 10 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 12 [ 19]: Content-Length: 243 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 13 [ 0]: [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90812036 90812037 IN IP4 10.24.18.138 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4014 RTP/AVP 0 96 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4015 IN IP4 10.24.18.138 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:45] VERBOSE[10274] chan_sip.c: --- (13 headers 12 lines) --- [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 (Checking To) --From tag as5656d5d0 --To-tag HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Acked pending invite 102 [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Stopping retransmission on '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' of Request 102: Match Found [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: SIP response 200 to standard invite [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP o=- 90812036 90812037 IN IP4 10.24.18.138... OK. [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:25:45] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:25:45] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:45] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:45] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4015 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dc800b488' [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] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dc800b5c8 [Mar 21 17:25:45] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dc800b5c8 [Mar 21 17:25:45] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5dc800b488' [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: build_route: Contact hop: "RustyTWO" [Mar 21 17:25:45] VERBOSE[10274][C-00000002] chan_sip.c: list_route: route/path hop: [Mar 21 17:25:45] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:45] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Strict routing enforced for session 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [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] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:45] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:45] DEBUG[10357][C-00000002] channel.c: SIP/6001-00000004: Dropping redundant connected line update "Bob" <6002>. [Mar 21 17:25:45] VERBOSE[10357][C-00000002] app_dial.c: -- SIP/6002-00000005 answered SIP/6001-00000004 [Mar 21 17:25:45] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:25:45] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:25:45] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:25:45] DEBUG[10357][C-00000002] rtp_engine.c: Setting early bridge SDP of 'SIP/6001-00000004' with that of 'SIP/6002-00000005' [Mar 21 17:25:45] DEBUG[10357][C-00000002] chan_sip.c: SIP answering channel: SIP/6001-00000004 [Mar 21 17:25:45] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6001 [Mar 21 17:25:45] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Mar 21 17:25:45] DEBUG[10240] chan_sip.c: Checking device state for peer 6001 [Mar 21 17:25:45] DEBUG[10240] devicestate.c: Changing state for SIP/6001 - state 1 (Not in use) [Mar 21 17:25:45] DEBUG[10357][C-00000002] chan_sip.c: Setting framing from config on incoming call [Mar 21 17:25:45] DEBUG[10357][C-00000002] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:45] DEBUG[10357][C-00000002] chan_sip.c: ** Our prefcodec: (nothing) [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] DEBUG[10357][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:45] DEBUG[10357][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [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] DEBUG[10357][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #84 [Mar 21 17:25:45] DEBUG[10357][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:25:45] DEBUG[10357][C-00000002] features.c: Removing dialed interfaces datastore on SIP/6002-00000005 since we're bridging [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge_native_rtp.c: Bridge 'c283aedb-4fdf-4d87-97bd-58d291b1382c' can not use native RTP bridge as two channels are required [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Chose bridge technology simple_bridge [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling simple_bridge technology constructor [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling simple_bridge technology start [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge_channel.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: 0x7f5dc8014168(SIP/6001-00000004) is joining [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge_channel.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: 0x7f5dc801a1e8(SIP/6002-00000005) is joining [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge_channel.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: pushing 0x7f5dc8014168(SIP/6001-00000004) [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] DEBUG[10357][C-00000002] bridge_native_rtp.c: Bridge 'c283aedb-4fdf-4d87-97bd-58d291b1382c' can not use native RTP bridge as two channels are required [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Chose bridge technology simple_bridge [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c is already using the new technology. [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c is happy that channel SIP/6001-00000004 already has read format ulaw [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c is happy that channel SIP/6001-00000004 already has write format ulaw [Mar 21 17:25:45] DEBUG[10357][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: 0x7f5dc8014168(SIP/6001-00000004) is joining simple_bridge technology [Mar 21 17:25:45] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge_channel.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: pushing 0x7f5dc801a1e8(SIP/6002-00000005) [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] DEBUG[10242] cdr.c: Finalized CDR for SIP/6002-00000005 - start 1395440743.954297 answer 1395440745.714093 end 1395440745.715787 dispo ANSWERED [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Chose bridge technology native_rtp [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] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling native_rtp technology constructor [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling simple_bridge technology stop [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: 0x7f5dc8014168(SIP/6001-00000004) is leaving simple_bridge technology (dummy) [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c is happy that channel SIP/6001-00000004 already has read format ulaw [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c is happy that channel SIP/6001-00000004 already has write format ulaw [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: 0x7f5dc8014168(SIP/6001-00000004) is joining native_rtp technology [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Deferring reinvite on SIP 'XgkXgNogOMUJe2kUonThRDjEg.90Gq9d' - It's audio will be redirected to IP 10.24.18.138:4014 [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Sending reinvite on SIP '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' - It's audio soon redirected to IP 10.24.18.16:4022 [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Strict routing enforced for session 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [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] DEBUG[10358][C-00000002] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:45] DEBUG[10358][C-00000002] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10358][C-00000002] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: ** Our prefcodec: (ulaw) [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10358][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Initializing already initialized SIP dialog 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 (presumably reinvite) [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 0 [ 44]: INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK15650d62 [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 4 [ 72]: To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 6 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10358][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #85 [Mar 21 17:25:45] DEBUG[10358][C-00000002] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge_native_rtp.c: Remotely bridged 'SIP/6001-00000004' and 'SIP/6002-00000005' - media will flow directly between them [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c is happy that channel SIP/6002-00000005 already has read format ulaw [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c is happy that channel SIP/6002-00000005 already has write format ulaw [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: 0x7f5dc801a1e8(SIP/6002-00000005) is joining native_rtp technology [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge_native_rtp.c: Remotely bridged 'SIP/6001-00000004' and 'SIP/6002-00000005' - media will flow directly between them [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling native_rtp technology start [Mar 21 17:25:45] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling simple_bridge technology destructor [Mar 21 17:25:45] DEBUG[10358][C-00000002] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Mar 21 17:25:45] DEBUG[10358][C-00000002] res_rtp_asterisk.c: 0x7f5dc800f9d0 -- Probation learning mode pass with source address 10.24.18.138:4014 [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] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Mar 21 17:25:45] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Mar 21 17:25:45] DEBUG[10357][C-00000002] res_rtp_asterisk.c: 0x7f5d68005f40 -- Probation learning mode pass with source address 10.24.18.16:4022 [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] DEBUG[10358][C-00000002] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Mar 21 17:25:45] DEBUG[10358][C-00000002] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: ACK sip:6002@10.24.18.124:5060 SIP/2.0 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjeUPk8jKfWRBFLK0ilXBPsbA0iwkfwkU0 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 4 [ 42]: To: ;tag=as4a9b6f65 [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 6 [ 14]: CSeq: 7928 ACK [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 21 17:25:45] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:25:45] DEBUG[10274] chan_sip.c: = Looking for Call ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d (Checking From) --From tag Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF --To-tag as4a9b6f65 [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #84 [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Stopping retransmission on 'XgkXgNogOMUJe2kUonThRDjEg.90Gq9d' of Response 7928: Match Found [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Sending pending reinvite on 'XgkXgNogOMUJe2kUonThRDjEg.90Gq9d' [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Strict routing enforced for session XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [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] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:25:45] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: ** Our prefcodec: (nothing) [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10274][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Initializing already initialized SIP dialog XgkXgNogOMUJe2kUonThRDjEg.90Gq9d (presumably reinvite) [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 0 [ 43]: INVITE sip:6001@10.24.18.16:5060;ob SIP/2.0 [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK55f0b6f7;rport [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 3 [ 44]: From: ;tag=as4a9b6f65 [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 6 [ 41]: Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Header 14 [ 29]: Content-Type: application/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 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #86 [Mar 21 17:25:45] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.16:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK15650d62 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 2 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 4 [ 67]: To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 6 [ 51]: Contact: "RustyTWO" [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Content-Length: 243 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 11 [ 0]: [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90812036 90812038 IN IP4 10.24.18.138 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4014 RTP/AVP 0 96 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4015 IN IP4 10.24.18.138 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:46] VERBOSE[10274] chan_sip.c: --- (11 headers 12 lines) --- [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 (Checking To) --From tag as5656d5d0 --To-tag HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Acked pending invite 103 [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #85 [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Stopping retransmission on '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' of Request 103: Match Found [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP o=- 90812036 90812038 IN IP4 10.24.18.138... OK. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:46] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:46] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4015 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dc800b5c8 [Mar 21 17:25:46] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dc800b5c8 [Mar 21 17:25:46] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dc800b488' [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Strict routing enforced for session 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [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] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK55f0b6f7 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as4a9b6f65 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 6 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 7 [ 50]: Contact: "RustyONE" [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 11 [ 19]: Content-Length: 240 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 12 [ 0]: [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 1 [ 40]: o=- 90812036 90812037 IN IP4 10.24.18.16 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4022 RTP/AVP 0 96 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 7 [ 30]: a=rtcp:4023 IN IP4 10.24.18.16 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:46] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: = Looking for Call ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d (Checking To) --From tag as4a9b6f65 --To-tag Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Acked pending invite 102 [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #86 [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Stopping retransmission on 'XgkXgNogOMUJe2kUonThRDjEg.90Gq9d' of Request 102: Match Found [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: SIP response 200 to RE-invite on outgoing call XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP o=- 90812036 90812037 IN IP4 10.24.18.16... OK. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16' into... [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port ''. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.16... OK. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:46] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:46] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4023 IN IP4 10.24.18.16... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5d6800bd68 [Mar 21 17:25:46] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5d6800bd68 [Mar 21 17:25:46] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6800bc28' [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Updating call counter for incoming call [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Session-Expires: 1800 [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Refresher: UAC [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Session timer stopped: 79 - XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Session timer started: 87 - XgkXgNogOMUJe2kUonThRDjEg.90Gq9d 900000ms [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Strict routing enforced for session XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [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] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Sending reinvite on SIP '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' - It's audio soon redirected to IP 10.24.18.16:4022 [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Strict routing enforced for session 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [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] DEBUG[10358][C-00000002] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:46] DEBUG[10358][C-00000002] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10358][C-00000002] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: ** Our prefcodec: (ulaw) [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10358][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Initializing already initialized SIP dialog 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 (presumably reinvite) [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 0 [ 44]: INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK2a4e65fa [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 4 [ 72]: To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 6 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10358][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #88 [Mar 21 17:25:46] DEBUG[10358][C-00000002] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK2a4e65fa [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 2 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 4 [ 67]: To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 6 [ 51]: Contact: "RustyTWO" [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Content-Length: 243 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Header 11 [ 0]: [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90812036 90812039 IN IP4 10.24.18.138 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4014 RTP/AVP 0 96 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4015 IN IP4 10.24.18.138 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:46] VERBOSE[10274] chan_sip.c: --- (11 headers 12 lines) --- [Mar 21 17:25:46] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 (Checking To) --From tag as5656d5d0 --To-tag HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Acked pending invite 104 [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #88 [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Stopping retransmission on '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' of Request 104: Match Found [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP o=- 90812036 90812039 IN IP4 10.24.18.138... OK. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:46] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:46] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:46] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4015 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dc800b5c8 [Mar 21 17:25:46] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dc800b5c8 [Mar 21 17:25:46] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dc800b488' [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Strict routing enforced for session 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [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] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:46] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port '5060'. [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 17:25:46] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 41]: INVITE sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPj7Z8t4PoWlL9WQ15erJWpeBg8f5csmsGe [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 3 [ 69]: From: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 4 [ 50]: To: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 5 [ 51]: Contact: "RustyTWO" [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 6 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 7 [ 18]: CSeq: 24145 INVITE [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 12 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 14 [ 19]: Content-Length: 303 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 15 [ 0]: [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90812036 90812040 IN IP4 10.24.18.138 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 6 [ 10]: a=sendonly [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 7 [ 29]: m=audio 4014 RTP/AVP 0 8 9 96 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 8 [ 31]: a=rtcp:4015 IN IP4 10.24.18.138 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 10 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 11 [ 20]: a=rtpmap:9 G722/8000 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 12 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 13 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Body 14 [ 10]: a=sendonly [Mar 21 17:25:51] VERBOSE[10274] chan_sip.c: --- (15 headers 15 lines) --- [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 (Checking From) --From tag HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt --To-tag as5656d5d0 [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 21 17:25:51] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, timer, norefersub" [Mar 21 17:25:51] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Found SIP option: -replaces- [Mar 21 17:25:51] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Matched SIP option: replaces [Mar 21 17:25:51] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Found SIP option: -100rel- [Mar 21 17:25:51] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Matched SIP option: 100rel [Mar 21 17:25:51] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Found SIP option: -timer- [Mar 21 17:25:51] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Matched SIP option: timer [Mar 21 17:25:51] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Found SIP option: -norefersub- [Mar 21 17:25:51] DEBUG[10274][C-00000002] sip/reqresp_parser.c: Matched SIP option: norefersub [Mar 21 17:25:51] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:51] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: Initializing initreq for method INVITE - callid 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP o=- 90812036 90812040 IN IP4 10.24.18.138... OK. [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:25:51] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:25:51] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP a=sendonly... OK. [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:51] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f8590 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 8 [Mar 21 17:25:51] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 8 based on m type on 0x7f5d818f8590 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 9 [Mar 21 17:25:51] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 9 based on m type on 0x7f5d818f8590 [Mar 21 17:25:51] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:51] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f8590 [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4015 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [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] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dc800b488' [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] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 0 from 0x7f5d818f8590 to 0x7f5dc800b5c8 [Mar 21 17:25:51] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 8 from 0x7f5d818f8590 to 0x7f5dc800b5c8 [Mar 21 17:25:51] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 9 from 0x7f5d818f8590 to 0x7f5dc800b5c8 [Mar 21 17:25:51] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 96 from 0x7f5d818f8590 to 0x7f5dc800b5c8 [Mar 21 17:25:51] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5dc800b488' [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:25:51] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dc800b488' [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Got a SIP re-invite for call 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Incoming INVITE with 'timer' option supported [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: INVITE also has "Session-Expires" header. [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Session-Expires: 1800 [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: INVITE also has "Min-SE" header. [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Received Min-SE: 90 [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: SIP/6002-00000005: This call is UP.... [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] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Setting framing from config on incoming call [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: ** Our prefcodec: (ulaw) [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] DEBUG[10274][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [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] DEBUG[10274][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #89 [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:51] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6800bc28' [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Sending reinvite on SIP 'XgkXgNogOMUJe2kUonThRDjEg.90Gq9d' - It's audio soon redirected to IP 10.24.18.124:5060 [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Strict routing enforced for session XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [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] DEBUG[10357][C-00000002] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:25:51] DEBUG[10357][C-00000002] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10357][C-00000002] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: ** Our prefcodec: (nothing) [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] DEBUG[10357][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Initializing already initialized SIP dialog XgkXgNogOMUJe2kUonThRDjEg.90Gq9d (presumably reinvite) [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 0 [ 43]: INVITE sip:6001@10.24.18.16:5060;ob SIP/2.0 [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK0a49f001;rport [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 3 [ 44]: From: ;tag=as4a9b6f65 [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 6 [ 41]: Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Header 14 [ 29]: Content-Type: application/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] DEBUG[10357][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #90 [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:25:51] DEBUG[10357][C-00000002] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/6001-00000004' and 'SIP/6002-00000005' - media will flow through Asterisk core [Mar 21 17:25:51] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [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] DEBUG[10357][C-00000002] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Mar 21 17:25:51] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Mar 21 17:25:51] DEBUG[10357][C-00000002] chan_sip.c: Deferring reinvite on SIP 'XgkXgNogOMUJe2kUonThRDjEg.90Gq9d' - It's audio will be redirected to IP (null) [Mar 21 17:25:51] DEBUG[10357][C-00000002] channel.c: Set channel SIP/6001-00000004 to write format slin [Mar 21 17:25:51] DEBUG[10357][C-00000002] res_musiconhold.c: SIP/6001-00000004 Opened file 0 '/var/lib/asterisk/moh/macroform-cold_day' [Mar 21 17:25:51] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Difference is 48272, ms is 6054 [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: ACK sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjjCl9j7osTxzrobciXFOKiVfVcdWdjxaE [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 3 [ 69]: From: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 4 [ 50]: To: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 6 [ 15]: CSeq: 24145 ACK [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 21 17:25:51] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:25:51] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 (Checking From) --From tag HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt --To-tag as5656d5d0 [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #89 [Mar 21 17:25:51] DEBUG[10274][C-00000002] chan_sip.c: Stopping retransmission on '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' of Response 24145: Match Found [Mar 21 17:25:52] DEBUG[10357][C-00000002] res_rtp_asterisk.c: 0x7f5d68005f40 -- Probation learning mode pass with source address 10.24.18.16:4022 [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] DEBUG[10357][C-00000002] channel.c: Generator got voice, switching to phase locked mode [Mar 21 17:25:52] DEBUG[10357][C-00000002] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1497141799 to 773689287 due to a source change [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK0a49f001 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as4a9b6f65 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 6 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 7 [ 50]: Contact: "RustyONE" [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 11 [ 19]: Content-Length: 240 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 12 [ 0]: [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 1 [ 40]: o=- 90812036 90812038 IN IP4 10.24.18.16 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4022 RTP/AVP 0 96 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 7 [ 30]: a=rtcp:4023 IN IP4 10.24.18.16 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:52] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: = Looking for Call ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d (Checking To) --From tag as4a9b6f65 --To-tag Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Acked pending invite 103 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #90 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Stopping retransmission on 'XgkXgNogOMUJe2kUonThRDjEg.90Gq9d' of Request 103: Match Found [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: SIP response 200 to RE-invite on outgoing call XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP o=- 90812036 90812038 IN IP4 10.24.18.16... OK. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:25:52] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16' into... [Mar 21 17:25:52] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port ''. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.16... OK. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:52] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:52] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4023 IN IP4 10.24.18.16... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5d6800bd68 [Mar 21 17:25:52] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5d6800bd68 [Mar 21 17:25:52] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6800bc28' [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Updating call counter for incoming call [Mar 21 17:25:52] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:25:52] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Session-Expires: 1800 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Refresher: UAC [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Session timer stopped: 87 - XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Session timer started: 91 - XgkXgNogOMUJe2kUonThRDjEg.90Gq9d 900000ms [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Strict routing enforced for session XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [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] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:25:52] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Sending pending reinvite on 'XgkXgNogOMUJe2kUonThRDjEg.90Gq9d' [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Strict routing enforced for session XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [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] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:25:52] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: ** Our prefcodec: (nothing) [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] DEBUG[10274][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Initializing already initialized SIP dialog XgkXgNogOMUJe2kUonThRDjEg.90Gq9d (presumably reinvite) [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 0 [ 43]: INVITE sip:6001@10.24.18.16:5060;ob SIP/2.0 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK029e8025;rport [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 3 [ 44]: From: ;tag=as4a9b6f65 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 6 [ 41]: Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Header 14 [ 29]: Content-Type: application/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] DEBUG[10274][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #92 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:25:52] DEBUG[10357][C-00000002] res_rtp_asterisk.c: 0x7f5d68005f40 -- Probation learning mode pass with source address 10.24.18.16:4022 [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] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f5d6800bc28' [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: Changing ssrc from 773689287 to 852223630 due to a source change [Mar 21 17:25:52] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Difference is 1752, ms is 239 [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK029e8025 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as4a9b6f65 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 6 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 7 [ 50]: Contact: "RustyONE" [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 11 [ 19]: Content-Length: 240 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Header 12 [ 0]: [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 1 [ 40]: o=- 90812036 90812039 IN IP4 10.24.18.16 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4022 RTP/AVP 0 96 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 7 [ 30]: a=rtcp:4023 IN IP4 10.24.18.16 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:52] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:25:52] DEBUG[10274] chan_sip.c: = Looking for Call ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d (Checking To) --From tag as4a9b6f65 --To-tag Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Acked pending invite 104 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #92 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Stopping retransmission on 'XgkXgNogOMUJe2kUonThRDjEg.90Gq9d' of Request 104: Match Found [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: SIP response 200 to RE-invite on outgoing call XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP o=- 90812036 90812039 IN IP4 10.24.18.16... OK. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:25:52] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16' into... [Mar 21 17:25:52] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port ''. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.16... OK. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:52] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:52] VERBOSE[10274][C-00000002] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:52] DEBUG[10274][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4023 IN IP4 10.24.18.16... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5d6800bc28' [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] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5d6800bd68 [Mar 21 17:25:52] DEBUG[10274][C-00000002] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5d6800bd68 [Mar 21 17:25:52] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5d6800bc28' [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Updating call counter for incoming call [Mar 21 17:25:52] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:25:52] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Session-Expires: 1800 [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Refresher: UAC [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Session timer stopped: 91 - XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Session timer started: 94 - XgkXgNogOMUJe2kUonThRDjEg.90Gq9d 900000ms [Mar 21 17:25:52] DEBUG[10274][C-00000002] chan_sip.c: Strict routing enforced for session XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [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] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:25:52] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:25:52] DEBUG[10357][C-00000002] res_rtp_asterisk.c: 0x7f5d68005f40 -- Probation learning mode pass with source address 10.24.18.16:4022 [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] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:52] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:53] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 36]: INVITE sip:6003@10.24.18.124 SIP/2.0 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjw7pegsyVY-B9abW1XjD7JCR0oIGAFMWw [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 4 [ 27]: To: [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 5 [ 51]: Contact: "RustyTWO" [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 7 [ 18]: CSeq: 28049 INVITE [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 12 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 13 [ 37]: X-Digium-Call-Hint: potentialTransfer [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 15 [ 19]: Content-Length: 291 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 16 [ 0]: [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90812046 90812046 IN IP4 10.24.18.138 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 6 [ 29]: m=audio 4016 RTP/AVP 0 8 9 96 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4017 IN IP4 10.24.18.138 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 10 [ 20]: a=rtpmap:9 G722/8000 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 11 [ 10]: a=sendrecv [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 12 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 13 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: --- (16 headers 14 lines) --- [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: = Looking for Call ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 (Checking From) --From tag t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 --To-tag [Mar 21 17:25:54] DEBUG[10274] acl.c: For destination '10.24.18.138', our source address is '10.24.18.124'. [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:25:54] DEBUG[10274] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:54] DEBUG[10274] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for lmADfYGV48idoojF8CraA6m0qd3bPTw1 - INVITE (No RTP) [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 21 17:25:54] DEBUG[10274][C-00000003] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, timer, norefersub" [Mar 21 17:25:54] DEBUG[10274][C-00000003] sip/reqresp_parser.c: Found SIP option: -replaces- [Mar 21 17:25:54] DEBUG[10274][C-00000003] sip/reqresp_parser.c: Matched SIP option: replaces [Mar 21 17:25:54] DEBUG[10274][C-00000003] sip/reqresp_parser.c: Found SIP option: -100rel- [Mar 21 17:25:54] DEBUG[10274][C-00000003] sip/reqresp_parser.c: Matched SIP option: 100rel [Mar 21 17:25:54] DEBUG[10274][C-00000003] sip/reqresp_parser.c: Found SIP option: -timer- [Mar 21 17:25:54] DEBUG[10274][C-00000003] sip/reqresp_parser.c: Matched SIP option: timer [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10274][C-00000003] sip/reqresp_parser.c: Found SIP option: -norefersub- [Mar 21 17:25:54] DEBUG[10274][C-00000003] sip/reqresp_parser.c: Matched SIP option: norefersub [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000003] chan_sip.c: Initializing initreq for method INVITE - callid lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Using INVITE request as basis request - lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #95 [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 33]: ACK sip:6003@10.24.18.124 SIP/2.0 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjw7pegsyVY-B9abW1XjD7JCR0oIGAFMWw [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 4 [ 42]: To: ;tag=as45476b2b [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 6 [ 15]: CSeq: 28049 ACK [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: = Looking for Call ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 (Checking From) --From tag t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 --To-tag as45476b2b [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #95 [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Stopping retransmission on 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' of Response 28049: Match Found [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 36]: INVITE sip:6003@10.24.18.124 SIP/2.0 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjc8qRbpNocXKgP-.mIxJwQTqPH28TiDV1 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 4 [ 27]: To: [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 5 [ 51]: Contact: "RustyTWO" [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 7 [ 18]: CSeq: 28050 INVITE [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 12 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 13 [ 37]: X-Digium-Call-Hint: potentialTransfer [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 14 [162]: Authorization: Digest username="6002", realm="asterisk", nonce="16cf0b93", uri="sip:6003@10.24.18.124", response="bff28b032e6436f2ab63614ad6b00c81", algorithm=MD5 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 16 [ 19]: Content-Length: 291 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 17 [ 0]: [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90812046 90812046 IN IP4 10.24.18.138 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 6 [ 29]: m=audio 4016 RTP/AVP 0 8 9 96 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4017 IN IP4 10.24.18.138 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 10 [ 20]: a=rtpmap:9 G722/8000 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 11 [ 10]: a=sendrecv [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 12 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Body 13 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: --- (17 headers 14 lines) --- [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: = Looking for Call ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 (Checking From) --From tag t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 --To-tag [Mar 21 17:25:54] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:54] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:25:54] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:54] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000003] chan_sip.c: Initializing initreq for method INVITE - callid lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Using INVITE request as basis request - lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274][C-00000003] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f5d6802c258' [Mar 21 17:25:54] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Allocated port 12276 for RTP instance '0x7f5d6802c258' [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:25:54] DEBUG[10274][C-00000003] rtp_engine.c: RTP instance '0x7f5d6802c258' is setup and ready to go [Mar 21 17:25:54] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6802c258' [Mar 21 17:25:54] VERBOSE[10274][C-00000003] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Setting NAT on RTP to Off [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP o=- 90812046 90812046 IN IP4 10.24.18.138... OK. [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:54] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f8590 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 8 [Mar 21 17:25:54] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 8 based on m type on 0x7f5d818f8590 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 9 [Mar 21 17:25:54] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 9 based on m type on 0x7f5d818f8590 [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:54] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f8590 [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4017 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5d6802c258' [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] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 0 from 0x7f5d818f8590 to 0x7f5d6802c398 [Mar 21 17:25:54] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 8 from 0x7f5d818f8590 to 0x7f5d6802c398 [Mar 21 17:25:54] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 9 from 0x7f5d818f8590 to 0x7f5d6802c398 [Mar 21 17:25:54] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 96 from 0x7f5d818f8590 to 0x7f5d6802c398 [Mar 21 17:25:54] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5d6802c258' [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Checking SIP call limits for device 6002 [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Updating call counter for incoming call [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:54] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274][C-00000003] chan_sip.c: Incoming INVITE with 'timer' option supported [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: INVITE also has "Session-Expires" header. [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Session-Expires: 1800 [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: INVITE also has "Min-SE" header. [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Received Min-SE: 90 [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: *** Our native formats are (ulaw) [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: *** Joint capabilities are (ulaw) [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: *** Our capabilities are (ulaw) [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: This channel will not be able to handle video. [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: build_route: Contact hop: "RustyTWO" [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: list_route: route/path hop: [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: Session timer started: 97 - lmADfYGV48idoojF8CraA6m0qd3bPTw1 900000ms [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: SIP/6002-00000006: New call is still down.... Trying... [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] DEBUG[10274][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:54] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:25:54] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:25:54] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:25:54] DEBUG[10359][C-00000003] pbx.c: Result of 'EXTEN' is '6003' [Mar 21 17:25:54] DEBUG[10359][C-00000003] pbx.c: Launching 'Dial' [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] DEBUG[10359][C-00000003] chan_sip.c: Asked to create a SIP channel with formats: (ulaw) [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Allocating new SIP dialog for 50c797a368bfee0578720530480468cd@127.0.1.1:5060 - INVITE (No RTP) [Mar 21 17:25:54] DEBUG[10359][C-00000003] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f5dd400a7a8' [Mar 21 17:25:54] DEBUG[10359][C-00000003] res_rtp_asterisk.c: Allocated port 19596 for RTP instance '0x7f5dd400a7a8' [Mar 21 17:25:54] DEBUG[10359][C-00000003] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:25:54] DEBUG[10359][C-00000003] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:25:54] DEBUG[10359][C-00000003] rtp_engine.c: RTP instance '0x7f5dd400a7a8' is setup and ready to go [Mar 21 17:25:54] DEBUG[10359][C-00000003] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dd400a7a8' [Mar 21 17:25:54] VERBOSE[10359][C-00000003] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Setting NAT on RTP to Off [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Mar 21 17:25:54] DEBUG[10359][C-00000003] acl.c: For destination '10.24.18.180', our source address is '10.24.18.124'. [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Setting NAT on RTP to Off [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: SIP call-id changed from '50c797a368bfee0578720530480468cd@127.0.1.1:5060' to '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: *** Our native formats are (ulaw) [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: *** Joint capabilities are (ulaw) [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: *** Our capabilities are (ulaw) [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: *** Our preferred formats from the incoming channel are (ulaw) [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: This channel will not be able to handle video. [Mar 21 17:25:54] DEBUG[10359][C-00000003] channel_internal_api.c: Channel Call ID changing from [C-00000003] to [C-00000003] [Mar 21 17:25:54] DEBUG[10359][C-00000003] rtp_engine.c: Copying payload 0 from 0x7f5d6802c398 to 0x7f5dd400a8e8 [Mar 21 17:25:54] DEBUG[10359][C-00000003] rtp_engine.c: Copying payload 8 from 0x7f5d6802c398 to 0x7f5dd400a8e8 [Mar 21 17:25:54] DEBUG[10359][C-00000003] rtp_engine.c: Copying payload 9 from 0x7f5d6802c398 to 0x7f5dd400a8e8 [Mar 21 17:25:54] DEBUG[10359][C-00000003] rtp_engine.c: Copying payload 96 from 0x7f5d6802c398 to 0x7f5dd400a8e8 [Mar 21 17:25:54] DEBUG[10359][C-00000003] rtp_engine.c: Seeded SDP of 'SIP/6003-00000007' with that of 'SIP/6002-00000006' [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Outgoing Call for 6003 [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: ** Our capability: (ulaw) Video flag: False Text flag: False [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: ** Our prefcodec: (ulaw) [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] DEBUG[10359][C-00000003] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Initializing initreq for method INVITE - callid 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 0 [ 41]: INVITE sip:6003@10.24.18.180:5060 SIP/2.0 [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK54a6425b [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 3 [ 50]: From: "Bob" ;tag=as1e36869b [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 4 [ 32]: To: [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 6 [ 59]: Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 9 [ 35]: Date: Fri, 21 Mar 2014 22:25:54 GMT [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10359][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #99 [Mar 21 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:25:54] VERBOSE[10359][C-00000003] app_dial.c: -- Called SIP/6003 [Mar 21 17:25:54] DEBUG[10359][C-00000003] channel.c: SIP/6002-00000006: Dropping redundant connected line update "Charlie" <6003>. [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK54a6425b [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as1e36869b [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=E3364A2C-61B601E7 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: = Looking for Call ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 (Checking To) --From tag as1e36869b --To-tag E3364A2C-61B601E7 [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: *** SIP TIMER: Cancelling retransmission #99 - INVITE (got response) [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' Request 102: Found [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: SIP response 100 to standard invite [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Auto destroying SIP dialog '21608363712124-1345167932089@10.24.18.166' [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Destroying SIP dialog 21608363712124-1345167932089@10.24.18.166 [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] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK54a6425b [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as1e36869b [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=E3364A2C-61B601E7 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 8 [ 34]: Allow-Events: talk,hold,conference [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 9 [ 19]: Accept-Language: en [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Mar 21 17:25:54] VERBOSE[10274] chan_sip.c: --- (11 headers 0 lines) --- [Mar 21 17:25:54] DEBUG[10274] chan_sip.c: = Looking for Call ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 (Checking To) --From tag as1e36869b --To-tag E3364A2C-61B601E7 [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' Request 102: Found [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: SIP response 180 to standard invite [Mar 21 17:25:54] DEBUG[10274][C-00000003] chan_sip.c: build_route: Contact hop: [Mar 21 17:25:54] VERBOSE[10274][C-00000003] chan_sip.c: list_route: route/path hop: [Mar 21 17:25:54] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6003 [Mar 21 17:25:54] DEBUG[10240] chan_sip.c: Checking device state for peer 6003 [Mar 21 17:25:54] DEBUG[10240] devicestate.c: Changing state for SIP/6003 - state 1 (Not in use) [Mar 21 17:25:54] VERBOSE[10359][C-00000003] app_dial.c: -- SIP/6003-00000007 is ringing [Mar 21 17:25:54] DEBUG[10359][C-00000003] rtp_engine.c: Setting early bridge SDP of 'SIP/6002-00000006' with that of 'SIP/6003-00000007' [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 17:25:54] DEBUG[10359][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:54] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK54a6425b [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as1e36869b [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=E3364A2C-61B601E7 [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 8 [ 26]: Supported: 100rel,replaces [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 12 [ 19]: Content-Length: 197 [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Header 13 [ 0]: [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Body 1 [ 45]: o=- 1395440752 1395440752 IN IP4 10.24.18.180 [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.180 [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Body 5 [ 25]: m=audio 2226 RTP/AVP 0 96 [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Body 6 [ 10]: a=sendrecv [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:55] DEBUG[10274] chan_sip.c: Body 8 [ 32]: 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] DEBUG[10274] chan_sip.c: = Looking for Call ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 (Checking To) --From tag as1e36869b --To-tag E3364A2C-61B601E7 [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: Acked pending invite 102 [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: Stopping retransmission on '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' of Request 102: Match Found [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: SIP response 200 to standard invite [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP o=- 1395440752 1395440752 IN IP4 10.24.18.180... OK. [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Mar 21 17:25:55] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.180' into... [Mar 21 17:25:55] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.180' and port ''. [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.180... OK. [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:55] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:55] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [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] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dd400a7a8' [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] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dd400a8e8 [Mar 21 17:25:55] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dd400a8e8 [Mar 21 17:25:55] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5dd400a7a8' [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: build_route: Contact hop: [Mar 21 17:25:55] VERBOSE[10274][C-00000003] chan_sip.c: list_route: route/path hop: [Mar 21 17:25:55] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:25:55] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:25:55] DEBUG[10274][C-00000003] chan_sip.c: Strict routing enforced for session 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [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] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:25:55] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10274][C-00000003] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:25:55] DEBUG[10359][C-00000003] channel.c: SIP/6002-00000006: Dropping redundant connected line update "Charlie" <6003>. [Mar 21 17:25:55] VERBOSE[10359][C-00000003] app_dial.c: -- SIP/6003-00000007 answered SIP/6002-00000006 [Mar 21 17:25:55] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6003 [Mar 21 17:25:55] DEBUG[10240] chan_sip.c: Checking device state for peer 6003 [Mar 21 17:25:55] DEBUG[10240] devicestate.c: Changing state for SIP/6003 - state 1 (Not in use) [Mar 21 17:25:55] DEBUG[10359][C-00000003] rtp_engine.c: Setting early bridge SDP of 'SIP/6002-00000006' with that of 'SIP/6003-00000007' [Mar 21 17:25:55] DEBUG[10359][C-00000003] chan_sip.c: SIP answering channel: SIP/6002-00000006 [Mar 21 17:25:55] DEBUG[10359][C-00000003] res_rtp_asterisk.c: Setting the marker bit due to a source update [Mar 21 17:25:55] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:25:55] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:25:55] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:25:55] DEBUG[10359][C-00000003] chan_sip.c: Setting framing from config on incoming call [Mar 21 17:25:55] DEBUG[10359][C-00000003] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:55] DEBUG[10359][C-00000003] chan_sip.c: ** Our prefcodec: (nothing) [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] DEBUG[10359][C-00000003] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:55] DEBUG[10359][C-00000003] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [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] DEBUG[10359][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #102 [Mar 21 17:25:55] DEBUG[10359][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:55] DEBUG[10359][C-00000003] features.c: Removing dialed interfaces datastore on SIP/6003-00000007 since we're bridging [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge_native_rtp.c: Bridge 'ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6' can not use native RTP bridge as two channels are required [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Chose bridge technology simple_bridge [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling simple_bridge technology constructor [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling simple_bridge technology start [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge_channel.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: 0x7f5dd4016198(SIP/6002-00000006) is joining [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge_channel.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: 0x7f5dd4017b08(SIP/6003-00000007) is joining [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge_channel.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: pushing 0x7f5dd4016198(SIP/6002-00000006) [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] DEBUG[10359][C-00000003] bridge_native_rtp.c: Bridge 'ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6' can not use native RTP bridge as two channels are required [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Chose bridge technology simple_bridge [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is already using the new technology. [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is happy that channel SIP/6002-00000006 already has read format ulaw [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is happy that channel SIP/6002-00000006 already has write format ulaw [Mar 21 17:25:55] DEBUG[10359][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: 0x7f5dd4016198(SIP/6002-00000006) is joining simple_bridge technology [Mar 21 17:25:55] DEBUG[10359][C-00000003] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge_channel.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: pushing 0x7f5dd4017b08(SIP/6003-00000007) [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] DEBUG[10360][C-00000003] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Chose bridge technology native_rtp [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] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling native_rtp technology constructor [Mar 21 17:25:55] DEBUG[10242] cdr.c: Finalized CDR for SIP/6003-00000007 - start 1395440754.785311 answer 1395440755.978664 end 1395440755.980529 dispo ANSWERED [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling simple_bridge technology stop [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: 0x7f5dd4016198(SIP/6002-00000006) is leaving simple_bridge technology (dummy) [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is happy that channel SIP/6002-00000006 already has read format ulaw [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is happy that channel SIP/6002-00000006 already has write format ulaw [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: 0x7f5dd4016198(SIP/6002-00000006) is joining native_rtp technology [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Deferring reinvite on SIP 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' - It's audio will be redirected to IP 10.24.18.180:2226 [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Sending reinvite on SIP '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' - It's audio soon redirected to IP 10.24.18.138:4016 [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Strict routing enforced for session 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [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] DEBUG[10360][C-00000003] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:25:55] DEBUG[10360][C-00000003] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10360][C-00000003] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: ** Our prefcodec: (ulaw) [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10360][C-00000003] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Initializing already initialized SIP dialog 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 (presumably reinvite) [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 0 [ 41]: INVITE sip:6003@10.24.18.180:5060 SIP/2.0 [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK2e718d62 [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 3 [ 50]: From: "Bob" ;tag=as1e36869b [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 4 [ 54]: To: ;tag=E3364A2C-61B601E7 [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 6 [ 59]: Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Header 12 [ 29]: Content-Type: application/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 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #103 [Mar 21 17:25:55] DEBUG[10360][C-00000003] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge_native_rtp.c: Remotely bridged 'SIP/6002-00000006' and 'SIP/6003-00000007' - media will flow directly between them [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is happy that channel SIP/6003-00000007 already has read format ulaw [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is happy that channel SIP/6003-00000007 already has write format ulaw [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: 0x7f5dd4017b08(SIP/6003-00000007) is joining native_rtp technology [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge_native_rtp.c: Remotely bridged 'SIP/6002-00000006' and 'SIP/6003-00000007' - media will flow directly between them [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling native_rtp technology start [Mar 21 17:25:55] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling simple_bridge technology destructor [Mar 21 17:25:55] DEBUG[10360][C-00000003] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Mar 21 17:25:55] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK2e718d62 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as1e36869b [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=E3364A2C-61B601E7 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 103 INVITE [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 8 [ 26]: Supported: 100rel,replaces [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 12 [ 19]: Content-Length: 197 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 13 [ 0]: [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 1 [ 45]: o=- 1395440752 1395440753 IN IP4 10.24.18.180 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.180 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 5 [ 25]: m=audio 2226 RTP/AVP 0 96 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 6 [ 10]: a=sendrecv [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 8 [ 32]: 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] DEBUG[10274] chan_sip.c: = Looking for Call ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 (Checking To) --From tag as1e36869b --To-tag E3364A2C-61B601E7 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Acked pending invite 103 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #103 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Stopping retransmission on '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' of Request 103: Match Found [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: SIP response 200 to RE-invite on outgoing call 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP o=- 1395440752 1395440753 IN IP4 10.24.18.180... OK. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.180' into... [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.180' and port ''. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.180... OK. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:56] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:56] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [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] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dd400a8e8 [Mar 21 17:25:56] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dd400a8e8 [Mar 21 17:25:56] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dd400a7a8' [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Strict routing enforced for session 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [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] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10274][C-00000003] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10360][C-00000003] res_rtp_asterisk.c: 0x7f5dd400f9d0 -- Probation learning mode pass with source address 10.24.18.180:2226 [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] DEBUG[10359][C-00000003] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Mar 21 17:25:56] DEBUG[10359][C-00000003] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10359][C-00000003] res_rtp_asterisk.c: 0x7f5d6803b500 -- Probation learning mode pass with source address 10.24.18.138:4016 [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] DEBUG[10360][C-00000003] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Mar 21 17:25:56] DEBUG[10360][C-00000003] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Mar 21 17:25:56] DEBUG[10360][C-00000003] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f5dd400a7a8' [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: ACK sip:6003@10.24.18.124:5060 SIP/2.0 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjOCWM3Y.gaYpoWCl45HulIPIpGurFHf9g [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 4 [ 42]: To: ;tag=as5b57c5ec [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 6 [ 15]: CSeq: 28050 ACK [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 21 17:25:56] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: = Looking for Call ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 (Checking From) --From tag t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 --To-tag as5b57c5ec [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #102 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Stopping retransmission on 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' of Response 28050: Match Found [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Sending pending reinvite on 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Strict routing enforced for session lmADfYGV48idoojF8CraA6m0qd3bPTw1 [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] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000003] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: ** Our prefcodec: (nothing) [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10274][C-00000003] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Initializing already initialized SIP dialog lmADfYGV48idoojF8CraA6m0qd3bPTw1 (presumably reinvite) [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 0 [ 44]: INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK756728b5;rport [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 3 [ 44]: From: ;tag=as5b57c5ec [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 4 [ 75]: To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 6 [ 41]: Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Header 14 [ 29]: Content-Type: application/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] DEBUG[10274][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #105 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK756728b5 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as5b57c5ec [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 6 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 7 [ 51]: Contact: "RustyTWO" [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 11 [ 19]: Content-Length: 243 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 12 [ 0]: [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90812046 90812047 IN IP4 10.24.18.138 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4016 RTP/AVP 0 96 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4017 IN IP4 10.24.18.138 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:56] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: = Looking for Call ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 (Checking To) --From tag as5b57c5ec --To-tag t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Acked pending invite 102 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #105 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Stopping retransmission on 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' of Request 102: Match Found [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: SIP response 200 to RE-invite on outgoing call lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP o=- 90812046 90812047 IN IP4 10.24.18.138... OK. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:56] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:56] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4017 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5d6802c398 [Mar 21 17:25:56] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5d6802c398 [Mar 21 17:25:56] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6802c258' [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Updating call counter for incoming call [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Session-Expires: 1800 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Refresher: UAC [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Session timer stopped: 97 - lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Session timer started: 106 - lmADfYGV48idoojF8CraA6m0qd3bPTw1 900000ms [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Strict routing enforced for session lmADfYGV48idoojF8CraA6m0qd3bPTw1 [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] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000003] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Sending reinvite on SIP '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' - It's audio soon redirected to IP 10.24.18.138:4016 [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Strict routing enforced for session 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [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] DEBUG[10360][C-00000003] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:25:56] DEBUG[10360][C-00000003] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10360][C-00000003] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: ** Our prefcodec: (ulaw) [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10360][C-00000003] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Initializing already initialized SIP dialog 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 (presumably reinvite) [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 0 [ 41]: INVITE sip:6003@10.24.18.180:5060 SIP/2.0 [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK28f62ab6 [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 3 [ 50]: From: "Bob" ;tag=as1e36869b [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 4 [ 54]: To: ;tag=E3364A2C-61B601E7 [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 6 [ 59]: Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10360][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #107 [Mar 21 17:25:56] DEBUG[10360][C-00000003] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK28f62ab6 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as1e36869b [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=E3364A2C-61B601E7 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 104 INVITE [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 8 [ 26]: Supported: 100rel,replaces [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 12 [ 19]: Content-Length: 197 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Header 13 [ 0]: [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 1 [ 45]: o=- 1395440752 1395440754 IN IP4 10.24.18.180 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.180 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 5 [ 25]: m=audio 2226 RTP/AVP 0 96 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 6 [ 10]: a=sendrecv [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:56] DEBUG[10274] chan_sip.c: Body 8 [ 32]: 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] DEBUG[10274] chan_sip.c: = Looking for Call ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 (Checking To) --From tag as1e36869b --To-tag E3364A2C-61B601E7 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Acked pending invite 104 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #107 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Stopping retransmission on '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' of Request 104: Match Found [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: SIP response 200 to RE-invite on outgoing call 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP o=- 1395440752 1395440754 IN IP4 10.24.18.180... OK. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.180' into... [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.180' and port ''. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.180... OK. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:56] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:56] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:56] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [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] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dd400a8e8 [Mar 21 17:25:56] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dd400a8e8 [Mar 21 17:25:56] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dd400a7a8' [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Strict routing enforced for session 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [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] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:25:56] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.180' and port '5060'. [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 17:25:56] DEBUG[10274][C-00000003] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:56] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:57] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b488' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 40]: REFER sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjZh9QmnwlqxhyOPbxAgRml.0BZG6R6FyQ [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 3 [ 69]: From: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 4 [ 50]: To: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 5 [ 51]: Contact: "RustyTWO" [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 6 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 7 [ 17]: CSeq: 24146 REFER [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 8 [ 12]: Event: refer [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 9 [ 12]: Expires: 600 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 10 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 11 [ 35]: Accept: message/sipfrag;version=2.0 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 12 [ 46]: Allow-Events: presence, message-summary, refer [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 13 [143]: Refer-To: [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 14 [ 39]: Referred-By: [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 15 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 16 [ 17]: Content-Length: 0 [Mar 21 17:25:58] VERBOSE[10274] chan_sip.c: --- (17 headers 0 lines) --- [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 (Checking From) --From tag HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt --To-tag as5656d5d0 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [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] DEBUG[10274][C-00000002] chan_sip.c: Attended transfer: Will use Replace-Call-ID : lmADfYGV48idoojF8CraA6m0qd3bPTw1 F-tag: t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 T-tag: as5b57c5ec [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] DEBUG[10274][C-00000002] chan_sip.c: SIP attended transfer: Transferer channel SIP/6002-00000005 [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] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Looking for callid lmADfYGV48idoojF8CraA6m0qd3bPTw1 (fromtag t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 totag as5b57c5ec) [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Matched INCOMING call - their tag is t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 Our tag is as5b57c5ec [Mar 21 17:25:58] DEBUG[10357][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Mar 21 17:25:58] DEBUG[10360][C-00000003] res_rtp_asterisk.c: Setting the marker bit due to a source update [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] DEBUG[10357][C-00000002] channel.c: Set channel SIP/6001-00000004 to write format ulaw [Mar 21 17:25:58] DEBUG[10357][C-00000002] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Moving 0x7f5dc8014168(SIP/6001-00000004) into bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 swapping with SIP/6002-00000006 [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge_channel.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: pulling 0x7f5dc8014168(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] DEBUG[10274][C-00000002] bridge_channel.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: 0x7f5dc8014168(SIP/6001-00000004) is leaving native_rtp technology [Mar 21 17:25:58] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dc800b488' [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/6001-00000004' and 'SIP/6002-00000005' - media will flow through Asterisk core [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge_channel.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: pushing 0x7f5dc8014168(SIP/6001-00000004) by swapping with 0x7f5dd4016198(SIP/6002-00000006) [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] DEBUG[10274][C-00000002] bridge_channel.c: Setting 0x7f5dd4016198(SIP/6002-00000006) state from:0 to:2 [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge_channel.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: pulling 0x7f5dd4016198(SIP/6002-00000006) [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] DEBUG[10274][C-00000002] bridge_channel.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: 0x7f5dd4016198(SIP/6002-00000006) is leaving native_rtp technology [Mar 21 17:25:58] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6802c258' [Mar 21 17:25:58] DEBUG[10242] cdr.c: Finalized CDR for SIP/6001-00000004 - start 1395440743.951954 answer 1395440745.714705 end 1395440758.660376 dispo ANSWERED [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Sending reinvite on SIP 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' - It's audio soon redirected to IP 10.24.18.124:5060 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Strict routing enforced for session lmADfYGV48idoojF8CraA6m0qd3bPTw1 [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] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:58] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: ** Our prefcodec: (nothing) [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] DEBUG[10274][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Initializing already initialized SIP dialog lmADfYGV48idoojF8CraA6m0qd3bPTw1 (presumably reinvite) [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 0 [ 44]: INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK411ea1b4;rport [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 3 [ 44]: From: ;tag=as5b57c5ec [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 4 [ 75]: To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 6 [ 41]: Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 14 [ 29]: Content-Type: application/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] DEBUG[10274][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #108 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/6002-00000006' and 'SIP/6001-00000004' - media will flow through Asterisk core [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge_native_rtp.c: Bridge 'ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6' can not use native RTP bridge as channel 'SIP/6003-00000007' has features which prevent it [Mar 21 17:25:58] DEBUG[10242] cdr.c: Finalized CDR for SIP/6002-00000006 - start 1395440754.782917 answer 1395440755.979139 end 1395440758.661037 dispo ANSWERED [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Chose bridge technology simple_bridge [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] DEBUG[10274][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling simple_bridge technology constructor [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling native_rtp technology stop [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: 0x7f5dd4017b08(SIP/6003-00000007) is leaving native_rtp technology (dummy) [Mar 21 17:25:58] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dd400a7a8' [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Sending reinvite on SIP '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' - It's audio soon redirected to IP 10.24.18.124:5060 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Strict routing enforced for session 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [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] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:25:58] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: ** Our prefcodec: (ulaw) [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] DEBUG[10274][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Initializing already initialized SIP dialog 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 (presumably reinvite) [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 0 [ 41]: INVITE sip:6003@10.24.18.180:5060 SIP/2.0 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK0e356a61 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 3 [ 50]: From: "Bob" ;tag=as1e36869b [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 4 [ 54]: To: ;tag=E3364A2C-61B601E7 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 6 [ 59]: Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 7 [ 16]: CSeq: 105 INVITE [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10274][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #109 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is happy that channel SIP/6003-00000007 already has read format ulaw [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is happy that channel SIP/6003-00000007 already has write format ulaw [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: 0x7f5dd4017b08(SIP/6003-00000007) is joining simple_bridge technology [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is happy that channel SIP/6001-00000004 already has read format ulaw [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is happy that channel SIP/6001-00000004 already has write format ulaw [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: 0x7f5dc8014168(SIP/6001-00000004) is joining simple_bridge technology [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling simple_bridge technology start [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling native_rtp technology destructor [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge_native_rtp.c: Bridge 'c283aedb-4fdf-4d87-97bd-58d291b1382c' can not use native RTP bridge as two channels are required [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Chose bridge technology simple_bridge [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] DEBUG[10274][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling simple_bridge technology constructor [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling native_rtp technology stop [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: 0x7f5dc801a1e8(SIP/6002-00000005) is leaving native_rtp technology (dummy) [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c is happy that channel SIP/6002-00000005 already has read format ulaw [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c is happy that channel SIP/6002-00000005 already has write format ulaw [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: 0x7f5dc801a1e8(SIP/6002-00000005) is joining simple_bridge technology [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling simple_bridge technology start [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling native_rtp technology destructor [Mar 21 17:25:58] DEBUG[10274][C-00000002] bridge_channel.c: Setting 0x7f5dc801a1e8(SIP/6002-00000005) state from:0 to:2 [Mar 21 17:25:58] DEBUG[10358][C-00000002] bridge_channel.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: pulling 0x7f5dc801a1e8(SIP/6002-00000005) [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Strict routing enforced for session 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [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] DEBUG[10358][C-00000002] bridge_channel.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: 0x7f5dc801a1e8(SIP/6002-00000005) is leaving simple_bridge technology [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] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: dissolving bridge with cause 16(Normal Clearing) [Mar 21 17:25:58] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:58] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: queueing action type:13 sub:1001 [Mar 21 17:25:58] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c is dissolved, not performing smart bridge operation. [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] DEBUG[10274][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #110 [Mar 21 17:25:58] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:58] DEBUG[10358][C-00000002] res_rtp_asterisk.c: Changing ssrc from 852223630 to 1162176200 due to a source change [Mar 21 17:25:58] DEBUG[10359][C-00000003] res_rtp_asterisk.c: Changing ssrc from 1256943942 to 1795518893 due to a source change [Mar 21 17:25:58] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: actually destroying basic bridge, nobody wants it anymore [Mar 21 17:25:58] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling basic bridge destructor [Mar 21 17:25:58] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling simple_bridge technology stop [Mar 21 17:25:58] DEBUG[10358][C-00000002] bridge.c: Bridge c283aedb-4fdf-4d87-97bd-58d291b1382c: calling simple_bridge technology destructor [Mar 21 17:25:58] DEBUG[10358][C-00000002] channel.c: Hanging up channel 'SIP/6002-00000005' [Mar 21 17:25:58] DEBUG[10358][C-00000002] chan_sip.c: update_call_counter(6002) - decrement call limit counter on hangup [Mar 21 17:25:58] DEBUG[10358][C-00000002] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:25:58] DEBUG[10358][C-00000002] chan_sip.c: Call to peer '6002' removed from call limit 0 [Mar 21 17:25:58] DEBUG[10358][C-00000002] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060. [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] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:25:58] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:25:58] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:25:58] DEBUG[10242] cdr.c: CDR for SIP/6002-00000005 is dialed and has no Party B; discarding [Mar 21 17:25:58] DEBUG[10359][C-00000003] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Mar 21 17:25:58] DEBUG[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] pbx.c: == Spawn extension (from-internal, 6003, 1) exited non-zero on 'SIP/6002-00000006' [Mar 21 17:25:58] DEBUG[10359][C-00000003] channel.c: Soft-Hanging up channel 'SIP/6002-00000006' [Mar 21 17:25:58] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:25:58] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:25:58] DEBUG[10359][C-00000003] channel.c: Hanging up channel 'SIP/6002-00000006' [Mar 21 17:25:58] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:25:58] DEBUG[10359][C-00000003] chan_sip.c: Hangup call SIP/6002-00000006, SIP callid lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:58] DEBUG[10359][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5d6802c258' [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] DEBUG[10359][C-00000003] chan_sip.c: Session timer stopped: 106 - lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:58] DEBUG[10242] cdr.c: Finalized CDR for SIP/6001-00000004 - start 1395440758.660422 answer 1395440758.660423 end 1395440758.663183 dispo ANSWERED [Mar 21 17:25:58] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:25:58] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:25:58] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(src)})' (from 'CSV_QUOTE(${CDR(src)})},"Destination":${CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)})' len 8) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(src) result is '6002' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(6002) result is '"6002"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dst)})' (from 'CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)})' len 8) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(dst) result is '6003' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(6003) result is '"6003"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dcontext)})' (from 'CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(dcontext)' (from 'CDR(dcontext)})' len 13) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(dcontext) result is 'from-internal' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(from-internal) result is '"from-internal"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(channel)})' (from 'CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)})' len 12) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(channel) result is 'SIP/6002-00000006' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6002-00000006) result is '"SIP/6002-00000006"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dstchannel)})' (from 'CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)})' len 15) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(dstchannel) result is 'SIP/6003-00000007' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6003-00000007) result is '"SIP/6003-00000007"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastapp)})' (from 'CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(lastapp)' (from 'CDR(lastapp)})' len 12) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(lastapp) result is 'Dial' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(Dial) result is '"Dial"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastdata)})' (from 'CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(lastdata)' (from 'CDR(lastdata)})' len 13) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(lastdata) result is 'SIP/6003,15' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6003,15) result is '"SIP/6003,15"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(start)})' (from 'CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 24) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)})' len 10) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(start) result is '2014-03-21 17:25:54' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:25:54) result is '"2014-03-21 17:25:54"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(answer)})' (from 'CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 25) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)})' len 11) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(answer) result is '2014-03-21 17:25:55' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:25:55) result is '"2014-03-21 17:25:55"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(end)})' (from 'CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)})' len 8) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(end) result is '2014-03-21 17:25:58' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:25:58) result is '"2014-03-21 17:25:58"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(duration,f)})' (from 'CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(duration,f)' (from 'CDR(duration,f)})' len 15) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(duration,f) result is '0.003000' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.003000) result is '"0.003000"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(billsec,f)})' (from 'CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(billsec,f)' (from 'CDR(billsec,f)})' len 14) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(billsec,f) result is '0.002000' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.002000) result is '"0.002000"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(disposition)})' (from 'CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)})' len 16) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(disposition) result is 'ANSWERED' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(ANSWERED) result is '"ANSWERED"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(amaflags)})' (from 'CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(amaflags)' (from 'CDR(amaflags)})' len 13) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(amaflags) result is 'DOCUMENTATION' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(DOCUMENTATION) result is '"DOCUMENTATION"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(accountcode)})' (from 'CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(accountcode)' (from 'CDR(accountcode)})' len 16) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(accountcode) result is '' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(uniqueid)})' (from 'CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)})' len 13) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(uniqueid) result is '1395440754.10' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE(1395440754.10) result is '"1395440754.10"' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(userfield)})' (from 'CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(userfield)' (from 'CDR(userfield)})' len 14) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(userfield) result is '' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(sequence)' (from 'CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 13) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(sequence) result is '8' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield1)})' (from 'CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield1)' (from 'CDR(acustomfield1)})' len 18) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(acustomfield1) result is '' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield2)})' (from 'CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield2)' (from 'CDR(acustomfield2)})' len 18) [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CDR(acustomfield2) result is '' [Mar 21 17:25:58] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:25:58] DEBUG[10360][C-00000003] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f5dd400a7a8' [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK0e356a61 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as1e36869b [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=E3364A2C-61B601E7 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 105 INVITE [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 8 [ 26]: Supported: 100rel,replaces [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 12 [ 19]: Content-Length: 197 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Header 13 [ 0]: [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Body 1 [ 45]: o=- 1395440752 1395440755 IN IP4 10.24.18.180 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.180 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Body 5 [ 25]: m=audio 2226 RTP/AVP 0 96 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Body 6 [ 10]: a=sendrecv [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:58] DEBUG[10274] chan_sip.c: Body 8 [ 32]: 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] DEBUG[10274] chan_sip.c: = Looking for Call ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 (Checking To) --From tag as1e36869b --To-tag E3364A2C-61B601E7 [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: Acked pending invite 105 [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #109 [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: Stopping retransmission on '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' of Request 105: Match Found [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: SIP response 200 to RE-invite on outgoing call 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP o=- 1395440752 1395440755 IN IP4 10.24.18.180... OK. [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Mar 21 17:25:58] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.180' into... [Mar 21 17:25:58] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.180' and port ''. [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.180... OK. [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:58] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:58] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:58] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:58] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [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] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dd400a7a8' [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] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dd400a8e8 [Mar 21 17:25:58] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dd400a8e8 [Mar 21 17:25:58] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5dd400a7a8' [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:25:58] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:25:58] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:25:58] DEBUG[10274][C-00000003] chan_sip.c: Strict routing enforced for session 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [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] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:25:58] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10274][C-00000003] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:25:58] DEBUG[10360][C-00000003] res_rtp_asterisk.c: 0x7f5dd400f9d0 -- Probation learning mode pass with source address 10.24.18.180:2226 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK411ea1b4 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as5b57c5ec [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 6 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 7 [ 51]: Contact: "RustyTWO" [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 11 [ 19]: Content-Length: 243 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 12 [ 0]: [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90812046 90812048 IN IP4 10.24.18.138 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4016 RTP/AVP 0 96 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4017 IN IP4 10.24.18.138 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: = Looking for Call ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 (Checking To) --From tag as5b57c5ec --To-tag t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Acked pending invite 103 [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #108 [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Stopping retransmission on 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' of Request 103: Match Found [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: SIP response 200 to RE-invite on outgoing call lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP o=- 90812046 90812048 IN IP4 10.24.18.138... OK. [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:25:59] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:25:59] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 0 [Mar 21 17:25:59] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f7a40 [Mar 21 17:25:59] VERBOSE[10274][C-00000003] chan_sip.c: Found RTP audio format 96 [Mar 21 17:25:59] DEBUG[10274][C-00000003] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f7a40 [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4017 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5d6802c258' [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] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 0 from 0x7f5d818f7a40 to 0x7f5d6802c398 [Mar 21 17:25:59] DEBUG[10274][C-00000003] rtp_engine.c: Copying payload 96 from 0x7f5d818f7a40 to 0x7f5d6802c398 [Mar 21 17:25:59] DEBUG[10274][C-00000003] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5d6802c258' [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Updating call counter for incoming call [Mar 21 17:25:59] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:59] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Session-Expires: 1800 [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Refresher: UAC [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Session timer started: 115 - lmADfYGV48idoojF8CraA6m0qd3bPTw1 900000ms [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Strict routing enforced for session lmADfYGV48idoojF8CraA6m0qd3bPTw1 [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] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:59] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000003] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Strict routing enforced for session lmADfYGV48idoojF8CraA6m0qd3bPTw1 [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] DEBUG[10274][C-00000003] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:59] DEBUG[10274][C-00000003] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #116 [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Trying to put 'BYE sip:600' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10274][C-00000003] chan_sip.c: Session timer stopped: 115 - lmADfYGV48idoojF8CraA6m0qd3bPTw1 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK28040381 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 2 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 4 [ 67]: To: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 105 NOTIFY [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 6 [ 51]: Contact: "RustyTWO" [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 (Checking To) --From tag as5656d5d0 --To-tag HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:59] DEBUG[10274][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #110 [Mar 21 17:25:59] DEBUG[10274][C-00000002] chan_sip.c: Stopping retransmission on '4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060' of Request 105: Match Found [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK439668d5 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as5b57c5ec [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyTWO" ;tag=t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: --- (7 headers 0 lines) --- [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: = Looking for Call ID: lmADfYGV48idoojF8CraA6m0qd3bPTw1 (Checking To) --From tag as5b57c5ec --To-tag t48C.hhaVOuaBbSTpqaFPmAud0PL2Ju9 [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #116 [Mar 21 17:25:59] DEBUG[10274][C-00000003] chan_sip.c: Stopping retransmission on 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' of Request 104: Match Found [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Destroying SIP dialog lmADfYGV48idoojF8CraA6m0qd3bPTw1 [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog 'lmADfYGV48idoojF8CraA6m0qd3bPTw1' Method: ACK [Mar 21 17:25:59] DEBUG[10274] rtp_engine.c: Destroyed RTP instance '0x7f5d6802c258' [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: BYE sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjnnOrdpCiAGdquReHJdV0idt0IQDzx9eW [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 3 [ 69]: From: ;tag=HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 4 [ 50]: To: "Alice" ;tag=as5656d5d0 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 6 [ 15]: CSeq: 24147 BYE [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 7 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Mar 21 17:25:59] VERBOSE[10274] chan_sip.c: --- (9 headers 0 lines) --- [Mar 21 17:25:59] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 (Checking From) --From tag HSpWJ8omPDuavyhO2vySiUOvOAOCdLmt --To-tag as5656d5d0 [Mar 21 17:25:59] DEBUG[10274][C-00000002] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Mar 21 17:25:59] DEBUG[10274][C-00000002] chan_sip.c: Initializing initreq for method BYE - callid 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:59] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:25:59] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4adb4ab67ffa2e1b78f07c4e267386ae@10.24.18.124:5060 [Mar 21 17:25:59] DEBUG[10274][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dc800b488' [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] DEBUG[10274][C-00000002] chan_sip.c: Received bye, no owner, selfdestruct soon. [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 17:25:59] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:26:00] DEBUG[10360][C-00000003] res_rtp_asterisk.c: Got RTCP report of 92 bytes [Mar 21 17:26:05] DEBUG[10360][C-00000003] res_rtp_asterisk.c: Got RTCP report of 92 bytes [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: REGISTER sip:10.24.18.124:5060 SIP/2.0 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPja.yRq.JQv446PTtoOhOggpyoEP8Lc1Uc [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyTWO" ;tag=VsHseI8XNiWkjZdDFg8CSgK4EBTFISyt [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 4 [ 38]: To: "RustyTWO" [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: 8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 6 [ 20]: CSeq: 44933 REGISTER [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 7 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 8 [ 51]: Contact: "RustyTWO" [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 9 [ 12]: Expires: 300 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 10 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: --- (12 headers 0 lines) --- [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: = Looking for Call ID: 8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL (Checking From) --From tag VsHseI8XNiWkjZdDFg8CSgK4EBTFISyt --To-tag [Mar 21 17:26:06] DEBUG[10274] acl.c: For destination '10.24.18.138', our source address is '10.24.18.124'. [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for 8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL - REGISTER (No RTP) [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Initializing initreq for method REGISTER - callid 8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: REGISTER sip:10.24.18.124:5060 SIP/2.0 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjL-O9UP8sAhyzpEmVJV0Z2betbx9LskCh [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyTWO" ;tag=VsHseI8XNiWkjZdDFg8CSgK4EBTFISyt [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 4 [ 38]: To: "RustyTWO" [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: 8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 6 [ 20]: CSeq: 44934 REGISTER [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 7 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 8 [ 51]: Contact: "RustyTWO" [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 9 [ 12]: Expires: 300 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 10 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 11 [162]: Authorization: Digest username="6002", realm="asterisk", nonce="52d40800", uri="sip:10.24.18.124:5060", response="1e4f93a32344e5fb73676e4d87574b3d", algorithm=MD5 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: --- (13 headers 0 lines) --- [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: = Looking for Call ID: 8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL (Checking From) --From tag VsHseI8XNiWkjZdDFg8CSgK4EBTFISyt --To-tag [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Initializing initreq for method REGISTER - callid 8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Store REGISTER's Contact header for call routing. [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: build_path: do not use Path headers [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:26:06] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:26:06] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [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] DEBUG[10274] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:6002@10.24.18.124:5060 SIP/2.0 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjykOCVgt-CMcCD7Lw90xKJn1ku690zP1c [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyTWO" ;tag=BMMI-ojYAS5hOUEjieRW3xGdrVs6C07o [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 4 [ 38]: To: "RustyTWO" [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 5 [ 51]: Contact: "RustyTWO" [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 7 [ 21]: CSeq: 21461 SUBSCRIBE [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 8 [ 22]: Event: message-summary [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 9 [ 13]: Expires: 3600 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 10 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 11 [ 42]: Accept: application/simple-message-summary [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 12 [ 46]: Allow-Events: presence, message-summary, refer [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 13 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: --- (15 headers 0 lines) --- [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: = Looking for Call ID: edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx (Checking From) --From tag BMMI-ojYAS5hOUEjieRW3xGdrVs6C07o --To-tag [Mar 21 17:26:06] DEBUG[10274] acl.c: For destination '10.24.18.138', our source address is '10.24.18.124'. [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx - SUBSCRIBE (No RTP) [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: build_route: Contact hop: "RustyTWO" [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: list_route: route/path hop: [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:6002@10.24.18.124:5060 SIP/2.0 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjpZYiiiQViK1Aq9cAoEVg8ZDMxRTONXDe [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyTWO" ;tag=BMMI-ojYAS5hOUEjieRW3xGdrVs6C07o [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 4 [ 38]: To: "RustyTWO" [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 5 [ 51]: Contact: "RustyTWO" [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 7 [ 21]: CSeq: 21462 SUBSCRIBE [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 8 [ 22]: Event: message-summary [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 9 [ 13]: Expires: 3600 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 10 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 11 [ 42]: Accept: application/simple-message-summary [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 12 [ 46]: Allow-Events: presence, message-summary, refer [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 13 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 14 [167]: Authorization: Digest username="6002", realm="asterisk", nonce="25d12960", uri="sip:6002@10.24.18.124:5060", response="3b75fdab09732f3a23112e532be7bad0", algorithm=MD5 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: --- (16 headers 0 lines) --- [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: = Looking for Call ID: edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx (Checking From) --From tag BMMI-ojYAS5hOUEjieRW3xGdrVs6C07o --To-tag [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Got a new subscription edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx (possibly with auth) or retransmission [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:26:06] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: build_route: Retaining previous route: [Mar 21 17:26:06] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:26:06] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:26:06] NOTICE[10274] chan_sip.c: Received SIP subscribe for peer without mailbox: 6002 [Mar 21 17:26:06] DEBUG[10274] chan_sip.c: Destroying SIP dialog edNnQZGWQqpH1o5P-A9B6iM9bN.GpaOx [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: BYE sip:6002@10.24.18.124:5060 SIP/2.0 [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPj21jT3x58vRsqYsakH78W0nN1byCGPtKc [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 4 [ 42]: To: ;tag=as4a9b6f65 [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 6 [ 14]: CSeq: 7929 BYE [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 7 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Mar 21 17:26:07] VERBOSE[10274] chan_sip.c: --- (9 headers 0 lines) --- [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: = Looking for Call ID: XgkXgNogOMUJe2kUonThRDjEg.90Gq9d (Checking From) --From tag Ef9augZO4j1iNO6LzGDTQxAYcrgzhJsF --To-tag as4a9b6f65 [Mar 21 17:26:07] DEBUG[10274][C-00000002] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Mar 21 17:26:07] DEBUG[10274][C-00000002] chan_sip.c: Initializing initreq for method BYE - callid XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:26:07] DEBUG[10274][C-00000002] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:26:07] DEBUG[10274][C-00000002] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000002] chan_sip.c: Setting SIP_ALREADYGONE on dialog XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:26:07] DEBUG[10274][C-00000002] chan_sip.c: Session timer stopped: 94 - XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [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] DEBUG[10274][C-00000002] chan_sip.c: Received bye, issuing owner hangup [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] DEBUG[10274][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:26:07] DEBUG[10357][C-00000002] bridge_channel.c: Setting 0x7f5dc8014168(SIP/6001-00000004) state from:0 to:1 [Mar 21 17:26:07] DEBUG[10357][C-00000002] bridge_channel.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: pulling 0x7f5dc8014168(SIP/6001-00000004) [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] DEBUG[10357][C-00000002] bridge_channel.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: 0x7f5dc8014168(SIP/6001-00000004) is leaving simple_bridge technology [Mar 21 17:26:07] DEBUG[10357][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: dissolving bridge with cause 16(Normal Clearing) [Mar 21 17:26:07] DEBUG[10357][C-00000002] bridge_channel.c: Setting 0x7f5dd4017b08(SIP/6003-00000007) state from:0 to:2 [Mar 21 17:26:07] DEBUG[10357][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: queueing action type:13 sub:1001 [Mar 21 17:26:07] DEBUG[10357][C-00000002] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is dissolved, not performing smart bridge operation. [Mar 21 17:26:07] DEBUG[10242] cdr.c: Finalized CDR for SIP/6001-00000004 - start 1395440758.660436 answer 1395440758.660436 end 1395440767.353965 dispo ANSWERED [Mar 21 17:26:07] DEBUG[10360][C-00000003] bridge_channel.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: pulling 0x7f5dd4017b08(SIP/6003-00000007) [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] DEBUG[10360][C-00000003] bridge_channel.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: 0x7f5dd4017b08(SIP/6003-00000007) is leaving simple_bridge technology [Mar 21 17:26:07] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6 is dissolved, not performing smart bridge operation. [Mar 21 17:26:07] DEBUG[10360][C-00000003] res_rtp_asterisk.c: Changing ssrc from 1625324671 to 1265400895 due to a source change [Mar 21 17:26:07] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: actually destroying basic bridge, nobody wants it anymore [Mar 21 17:26:07] DEBUG[10357][C-00000002] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Mar 21 17:26:07] DEBUG[10357][C-00000002] pbx.c: Spawn extension (from-internal,6002,1) exited non-zero on 'SIP/6001-00000004' [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] DEBUG[10357][C-00000002] channel.c: Soft-Hanging up channel 'SIP/6001-00000004' [Mar 21 17:26:07] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling basic bridge destructor [Mar 21 17:26:07] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling simple_bridge technology stop [Mar 21 17:26:07] DEBUG[10360][C-00000003] bridge.c: Bridge ce452fd0-21a3-4c4c-b01c-b89d6c17d2a6: calling simple_bridge technology destructor [Mar 21 17:26:07] DEBUG[10360][C-00000003] channel.c: Hanging up channel 'SIP/6003-00000007' [Mar 21 17:26:07] DEBUG[10357][C-00000002] channel.c: Hanging up channel 'SIP/6001-00000004' [Mar 21 17:26:07] DEBUG[10360][C-00000003] chan_sip.c: Hangup call SIP/6003-00000007, SIP callid 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:26:07] DEBUG[10357][C-00000002] chan_sip.c: Hangup call SIP/6001-00000004, SIP callid XgkXgNogOMUJe2kUonThRDjEg.90Gq9d [Mar 21 17:26:07] DEBUG[10360][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dd400a7a8' [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] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6001 [Mar 21 17:26:07] DEBUG[10240] chan_sip.c: Checking device state for peer 6001 [Mar 21 17:26:07] DEBUG[10240] devicestate.c: Changing state for SIP/6001 - state 1 (Not in use) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(src)})' (from 'CSV_QUOTE(${CDR(src)})},"Destination":${CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)})' len 8) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(src) result is '6001' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(6001) result is '"6001"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dst)})' (from 'CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)})' len 8) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(dst) result is '6002' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(6002) result is '"6002"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dcontext)})' (from 'CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(dcontext)' (from 'CDR(dcontext)})' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(dcontext) result is 'from-internal' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(from-internal) result is '"from-internal"' [Mar 21 17:26:07] DEBUG[10360][C-00000003] chan_sip.c: Strict routing enforced for session 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(channel)})' (from 'CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)})' len 12) [Mar 21 17:26:07] VERBOSE[10360][C-00000003] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:26:07] DEBUG[10360][C-00000003] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:26:07] DEBUG[10360][C-00000003] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(channel) result is 'SIP/6001-00000004' [Mar 21 17:26:07] VERBOSE[10360][C-00000003] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6001-00000004) result is '"SIP/6001-00000004"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dstchannel)})' (from 'CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)})' len 15) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(dstchannel) result is 'SIP/6002-00000005' [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] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6002-00000005) result is '"SIP/6002-00000005"' [Mar 21 17:26:07] DEBUG[10360][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #125 [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastapp)})' (from 'CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:26:07] DEBUG[10360][C-00000003] chan_sip.c: Trying to put 'BYE sip:600' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(lastapp)' (from 'CDR(lastapp)})' len 12) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(lastapp) result is 'Dial' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(Dial) result is '"Dial"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastdata)})' (from 'CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(lastdata)' (from 'CDR(lastdata)})' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(lastdata) result is 'SIP/6002,15' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6002,15) result is '"SIP/6002,15"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(start)})' (from 'CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 24) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)})' len 10) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(start) result is '2014-03-21 17:25:43' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:25:43) result is '"2014-03-21 17:25:43"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(answer)})' (from 'CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 25) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)})' len 11) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(answer) result is '2014-03-21 17:25:45' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:25:45) result is '"2014-03-21 17:25:45"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(end)})' (from 'CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)})' len 8) [Mar 21 17:26:07] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6003 [Mar 21 17:26:07] DEBUG[10240] chan_sip.c: Checking device state for peer 6003 [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(end) result is '2014-03-21 17:25:58' [Mar 21 17:26:07] DEBUG[10240] devicestate.c: Changing state for SIP/6003 - state 1 (Not in use) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:25:58) result is '"2014-03-21 17:25:58"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(duration,f)})' (from 'CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(duration,f)' (from 'CDR(duration,f)})' len 15) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(duration,f) result is '0.014000' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.014000) result is '"0.014000"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(billsec,f)})' (from 'CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(billsec,f)' (from 'CDR(billsec,f)})' len 14) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(billsec,f) result is '0.012000' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.012000) result is '"0.012000"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(disposition)})' (from 'CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)})' len 16) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(disposition) result is 'ANSWERED' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(ANSWERED) result is '"ANSWERED"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(amaflags)})' (from 'CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(amaflags)' (from 'CDR(amaflags)})' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(amaflags) result is 'DOCUMENTATION' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(DOCUMENTATION) result is '"DOCUMENTATION"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(accountcode)})' (from 'CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(accountcode)' (from 'CDR(accountcode)})' len 16) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(accountcode) result is '' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(uniqueid)})' (from 'CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)})' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(uniqueid) result is '1395440743.8' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(1395440743.8) result is '"1395440743.8"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(userfield)})' (from 'CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(userfield)' (from 'CDR(userfield)})' len 14) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(userfield) result is '' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(sequence)' (from 'CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(sequence) result is '6' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield1)})' (from 'CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield1)' (from 'CDR(acustomfield1)})' len 18) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(acustomfield1) result is '' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield2)})' (from 'CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield2)' (from 'CDR(acustomfield2)})' len 18) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(acustomfield2) result is '' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(src)})' (from 'CSV_QUOTE(${CDR(src)})},"Destination":${CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)})' len 8) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(src) result is '6001' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(6001) result is '"6001"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dst)})' (from 'CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)})' len 8) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(dst) result is '6002' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(6002) result is '"6002"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dcontext)})' (from 'CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(dcontext)' (from 'CDR(dcontext)})' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(dcontext) result is 'from-internal' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(from-internal) result is '"from-internal"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(channel)})' (from 'CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)})' len 12) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(channel) result is 'SIP/6001-00000004' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6001-00000004) result is '"SIP/6001-00000004"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dstchannel)})' (from 'CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)})' len 15) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(dstchannel) result is 'SIP/6002-00000006' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6002-00000006) result is '"SIP/6002-00000006"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastapp)})' (from 'CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(lastapp)' (from 'CDR(lastapp)})' len 12) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(lastapp) result is 'Dial' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(Dial) result is '"Dial"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastdata)})' (from 'CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(lastdata)' (from 'CDR(lastdata)})' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(lastdata) result is 'SIP/6002,15' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6002,15) result is '"SIP/6002,15"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(start)})' (from 'CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 24) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)})' len 10) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(start) result is '2014-03-21 17:25:58' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:25:58) result is '"2014-03-21 17:25:58"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(answer)})' (from 'CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 25) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)})' len 11) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(answer) result is '2014-03-21 17:25:58' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:25:58) result is '"2014-03-21 17:25:58"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(end)})' (from 'CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)})' len 8) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(end) result is '2014-03-21 17:25:58' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:25:58) result is '"2014-03-21 17:25:58"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(duration,f)})' (from 'CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(duration,f)' (from 'CDR(duration,f)})' len 15) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(duration,f) result is '0.000000' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.000000) result is '"0.000000"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(billsec,f)})' (from 'CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(billsec,f)' (from 'CDR(billsec,f)})' len 14) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(billsec,f) result is '0.000000' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.000000) result is '"0.000000"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(disposition)})' (from 'CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)})' len 16) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(disposition) result is 'ANSWERED' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(ANSWERED) result is '"ANSWERED"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(amaflags)})' (from 'CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(amaflags)' (from 'CDR(amaflags)})' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(amaflags) result is 'DOCUMENTATION' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(DOCUMENTATION) result is '"DOCUMENTATION"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(accountcode)})' (from 'CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(accountcode)' (from 'CDR(accountcode)})' len 16) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(accountcode) result is '' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(uniqueid)})' (from 'CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)})' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(uniqueid) result is '1395440743.8' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(1395440743.8) result is '"1395440743.8"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(userfield)})' (from 'CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(userfield)' (from 'CDR(userfield)})' len 14) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(userfield) result is '' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(sequence)' (from 'CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(sequence) result is '10' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield1)})' (from 'CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield1)' (from 'CDR(acustomfield1)})' len 18) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(acustomfield1) result is '' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield2)})' (from 'CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield2)' (from 'CDR(acustomfield2)})' len 18) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(acustomfield2) result is '' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(src)})' (from 'CSV_QUOTE(${CDR(src)})},"Destination":${CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)})' len 8) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(src) result is '6001' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(6001) result is '"6001"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dst)})' (from 'CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)})' len 8) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(dst) result is '6002' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(6002) result is '"6002"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dcontext)})' (from 'CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(dcontext)' (from 'CDR(dcontext)})' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(dcontext) result is 'from-internal' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(from-internal) result is '"from-internal"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(channel)})' (from 'CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)})' len 12) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(channel) result is 'SIP/6001-00000004' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6001-00000004) result is '"SIP/6001-00000004"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dstchannel)})' (from 'CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)})' len 15) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(dstchannel) result is 'SIP/6003-00000007' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6003-00000007) result is '"SIP/6003-00000007"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastapp)})' (from 'CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(lastapp)' (from 'CDR(lastapp)})' len 12) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(lastapp) result is 'Dial' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(Dial) result is '"Dial"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastdata)})' (from 'CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(lastdata)' (from 'CDR(lastdata)})' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(lastdata) result is 'SIP/6002,15' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6002,15) result is '"SIP/6002,15"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(start)})' (from 'CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 24) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)})' len 10) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(start) result is '2014-03-21 17:25:58' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:25:58) result is '"2014-03-21 17:25:58"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(answer)})' (from 'CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 25) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)})' len 11) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(answer) result is '2014-03-21 17:25:58' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:25:58) result is '"2014-03-21 17:25:58"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(end)})' (from 'CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)})' len 8) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(end) result is '2014-03-21 17:26:07' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:26:07) result is '"2014-03-21 17:26:07"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(duration,f)})' (from 'CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(duration,f)' (from 'CDR(duration,f)})' len 15) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(duration,f) result is '0.008000' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.008000) result is '"0.008000"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(billsec,f)})' (from 'CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(billsec,f)' (from 'CDR(billsec,f)})' len 14) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(billsec,f) result is '0.008000' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.008000) result is '"0.008000"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(disposition)})' (from 'CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)})' len 16) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(disposition) result is 'ANSWERED' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(ANSWERED) result is '"ANSWERED"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(amaflags)})' (from 'CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(amaflags)' (from 'CDR(amaflags)})' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(amaflags) result is 'DOCUMENTATION' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(DOCUMENTATION) result is '"DOCUMENTATION"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(accountcode)})' (from 'CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(accountcode)' (from 'CDR(accountcode)})' len 16) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(accountcode) result is '' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(uniqueid)})' (from 'CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)})' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(uniqueid) result is '1395440743.8' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE(1395440743.8) result is '"1395440743.8"' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(userfield)})' (from 'CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(userfield)' (from 'CDR(userfield)})' len 14) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(userfield) result is '' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(sequence)' (from 'CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 13) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(sequence) result is '11' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield1)})' (from 'CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield1)' (from 'CDR(acustomfield1)})' len 18) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(acustomfield1) result is '' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield2)})' (from 'CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield2)' (from 'CDR(acustomfield2)})' len 18) [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CDR(acustomfield2) result is '' [Mar 21 17:26:07] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:26:07] DEBUG[10242] cdr.c: CDR for SIP/6003-00000007 is dialed and has no Party B; discarding [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK35830e09 [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as1e36869b [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=E3364A2C-61B601E7 [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 4 [ 13]: CSeq: 106 BYE [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Mar 21 17:26:07] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: = Looking for Call ID: 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 (Checking To) --From tag as1e36869b --To-tag E3364A2C-61B601E7 [Mar 21 17:26:07] DEBUG[10274][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #125 [Mar 21 17:26:07] DEBUG[10274][C-00000003] chan_sip.c: Stopping retransmission on '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' of Request 106: Match Found [Mar 21 17:26:07] DEBUG[10274] chan_sip.c: Destroying SIP dialog 47fba21e68dded54733885b613fc5e49@10.24.18.124:5060 [Mar 21 17:26:07] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '47fba21e68dded54733885b613fc5e49@10.24.18.124:5060' Method: INVITE [Mar 21 17:26:07] DEBUG[10274] rtp_engine.c: Destroyed RTP instance '0x7f5dd400a7a8' [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] DEBUG[10274] chan_sip.c: Header 0 [ 33]: REGISTER sip:10.24.18.124 SIP/2.0 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 1 [ 72]: Via: SIP/2.0/UDP 10.24.18.166:5060;branch=z9hG4bK76673914174292040;rport [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 2 [ 54]: From: 6004 ;tag=3167523177 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 3 [ 37]: To: 6004 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 4 [ 50]: Call-ID: 21608363712124-1345167932089@10.24.18.166 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 7 REGISTER [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 8 [ 11]: Expires: 60 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 9 [ 15]: Supported: path [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 10 [ 26]: User-Agent: Voip Phone 1.0 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: --- (12 headers 0 lines) --- [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: = Looking for Call ID: 21608363712124-1345167932089@10.24.18.166 (Checking From) --From tag 3167523177 --To-tag [Mar 21 17:26:20] DEBUG[10274] acl.c: For destination '10.24.18.166', our source address is '10.24.18.124'. [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:26:20] DEBUG[10274] netsock2.c: Splitting '10.24.18.166:5060' into... [Mar 21 17:26:20] DEBUG[10274] netsock2.c: ...host '10.24.18.166' and port '5060'. [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.166:5060 (no NAT) [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for 21608363712124-1345167932089@10.24.18.166 - REGISTER (No RTP) [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Initializing initreq for method REGISTER - callid 21608363712124-1345167932089@10.24.18.166 [Mar 21 17:26:20] DEBUG[10274] netsock2.c: Splitting '10.24.18.166:5060' into... [Mar 21 17:26:20] DEBUG[10274] netsock2.c: ...host '10.24.18.166' and port '5060'. [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.166:5060 (no NAT) [Mar 21 17:26:20] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:26:20] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.166:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 33]: REGISTER sip:10.24.18.124 SIP/2.0 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 10.24.18.166:5060;branch=z9hG4bK176871677511483793;rport [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 2 [ 54]: From: 6004 ;tag=3167523177 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 3 [ 37]: To: 6004 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 4 [ 50]: Call-ID: 21608363712124-1345167932089@10.24.18.166 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 8 REGISTER [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 7 [157]: Authorization: Digest username="6004", realm="asterisk", nonce="444e890e", uri="sip:10.24.18.124", response="8916eb4c088b2e8df0e28c81657d440d", algorithm=MD5 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 9 [ 11]: Expires: 60 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 10 [ 15]: Supported: path [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 11 [ 26]: User-Agent: Voip Phone 1.0 [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: --- (13 headers 0 lines) --- [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: = Looking for Call ID: 21608363712124-1345167932089@10.24.18.166 (Checking From) --From tag 3167523177 --To-tag [Mar 21 17:26:20] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:26:20] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:26:20] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:26:20] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Initializing initreq for method REGISTER - callid 21608363712124-1345167932089@10.24.18.166 [Mar 21 17:26:20] DEBUG[10274] netsock2.c: Splitting '10.24.18.166:5060' into... [Mar 21 17:26:20] DEBUG[10274] netsock2.c: ...host '10.24.18.166' and port '5060'. [Mar 21 17:26:20] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.166:5060 (no NAT) [Mar 21 17:26:20] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:26:20] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: Store REGISTER's Contact header for call routing. [Mar 21 17:26:20] DEBUG[10274] netsock2.c: Splitting '10.24.18.166:5060' into... [Mar 21 17:26:20] DEBUG[10274] netsock2.c: ...host '10.24.18.166' and port '5060'. [Mar 21 17:26:20] DEBUG[10274] chan_sip.c: build_path: do not use Path headers [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.166:5060 [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] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6004 [Mar 21 17:26:20] DEBUG[10240] chan_sip.c: Checking device state for peer 6004 [Mar 21 17:26:20] DEBUG[10240] devicestate.c: Changing state for SIP/6004 - state 1 (Not in use) [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: REGISTER sip:10.24.18.124:5060 SIP/2.0 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjwkdzAsZj7EnG8DXiEaJT-1TrPTXVU3Qi [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=7AQZq4QPsiEBNYkZv6mZ8eR..PSXYKmZ [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 4 [ 38]: To: "RustyONE" [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 6 [ 20]: CSeq: 35644 REGISTER [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 7 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 8 [ 50]: Contact: "RustyONE" [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 9 [ 12]: Expires: 300 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 10 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: --- (12 headers 0 lines) --- [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: = Looking for Call ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV (Checking From) --From tag 7AQZq4QPsiEBNYkZv6mZ8eR..PSXYKmZ --To-tag [Mar 21 17:26:23] DEBUG[10274] acl.c: For destination '10.24.18.16', our source address is '10.24.18.124'. [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for m6qybWzN.pQsFuItAEreYfCYc7g34OiV - REGISTER (No RTP) [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Initializing initreq for method REGISTER - callid m6qybWzN.pQsFuItAEreYfCYc7g34OiV [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.16:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: REGISTER sip:10.24.18.124:5060 SIP/2.0 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjrBZR4phaqmKI1ZzATLfe6N4V3-hzviii [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=7AQZq4QPsiEBNYkZv6mZ8eR..PSXYKmZ [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 4 [ 38]: To: "RustyONE" [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 6 [ 20]: CSeq: 35645 REGISTER [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 7 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 8 [ 50]: Contact: "RustyONE" [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 9 [ 12]: Expires: 300 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 10 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 11 [162]: Authorization: Digest username="6001", realm="asterisk", nonce="659a9f74", uri="sip:10.24.18.124:5060", response="702a63e8cba541d825290c5fb233f6ae", algorithm=MD5 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: --- (13 headers 0 lines) --- [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: = Looking for Call ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV (Checking From) --From tag 7AQZq4QPsiEBNYkZv6mZ8eR..PSXYKmZ --To-tag [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Initializing initreq for method REGISTER - callid m6qybWzN.pQsFuItAEreYfCYc7g34OiV [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Store REGISTER's Contact header for call routing. [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: build_path: do not use Path headers [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.16:5060 [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] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6001 [Mar 21 17:26:23] DEBUG[10240] chan_sip.c: Checking device state for peer 6001 [Mar 21 17:26:23] DEBUG[10240] devicestate.c: Changing state for SIP/6001 - state 1 (Not in use) [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] DEBUG[10274] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjhCgIKsh0gD7VgD888leVafuC3k5Vb8Q. [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=yeuK.RmIBLmDywGlApTuWD1lqecjeUJt [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 4 [ 38]: To: "RustyONE" [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 5 [ 50]: Contact: "RustyONE" [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 7 [ 20]: CSeq: 5665 SUBSCRIBE [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 8 [ 22]: Event: message-summary [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 9 [ 13]: Expires: 3600 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 10 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 11 [ 42]: Accept: application/simple-message-summary [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 12 [ 46]: Allow-Events: presence, message-summary, refer [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 13 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: --- (15 headers 0 lines) --- [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: = Looking for Call ID: hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 (Checking From) --From tag yeuK.RmIBLmDywGlApTuWD1lqecjeUJt --To-tag [Mar 21 17:26:23] DEBUG[10274] acl.c: For destination '10.24.18.16', our source address is '10.24.18.124'. [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 - SUBSCRIBE (No RTP) [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: build_route: Contact hop: "RustyONE" [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: list_route: route/path hop: [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.16:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjvWDQ4j20ZP.DyjFl8iiZu3irVS8078Zh [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=yeuK.RmIBLmDywGlApTuWD1lqecjeUJt [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 4 [ 38]: To: "RustyONE" [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 5 [ 50]: Contact: "RustyONE" [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 7 [ 20]: CSeq: 5666 SUBSCRIBE [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 8 [ 22]: Event: message-summary [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 9 [ 13]: Expires: 3600 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 10 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 11 [ 42]: Accept: application/simple-message-summary [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 12 [ 46]: Allow-Events: presence, message-summary, refer [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 13 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 14 [167]: Authorization: Digest username="6001", realm="asterisk", nonce="557ca6ae", uri="sip:6001@10.24.18.124:5060", response="9abd4ad6331288d6d8410a3a2eea7cd9", algorithm=MD5 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: --- (16 headers 0 lines) --- [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: = Looking for Call ID: hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 (Checking From) --From tag yeuK.RmIBLmDywGlApTuWD1lqecjeUJt --To-tag [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Got a new subscription hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 (possibly with auth) or retransmission [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: build_route: Retaining previous route: [Mar 21 17:26:23] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:26:23] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:26:23] NOTICE[10274] chan_sip.c: Received SIP subscribe for peer without mailbox: 6001 [Mar 21 17:26:23] DEBUG[10274] chan_sip.c: Destroying SIP dialog hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3 [Mar 21 17:26:23] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog 'hrLPxszRFjC7l.cpWGQOdzu4gJKN36P3' Method: SUBSCRIBE