[Dec 9 19:33:10] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 6a08235c523884745119dab14bbc3f35@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:33:10] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:33:10] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:33:10] DEBUG[85170] chan_sip.c: SIP call-id changed from '6a08235c523884745119dab14bbc3f35@207.91.156.217:5060' to '4ef221e80a51e28c258549e80707b066@207.91.156.217:5060' [Dec 9 19:33:10] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 4ef221e80a51e28c258549e80707b066@207.91.156.217:5060 [Dec 9 19:33:10] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:33:10] DEBUG[85170] chan_sip.c: = Looking for Call ID: 4ef221e80a51e28c258549e80707b066@207.91.156.217:5060 (Checking To) --From tag as5185b2c3 --To-tag 150794584 [Dec 9 19:33:10] DEBUG[85170] chan_sip.c: Stopping retransmission on '4ef221e80a51e28c258549e80707b066@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:33:10] DEBUG[85170] chan_sip.c: Destroying SIP dialog 4ef221e80a51e28c258549e80707b066@207.91.156.217:5060 [Dec 9 19:33:11] VERBOSE[85143] asterisk.c: Remote UNIX connection [Dec 9 19:33:11] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 2dd9ba5018a5ead132ea66660cf0197b@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:33:18] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: SIP call-id changed from '2dd9ba5018a5ead132ea66660cf0197b@207.91.156.217:5060' to '676069b414726034105d139304947944@207.91.156.217:5060' [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 676069b414726034105d139304947944@207.91.156.217:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: = Looking for Call ID: 676069b414726034105d139304947944@207.91.156.217:5060 (Checking To) --From tag as1d6d15eb --To-tag 1980999769 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Stopping retransmission on '676069b414726034105d139304947944@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Destroying SIP dialog 676069b414726034105d139304947944@207.91.156.217:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: = Looking for Call ID: 1255025376-5060-14@BJC.BGI.B.G (Checking From) --From tag 725432686 --To-tag [Dec 9 19:33:18] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:33:18] DEBUG[85170] netsock2.c: Splitting '192.168.1.6:5060' into... [Dec 9 19:33:18] DEBUG[85170] netsock2.c: ...host '192.168.1.6' and port '5060'. [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: NAT detected for 192.168.1.6 / 72.190.32.250 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 1255025376-5060-14@BJC.BGI.B.G - REGISTER (No RTP) [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Dec 9 19:33:18] DEBUG[85170] netsock2.c: Splitting '192.168.1.6:5060' into... [Dec 9 19:33:18] DEBUG[85170] netsock2.c: ...host '192.168.1.6' and port '5060'. [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: NAT detected for 192.168.1.6 / 72.190.32.250 [Dec 9 19:33:18] DEBUG[85170] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:33:18] DEBUG[85170] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:33:18] DEBUG[94779] manager.c: Examining AMI event: Event: ChallengeSent Privilege: security,all EventTV: 2016-12-09T19:33:18.800+0000 Severity: Informational Service: SIP EventVersion: 1 AccountID: 100 SessionID: 0x7f4e80023870 LocalAddress: IPV4/UDP/207.91.156.217/5060 RemoteAddress: IPV4/UDP/72.190.32.250/5060 Challenge: 0edcf0c9 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: = Looking for Call ID: 1255025376-5060-14@BJC.BGI.B.G (Checking From) --From tag 725432686 --To-tag [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Dec 9 19:33:18] DEBUG[85170] netsock2.c: Splitting '192.168.1.6:5060' into... [Dec 9 19:33:18] DEBUG[85170] netsock2.c: ...host '192.168.1.6' and port '5060'. [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: NAT detected for 192.168.1.6 / 72.190.32.250 [Dec 9 19:33:18] DEBUG[85170] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:33:18] DEBUG[85170] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: build_path: do not use Path headers [Dec 9 19:33:18] DEBUG[85138] threadpool.c: Increasing threadpool stasis-core's size by 1 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 7b412ea8576f1b5b430a6f2974f1f128@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:33:18] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: SIP call-id changed from '7b412ea8576f1b5b430a6f2974f1f128@207.91.156.217:5060' to '16c019ed3820b7a92e3fa8a3048ebe0b@207.91.156.217:5060' [Dec 9 19:33:18] DEBUG[94779] manager.c: Examining AMI event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/100 PeerStatus: Registered Address: 72.190.32.250:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 16c019ed3820b7a92e3fa8a3048ebe0b@207.91.156.217:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 4e0ee3e2758651511232519b74fbbc4e@207.91.156.217:5060 - NOTIFY (No RTP) [Dec 9 19:33:18] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: SIP call-id changed from '4e0ee3e2758651511232519b74fbbc4e@207.91.156.217:5060' to '2132fa2e44efed61677aa2730020d25a@207.91.156.217:5060' [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Initializing initreq for method NOTIFY - callid 2132fa2e44efed61677aa2730020d25a@207.91.156.217:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:33:18] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Dec 9 19:33:18] DEBUG[85149] chan_sip.c: Checking device state for peer 100 [Dec 9 19:33:18] DEBUG[85149] devicestate.c: Changing state for SIP/100 - state 1 (Not in use) [Dec 9 19:33:18] DEBUG[94779] manager.c: Examining AMI event: Event: SuccessfulAuth Privilege: security,all EventTV: 2016-12-09T19:33:18.819+0000 Severity: Informational Service: SIP EventVersion: 1 AccountID: 100 SessionID: 0x7f4e80023870 LocalAddress: IPV4/UDP/207.91.156.217/5060 RemoteAddress: IPV4/UDP/72.190.32.250/5060 UsingPassword: 1 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: = Looking for Call ID: 16c019ed3820b7a92e3fa8a3048ebe0b@207.91.156.217:5060 (Checking To) --From tag as7baf716f --To-tag 787075347 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Stopping retransmission on '16c019ed3820b7a92e3fa8a3048ebe0b@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Destroying SIP dialog 16c019ed3820b7a92e3fa8a3048ebe0b@207.91.156.217:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: = Looking for Call ID: 2132fa2e44efed61677aa2730020d25a@207.91.156.217:5060 (Checking To) --From tag as41657bd9 --To-tag 994059841 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Stopping retransmission on '2132fa2e44efed61677aa2730020d25a@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Got 200 accepted on NOTIFY 2132fa2e44efed61677aa2730020d25a@207.91.156.217:5060 [Dec 9 19:33:18] DEBUG[85170] chan_sip.c: Destroying SIP dialog 2132fa2e44efed61677aa2730020d25a@207.91.156.217:5060 [Dec 9 19:33:24] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:33:25] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 479545951d447f822b67164c59988649@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:33:25] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:33:25] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:33:25] DEBUG[85170] chan_sip.c: SIP call-id changed from '479545951d447f822b67164c59988649@207.91.156.217:5060' to '051c012e35659e06222e29d9164d6a00@207.91.156.217:5060' [Dec 9 19:33:25] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 051c012e35659e06222e29d9164d6a00@207.91.156.217:5060 [Dec 9 19:33:25] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:33:25] DEBUG[85170] chan_sip.c: = Looking for Call ID: 051c012e35659e06222e29d9164d6a00@207.91.156.217:5060 (Checking To) --From tag as2c9eaadf --To-tag 683122882 [Dec 9 19:33:25] DEBUG[85170] chan_sip.c: Stopping retransmission on '051c012e35659e06222e29d9164d6a00@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:33:25] DEBUG[85170] chan_sip.c: Destroying SIP dialog 051c012e35659e06222e29d9164d6a00@207.91.156.217:5060 [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 44197dbb18afe4753f5632f65fb82606@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:33:33] DEBUG[85170] acl.c: For destination '216.115.69.144', our source address is '207.91.156.217'. [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: SIP call-id changed from '44197dbb18afe4753f5632f65fb82606@207.91.156.217:5060' to '718c5a9c651616b21e9844d84996ae38@207.91.156.217:5060' [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 718c5a9c651616b21e9844d84996ae38@207.91.156.217:5060 [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: = Looking for Call ID: 718c5a9c651616b21e9844d84996ae38@207.91.156.217:5060 (Checking To) --From tag as2a3b2b29 --To-tag 20e1698a3241cbcc6677a39fcb65a0aa.4a50 [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: Stopping retransmission on '718c5a9c651616b21e9844d84996ae38@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: Destroying SIP dialog 718c5a9c651616b21e9844d84996ae38@207.91.156.217:5060 [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 24f4908a099b3705573ea0cc500eec7b@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:33:33] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: SIP call-id changed from '24f4908a099b3705573ea0cc500eec7b@207.91.156.217:5060' to '4d1334690e4ddd4a21e6006a0b939c12@207.91.156.217:5060' [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 4d1334690e4ddd4a21e6006a0b939c12@207.91.156.217:5060 [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: = Looking for Call ID: 4d1334690e4ddd4a21e6006a0b939c12@207.91.156.217:5060 (Checking To) --From tag as1da5563d --To-tag 1731851618 [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: Stopping retransmission on '4d1334690e4ddd4a21e6006a0b939c12@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:33:33] DEBUG[85170] chan_sip.c: Destroying SIP dialog 4d1334690e4ddd4a21e6006a0b939c12@207.91.156.217:5060 [Dec 9 19:33:37] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:33:38] DEBUG[113609] threadpool.c: Worker thread idle timeout reached. Dying. [Dec 9 19:33:38] DEBUG[85138] threadpool.c: Destroying worker thread 586 [Dec 9 19:33:40] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 6afe2eee22e3377568fbe8d344d71d5d@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:33:40] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:33:40] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:33:40] DEBUG[85170] chan_sip.c: SIP call-id changed from '6afe2eee22e3377568fbe8d344d71d5d@207.91.156.217:5060' to '3692e4764a4387b07bd999a807a7ba26@207.91.156.217:5060' [Dec 9 19:33:40] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 3692e4764a4387b07bd999a807a7ba26@207.91.156.217:5060 [Dec 9 19:33:40] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:33:40] DEBUG[85170] chan_sip.c: = Looking for Call ID: 3692e4764a4387b07bd999a807a7ba26@207.91.156.217:5060 (Checking To) --From tag as1bbe6290 --To-tag 2016120507 [Dec 9 19:33:40] DEBUG[85170] chan_sip.c: Stopping retransmission on '3692e4764a4387b07bd999a807a7ba26@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:33:40] DEBUG[85170] chan_sip.c: Destroying SIP dialog 3692e4764a4387b07bd999a807a7ba26@207.91.156.217:5060 [Dec 9 19:33:48] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 154c908b61fdd69270cec6c165625911@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:33:48] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:33:48] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:33:48] DEBUG[85170] chan_sip.c: SIP call-id changed from '154c908b61fdd69270cec6c165625911@207.91.156.217:5060' to '2febfeb75b722daa04403c8f3982973d@207.91.156.217:5060' [Dec 9 19:33:48] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 2febfeb75b722daa04403c8f3982973d@207.91.156.217:5060 [Dec 9 19:33:48] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:33:48] DEBUG[85170] chan_sip.c: = Looking for Call ID: 2febfeb75b722daa04403c8f3982973d@207.91.156.217:5060 (Checking To) --From tag as7a66d0fc --To-tag 160173959 [Dec 9 19:33:48] DEBUG[85170] chan_sip.c: Stopping retransmission on '2febfeb75b722daa04403c8f3982973d@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:33:48] DEBUG[85170] chan_sip.c: Destroying SIP dialog 2febfeb75b722daa04403c8f3982973d@207.91.156.217:5060 [Dec 9 19:33:50] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:33:50] DEBUG[85170] chan_sip.c: Auto destroying SIP dialog '1255025376-5060-14@BJC.BGI.B.G' [Dec 9 19:33:50] DEBUG[85170] chan_sip.c: Destroying SIP dialog 1255025376-5060-14@BJC.BGI.B.G [Dec 9 19:33:55] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 05e05660015a6da021d2065a0974be97@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:33:55] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:33:55] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:33:55] DEBUG[85170] chan_sip.c: SIP call-id changed from '05e05660015a6da021d2065a0974be97@207.91.156.217:5060' to '125a0f7101dee171078ada9176b25694@207.91.156.217:5060' [Dec 9 19:33:55] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 125a0f7101dee171078ada9176b25694@207.91.156.217:5060 [Dec 9 19:33:55] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:33:55] DEBUG[85170] chan_sip.c: = Looking for Call ID: 125a0f7101dee171078ada9176b25694@207.91.156.217:5060 (Checking To) --From tag as1057d638 --To-tag 933649874 [Dec 9 19:33:55] DEBUG[85170] chan_sip.c: Stopping retransmission on '125a0f7101dee171078ada9176b25694@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:33:55] DEBUG[85170] chan_sip.c: Destroying SIP dialog 125a0f7101dee171078ada9176b25694@207.91.156.217:5060 [Dec 9 19:34:02] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:34:03] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 5c7dfea0022e062f20429510745e0457@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:34:03] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:34:03] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:34:03] DEBUG[85170] chan_sip.c: SIP call-id changed from '5c7dfea0022e062f20429510745e0457@207.91.156.217:5060' to '6fe0d9f73e61b9bd4d40cea54d9eda59@207.91.156.217:5060' [Dec 9 19:34:03] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 6fe0d9f73e61b9bd4d40cea54d9eda59@207.91.156.217:5060 [Dec 9 19:34:03] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:34:03] DEBUG[85170] chan_sip.c: = Looking for Call ID: 6fe0d9f73e61b9bd4d40cea54d9eda59@207.91.156.217:5060 (Checking To) --From tag as014fef9d --To-tag 1114225593 [Dec 9 19:34:03] DEBUG[85170] chan_sip.c: Stopping retransmission on '6fe0d9f73e61b9bd4d40cea54d9eda59@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:34:03] DEBUG[85170] chan_sip.c: Destroying SIP dialog 6fe0d9f73e61b9bd4d40cea54d9eda59@207.91.156.217:5060 [Dec 9 19:34:10] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 2c6d6aef184785430373e2783a92ccfb@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:34:10] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:34:10] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:34:10] DEBUG[85170] chan_sip.c: SIP call-id changed from '2c6d6aef184785430373e2783a92ccfb@207.91.156.217:5060' to '33e6046041eeaee23d3d0ea278b4ba9a@207.91.156.217:5060' [Dec 9 19:34:10] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 33e6046041eeaee23d3d0ea278b4ba9a@207.91.156.217:5060 [Dec 9 19:34:10] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:34:10] DEBUG[85170] chan_sip.c: = Looking for Call ID: 33e6046041eeaee23d3d0ea278b4ba9a@207.91.156.217:5060 (Checking To) --From tag as3e1e83a5 --To-tag 534058499 [Dec 9 19:34:10] DEBUG[85170] chan_sip.c: Stopping retransmission on '33e6046041eeaee23d3d0ea278b4ba9a@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:34:10] DEBUG[85170] chan_sip.c: Destroying SIP dialog 33e6046041eeaee23d3d0ea278b4ba9a@207.91.156.217:5060 [Dec 9 19:34:15] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:34:18] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 059038b778822d013546a37c33c6ad92@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:34:18] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:34:18] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:34:18] DEBUG[85170] chan_sip.c: SIP call-id changed from '059038b778822d013546a37c33c6ad92@207.91.156.217:5060' to '32e4c6657161824371b16f4b4caf57fc@207.91.156.217:5060' [Dec 9 19:34:18] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 32e4c6657161824371b16f4b4caf57fc@207.91.156.217:5060 [Dec 9 19:34:18] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:34:18] DEBUG[85170] chan_sip.c: = Looking for Call ID: 32e4c6657161824371b16f4b4caf57fc@207.91.156.217:5060 (Checking To) --From tag as00cf4eaf --To-tag 433152778 [Dec 9 19:34:18] DEBUG[85170] chan_sip.c: Stopping retransmission on '32e4c6657161824371b16f4b4caf57fc@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:34:18] DEBUG[85170] chan_sip.c: Destroying SIP dialog 32e4c6657161824371b16f4b4caf57fc@207.91.156.217:5060 [Dec 9 19:34:25] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 746a2a17524401ca0e5124df06635603@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:34:25] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:34:25] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:34:25] DEBUG[85170] chan_sip.c: SIP call-id changed from '746a2a17524401ca0e5124df06635603@207.91.156.217:5060' to '655f94df08f2ddad7a37b7af4ba1323f@207.91.156.217:5060' [Dec 9 19:34:25] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 655f94df08f2ddad7a37b7af4ba1323f@207.91.156.217:5060 [Dec 9 19:34:25] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:34:25] DEBUG[85170] chan_sip.c: = Looking for Call ID: 655f94df08f2ddad7a37b7af4ba1323f@207.91.156.217:5060 (Checking To) --From tag as1262a737 --To-tag 1172241358 [Dec 9 19:34:25] DEBUG[85170] chan_sip.c: Stopping retransmission on '655f94df08f2ddad7a37b7af4ba1323f@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:34:25] DEBUG[85170] chan_sip.c: Destroying SIP dialog 655f94df08f2ddad7a37b7af4ba1323f@207.91.156.217:5060 [Dec 9 19:34:28] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 5dee585a53c76a1e79474b7308b89a4e@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:34:33] DEBUG[85170] acl.c: For destination '216.115.69.144', our source address is '207.91.156.217'. [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: SIP call-id changed from '5dee585a53c76a1e79474b7308b89a4e@207.91.156.217:5060' to '4fbda8526516b3aa1a65ca9c06b7374d@207.91.156.217:5060' [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 4fbda8526516b3aa1a65ca9c06b7374d@207.91.156.217:5060 [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: = Looking for Call ID: 4fbda8526516b3aa1a65ca9c06b7374d@207.91.156.217:5060 (Checking To) --From tag as1aaa27ce --To-tag 20e1698a3241cbcc6677a39fcb65a0aa.395a [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: Stopping retransmission on '4fbda8526516b3aa1a65ca9c06b7374d@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: Destroying SIP dialog 4fbda8526516b3aa1a65ca9c06b7374d@207.91.156.217:5060 [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 02a6897209b0544b0360211d08f40e41@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:34:33] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: SIP call-id changed from '02a6897209b0544b0360211d08f40e41@207.91.156.217:5060' to '74caa9953a2626c74e6dd79574f2a4ef@207.91.156.217:5060' [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 74caa9953a2626c74e6dd79574f2a4ef@207.91.156.217:5060 [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: = Looking for Call ID: 74caa9953a2626c74e6dd79574f2a4ef@207.91.156.217:5060 (Checking To) --From tag as3d2e6d96 --To-tag 1038752328 [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: Stopping retransmission on '74caa9953a2626c74e6dd79574f2a4ef@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:34:33] DEBUG[85170] chan_sip.c: Destroying SIP dialog 74caa9953a2626c74e6dd79574f2a4ef@207.91.156.217:5060 [Dec 9 19:34:40] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 2a8b8c6b06b2643c0495f40649d93489@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:34:40] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:34:40] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:34:40] DEBUG[85170] chan_sip.c: SIP call-id changed from '2a8b8c6b06b2643c0495f40649d93489@207.91.156.217:5060' to '19163b6301c1979d3d0f079a3d9db53a@207.91.156.217:5060' [Dec 9 19:34:40] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 19163b6301c1979d3d0f079a3d9db53a@207.91.156.217:5060 [Dec 9 19:34:40] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:34:40] DEBUG[85170] chan_sip.c: = Looking for Call ID: 19163b6301c1979d3d0f079a3d9db53a@207.91.156.217:5060 (Checking To) --From tag as4b8f3aa7 --To-tag 198256214 [Dec 9 19:34:40] DEBUG[85170] chan_sip.c: Stopping retransmission on '19163b6301c1979d3d0f079a3d9db53a@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:34:40] DEBUG[85170] chan_sip.c: Destroying SIP dialog 19163b6301c1979d3d0f079a3d9db53a@207.91.156.217:5060 [Dec 9 19:34:40] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:34:48] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 4816026f7726cf3f22f2608020681272@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:34:48] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:34:48] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:34:48] DEBUG[85170] chan_sip.c: SIP call-id changed from '4816026f7726cf3f22f2608020681272@207.91.156.217:5060' to '5f1f574135813ec61e2ac1911f97c6fb@207.91.156.217:5060' [Dec 9 19:34:48] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 5f1f574135813ec61e2ac1911f97c6fb@207.91.156.217:5060 [Dec 9 19:34:48] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:34:48] DEBUG[85170] chan_sip.c: = Looking for Call ID: 5f1f574135813ec61e2ac1911f97c6fb@207.91.156.217:5060 (Checking To) --From tag as2a86a34f --To-tag 714063858 [Dec 9 19:34:48] DEBUG[85170] chan_sip.c: Stopping retransmission on '5f1f574135813ec61e2ac1911f97c6fb@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:34:48] DEBUG[85170] chan_sip.c: Destroying SIP dialog 5f1f574135813ec61e2ac1911f97c6fb@207.91.156.217:5060 [Dec 9 19:34:53] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:34:55] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 28532d40145854223ad227f64a4bd0d2@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:34:55] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:34:55] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:34:55] DEBUG[85170] chan_sip.c: SIP call-id changed from '28532d40145854223ad227f64a4bd0d2@207.91.156.217:5060' to '5d400f9736fabb582854088309eec0f3@207.91.156.217:5060' [Dec 9 19:34:55] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 5d400f9736fabb582854088309eec0f3@207.91.156.217:5060 [Dec 9 19:34:55] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:34:55] DEBUG[85170] chan_sip.c: = Looking for Call ID: 5d400f9736fabb582854088309eec0f3@207.91.156.217:5060 (Checking To) --From tag as76c389e6 --To-tag 1145230203 [Dec 9 19:34:55] DEBUG[85170] chan_sip.c: Stopping retransmission on '5d400f9736fabb582854088309eec0f3@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:34:55] DEBUG[85170] chan_sip.c: Destroying SIP dialog 5d400f9736fabb582854088309eec0f3@207.91.156.217:5060 [Dec 9 19:35:00] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> <-------------> [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 27947c6a3889100314577c3a25870529@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:03] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: SIP call-id changed from '27947c6a3889100314577c3a25870529@207.91.156.217:5060' to '3ae5ea0146f6ec691a06d06641a2ff06@207.91.156.217:5060' [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 3ae5ea0146f6ec691a06d06641a2ff06@207.91.156.217:5060 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 0 [ 40]: OPTIONS sip:100@192.168.1.6:5060 SIP/2.0 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3303e381;rport [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as74d99003 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 4 [ 30]: To: [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 3ae5ea0146f6ec691a06d06641a2ff06@207.91.156.217:5060 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:35:03 GMT [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:35:03] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 72.190.32.250:5060: OPTIONS sip:100@192.168.1.6:5060 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3303e381;rport Max-Forwards: 70 From: "asterisk" ;tag=as74d99003 To: Contact: Call-ID: 3ae5ea0146f6ec691a06d06641a2ff06@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:35:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:03] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3303e381;rport=5060 From: "asterisk" ;tag=as74d99003 To: ;tag=632864728 Call-ID: 3ae5ea0146f6ec691a06d06641a2ff06@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3303e381;rport=5060 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as74d99003 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 3 [ 44]: To: ;tag=632864728 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 3ae5ea0146f6ec691a06d06641a2ff06@207.91.156.217:5060 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:35:03] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: = Looking for Call ID: 3ae5ea0146f6ec691a06d06641a2ff06@207.91.156.217:5060 (Checking To) --From tag as74d99003 --To-tag 632864728 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8 [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Stopping retransmission on '3ae5ea0146f6ec691a06d06641a2ff06@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:03] DEBUG[85170] chan_sip.c: Destroying SIP dialog 3ae5ea0146f6ec691a06d06641a2ff06@207.91.156.217:5060 [Dec 9 19:35:03] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '3ae5ea0146f6ec691a06d06641a2ff06@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:35:05] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> <-------------> [Dec 9 19:35:06] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 44c4705039a15e285a5ec74818f8998f@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:10] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: SIP call-id changed from '44c4705039a15e285a5ec74818f8998f@207.91.156.217:5060' to '32e848e76ce7b11c392d28cc3bbfd998@207.91.156.217:5060' [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 32e848e76ce7b11c392d28cc3bbfd998@207.91.156.217:5060 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 0 [ 44]: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK72329b79;rport [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as2305471a [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 4 [ 34]: To: [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 32e848e76ce7b11c392d28cc3bbfd998@207.91.156.217:5060 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:35:10 GMT [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 97.77.119.234:47976: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK72329b79;rport Max-Forwards: 70 From: "asterisk" ;tag=as2305471a To: Contact: Call-ID: 32e848e76ce7b11c392d28cc3bbfd998@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:35:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #45 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> INVITE sip:8018309060@testbed2.primevox.net SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK1641505246;rport From: "Local1" ;tag=2051421774 To: Call-ID: 1453900646-5060-25@BJC.BGI.B.G CSeq: 220 INVITE Contact: "Local1" Max-Forwards: 70 User-Agent: Grandstream GXP2140 1.0.7.97 Privacy: none P-Preferred-Identity: "Local1" Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 355 v=0 o=100 8001 8000 IN IP4 192.168.1.6 s=SIP Call c=IN IP4 192.168.1.6 t=0 0 m=audio 5006 RTP/AVP 125 0 97 18 9 101 a=sendrecv a=rtpmap:125 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 0 [ 51]: INVITE sip:8018309060@testbed2.primevox.net SIP/2.0 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK1641505246;rport [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "Local1" ;tag=2051421774 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 3 [ 42]: To: [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 4 [ 39]: Call-ID: 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 220 INVITE [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 6 [ 44]: Contact: "Local1" [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 8 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 9 [ 13]: Privacy: none [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 10 [ 62]: P-Preferred-Identity: "Local1" [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 11 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 12 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 14 [ 47]: Accept: application/sdp, application/dtmf-relay [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 15 [ 19]: Content-Length: 355 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 16 [ 0]: [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 0 [ 3]: v=0 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 1 [ 34]: o=100 8001 8000 IN IP4 192.168.1.6 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 2 [ 10]: s=SIP Call [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 3 [ 20]: c=IN IP4 192.168.1.6 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 5 [ 38]: m=audio 5006 RTP/AVP 125 0 97 18 9 101 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 6 [ 10]: a=sendrecv [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 7 [ 25]: a=rtpmap:125 opus/48000/2 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 8 [ 10]: a=ptime:20 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 10 [ 21]: a=rtpmap:97 iLBC/8000 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 11 [ 17]: a=fmtp:97 mode=30 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 12 [ 21]: a=rtpmap:18 G729/8000 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 13 [ 19]: a=fmtp:18 annexb=no [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 14 [ 20]: a=rtpmap:9 G722/8000 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 15 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Body 16 [ 15]: a=fmtp:101 0-15 [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: --- (16 headers 17 lines) --- [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: = Looking for Call ID: 1453900646-5060-25@BJC.BGI.B.G (Checking From) --From tag 2051421774 --To-tag [Dec 9 19:35:10] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:10] DEBUG[85170] netsock2.c: Splitting '192.168.1.6:5060' into... [Dec 9 19:35:10] DEBUG[85170] netsock2.c: ...host '192.168.1.6' and port '5060'. [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: NAT detected for 192.168.1.6 / 72.190.32.250 [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: Sending to 72.190.32.250:5060 (NAT) [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 1453900646-5060-25@BJC.BGI.B.G - INVITE (No RTP) [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Dec 9 19:35:10] DEBUG[85170][C-000006f8] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, path, timer" [Dec 9 19:35:10] DEBUG[85170][C-000006f8] sip/reqresp_parser.c: Found SIP option: -replaces- [Dec 9 19:35:10] DEBUG[85170][C-000006f8] sip/reqresp_parser.c: Matched SIP option: replaces [Dec 9 19:35:10] DEBUG[85170][C-000006f8] sip/reqresp_parser.c: Found SIP option: -path- [Dec 9 19:35:10] DEBUG[85170][C-000006f8] sip/reqresp_parser.c: Matched SIP option: path [Dec 9 19:35:10] DEBUG[85170][C-000006f8] sip/reqresp_parser.c: Found SIP option: -timer- [Dec 9 19:35:10] DEBUG[85170][C-000006f8] sip/reqresp_parser.c: Matched SIP option: timer [Dec 9 19:35:10] DEBUG[85170][C-000006f8] netsock2.c: Splitting '192.168.1.6:5060' into... [Dec 9 19:35:10] DEBUG[85170][C-000006f8] netsock2.c: ...host '192.168.1.6' and port '5060'. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: NAT detected for 192.168.1.6 / 72.190.32.250 [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Sending to 72.190.32.250:5060 (NAT) [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Initializing initreq for method INVITE - callid 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Using INVITE request as basis request - 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:10] DEBUG[85170][C-000006f8] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:35:10] DEBUG[85170][C-000006f8] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found peer '100' for '100' from 72.190.32.250:5060 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f4e80014da0' [Dec 9 19:35:10] DEBUG[85170][C-000006f8] res_rtp_asterisk.c: Allocated port 16466 for RTP instance '0x7f4e80014da0' [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: RTP instance '0x7f4e80014da0' is setup and ready to go [Dec 9 19:35:10] DEBUG[85170][C-000006f8] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:35:10] DEBUG[85170][C-000006f8] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f4e80014da0' [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] netsock2.c: Using SIP RTP CoS mark 5 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Setting NAT on RTP to On [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing session-level SDP o=100 8001 8000 IN IP4 192.168.1.6... OK. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED OR FAILED. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] netsock2.c: Splitting '192.168.1.6' into... [Dec 9 19:35:10] DEBUG[85170][C-000006f8] netsock2.c: ...host '192.168.1.6' and port ''. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.6... OK. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found RTP audio format 125 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Don't have a default tx payload type 125 format for m type on 0x7f4e89363180 [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found RTP audio format 0 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f4e89363180 [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found RTP audio format 97 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Setting tx payload type 97 based on m type on 0x7f4e89363180 [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found RTP audio format 18 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Setting tx payload type 18 based on m type on 0x7f4e89363180 [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found RTP audio format 9 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Setting tx payload type 9 based on m type on 0x7f4e89363180 [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found RTP audio format 101 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7f4e89363180 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found audio description format opus for ID 125 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:125 opus/48000/2... OK. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found audio description format PCMU for ID 0 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found audio description format iLBC for ID 97 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 iLBC/8000... OK. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=fmtp:97 mode=30... OK. [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found audio description format G729 for ID 18 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK. [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found audio description format G722 for ID 9 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Capabilities: us - (ulaw|ilbc|g729), peer - audio=(ulaw|g722|g729|ilbc|opus)/video=(nothing)/text=(nothing), combined - (ulaw|ilbc|g729) [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Dec 9 19:35:10] DEBUG[85170][C-000006f8] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] acl.c: For destination '192.168.1.6', our source address is '207.91.156.217'. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4e80014da0' [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Peer audio RTP is at port 192.168.1.6:5006 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Copying payload 0 (0x7f4e80004c58) from 0x7f4e89363180 to 0x7f4e80014f68 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Copying payload 9 (0x7f4e800323c8) from 0x7f4e89363180 to 0x7f4e80014f68 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Copying payload 18 (0x7f4e80040118) from 0x7f4e89363180 to 0x7f4e80014f68 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Copying payload 97 (0x7f4e8002f108) from 0x7f4e89363180 to 0x7f4e80014f68 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Copying payload 101 (0x7f4e8002f998) from 0x7f4e89363180 to 0x7f4e80014f68 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] rtp_engine.c: Copying payload 125 (0x7f4e80011998) from 0x7f4e89363180 to 0x7f4e80014f68 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f4e80014da0' [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: We're settling with these formats: (ulaw|ilbc|g729) [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Checking SIP call limits for device 100 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Updating call counter for incoming call [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Call from peer '100' is 1 out of 2147483647 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:35:10] DEBUG[85170][C-000006f8] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:35:10] DEBUG[85170][C-000006f8] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:35:10] DEBUG[85170][C-000006f8] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Looking for 8018309060 in from-internal (domain testbed2.primevox.net) [Dec 9 19:35:10] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Dec 9 19:35:10] DEBUG[85149] chan_sip.c: Checking device state for peer 100 [Dec 9 19:35:10] DEBUG[85149] devicestate.c: Changing state for SIP/100 - state 2 (In use) [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: DeviceStateChange Privilege: call,all Device: SIP/100 State: INUSE [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Incoming INVITE with 'timer' option supported [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: from-internal Hint: SIP/100 Status: 1 StatusText: InUse [Dec 9 19:35:10] DEBUG[85210] app_queue.c: Device 'SIP/100' changed to state '2' (In use) [Dec 9 19:35:10] DEBUG[85210] app_queue.c: Device 'Queue:Technical Support_avail' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 9 19:35:10] DEBUG[85151] app_queue.c: Extension '100@from-internal' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: QueueMemberStatus Privilege: agent,all Ringinuse: 0 StateInterface: SIP/100 CallsTaken: 0 Queue: Technical Support Membership: static Status: 2 MemberName: SIP/100 Penalty: 0 Interface: SIP/100 LastCall: 0 InCall: 0 Paused: 0 PausedReason: [Dec 9 19:35:10] DEBUG[85138] threadpool.c: Increasing threadpool stasis-core's size by 1 [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: DeviceStateChange Privilege: call,all Device: Queue:Technical Support_avail State: INUSE [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: *** Our native formats are (ulaw) [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: *** Joint capabilities are (ulaw|ilbc|g729) [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: *** Our capabilities are (ulaw|ilbc|g729) [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: This channel will not be able to handle video. [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newchannel Privilege: call,all Channel: SIP/100-0000006b ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-internal Exten: 8018309060 Priority: 1 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-internal Exten: 8018309060 Priority: 1 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: SIPURI Value: sip:100@192.168.1.6:5060 [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: SIP/100-0000006b: New call is still down.... Trying... [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: <--- Transmitting (NAT) to 72.190.32.250:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK1641505246;received=72.190.32.250;rport=5060 From: "Local1" ;tag=2051421774 To: Call-ID: 1453900646-5060-25@BJC.BGI.B.G CSeq: 220 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:10] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Dec 9 19:35:10] DEBUG[85149] chan_sip.c: Checking device state for peer 100 [Dec 9 19:35:10] DEBUG[85149] devicestate.c: Changing state for SIP/100 - state 2 (In use) [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-internal Exten: 8018309060 Priority: 1 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: SIPDOMAIN Value: testbed2.primevox.net [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK72329b79;rport=5060 From: "asterisk" ;tag=as2305471a To: ;tag=1850268267 Call-ID: 32e848e76ce7b11c392d28cc3bbfd998@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK72329b79;rport=5060 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as2305471a [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 3 [ 49]: To: ;tag=1850268267 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 32e848e76ce7b11c392d28cc3bbfd998@207.91.156.217:5060 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-internal Exten: 8018309060 Priority: 1 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: SIPCALLID Value: 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: = Looking for Call ID: 32e848e76ce7b11c392d28cc3bbfd998@207.91.156.217:5060 (Checking To) --From tag as2305471a --To-tag 1850268267 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #45 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Stopping retransmission on '32e848e76ce7b11c392d28cc3bbfd998@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newstate Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-internal Exten: 8018309060 Priority: 1 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Destroying SIP dialog 32e848e76ce7b11c392d28cc3bbfd998@207.91.156.217:5060 [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '32e848e76ce7b11c392d28cc3bbfd998@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: SuccessfulAuth Privilege: security,all EventTV: 2016-12-09T19:35:10.438+0000 Severity: Informational Service: SIP EventVersion: 1 AccountID: 8018309060 SessionID: 0x7f4e8002ab30 LocalAddress: IPV4/UDP/207.91.156.217/5060 RemoteAddress: IPV4/UDP/72.190.32.250/5060 UsingPassword: 1 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx_variables.c: Result of 'EXTEN' is '8018309060' [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx.c: Launching 'Goto' [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx.c: Executing [8018309060@from-internal:1] Goto("SIP/100-0000006b", "dial-out-100,8018309060,1") in new stack [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx_builtins.c: Goto (dial-out-100,8018309060,1) [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx_variables.c: Result of 'EXTEN' is '8018309060' [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx.c: Launching 'Set' [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx.c: Executing [8018309060@dial-out-100:1] Set("SIP/100-0000006b", "temp=18018309060") in new stack [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx.c: Launching 'Goto' [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx.c: Executing [8018309060@dial-out-100:2] Goto("SIP/100-0000006b", "_1XXXXXX.,modified") in new stack [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx_builtins.c: Goto (dial-out-100,_1XXXXXX.,2) [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx_variables.c: Function CDR(cdr_direction) result is '' [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx_variables.c: Expression result is '0' [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx.c: Launching 'GotoIf' [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx.c: Executing [_1XXXXXX.@dial-out-100:2] GotoIf("SIP/100-0000006b", "0?keepDir") in new stack [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx_builtins.c: Not taking any branch [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx.c: Launching 'Set' [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx.c: Executing [_1XXXXXX.@dial-out-100:3] Set("SIP/100-0000006b", "CDR(cdr_direction)=2") in new stack [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx_variables.c: Result of 'temp' is '18018309060' [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx.c: Launching 'Set' [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx.c: Executing [_1XXXXXX.@dial-out-100:4] Set("SIP/100-0000006b", "CDR(cdr_outbound_dialed_phonenumber)=18018309060") in new stack [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx.c: Launching 'Set' [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx.c: Executing [_1XXXXXX.@dial-out-100:5] Set("SIP/100-0000006b", "CDR(cdr_outbound_from_extension)=100") in new stack [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-internal Exten: 8018309060 Priority: 1 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Extension: 8018309060 Application: Goto AppData: dial-out-100,8018309060,1 [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: 8018309060 Priority: 1 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Extension: 8018309060 Application: Set AppData: temp=18018309060 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx.c: Launching 'Set' [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx.c: Executing [_1XXXXXX.@dial-out-100:6] Set("SIP/100-0000006b", "CDR(cdr_application)=Outbound Dial") in new stack [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: 8018309060 Priority: 1 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: temp Value: 18018309060 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx.c: Launching 'Set' [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: 8018309060 Priority: 2 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Extension: 8018309060 Application: Goto AppData: _1XXXXXX.,modified [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx.c: Executing [_1XXXXXX.@dial-out-100:7] Set("SIP/100-0000006b", "CALLERID(all)="Luke Apartment" <9726001152>") in new stack [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 2 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Extension: _1XXXXXX. Application: GotoIf AppData: 0?keepDir [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 3 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Extension: _1XXXXXX. Application: Set AppData: CDR(cdr_direction)=2 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx.c: Launching 'Set' [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx.c: Executing [_1XXXXXX.@dial-out-100:8] Set("SIP/100-0000006b", "CDR(cdr_outbound_did)=9726001152") in new stack [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 4 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Extension: _1XXXXXX. Application: Set AppData: CDR(cdr_outbound_dialed_phonenumber)=18018309060 [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 5 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Extension: _1XXXXXX. Application: Set AppData: CDR(cdr_outbound_from_extension)=100 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx_variables.c: Result of 'temp' is '18018309060' [Dec 9 19:35:10] DEBUG[114314][C-000006f8] pbx.c: Launching 'Dial' [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] pbx.c: Executing [_1XXXXXX.@dial-out-100:9] Dial("SIP/100-0000006b", "SIP/flowroute/18018309060") in new stack [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Asked to create a SIP channel with formats: (ulaw) [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Allocating new SIP dialog for 3c18d6ff0668c0120ecee87506cecf00@207.91.156.217:5060 - INVITE (No RTP) [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 6 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Extension: _1XXXXXX. Application: Set AppData: CDR(cdr_application)=Outbound Dial [Dec 9 19:35:10] DEBUG[114314][C-000006f8] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f4ecc03fd90' [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 7 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Extension: _1XXXXXX. Application: Set AppData: CALLERID(all)="Luke Apartment" <9726001152> [Dec 9 19:35:10] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Allocated port 16066 for RTP instance '0x7f4ecc03fd90' [Dec 9 19:35:10] DEBUG[114314][C-000006f8] rtp_engine.c: RTP instance '0x7f4ecc03fd90' is setup and ready to go [Dec 9 19:35:10] DEBUG[114314][C-000006f8] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: NewCallerid Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 7 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Dec 9 19:35:10] DEBUG[114314][C-000006f8] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:35:10] DEBUG[114314][C-000006f8] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 8 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Extension: _1XXXXXX. Application: Set AppData: CDR(cdr_outbound_did)=9726001152 [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Extension: _1XXXXXX. Application: Dial AppData: SIP/flowroute/18018309060 [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: DIALSTATUS Value: [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: DIALEDPEERNUMBER Value: [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: DIALEDPEERNAME Value: [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: ANSWEREDTIME Value: [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: DIALEDTIME Value: [Dec 9 19:35:10] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f4ecc03fd90' [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] netsock2.c: Using SIP RTP CoS mark 5 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Setting NAT on RTP to On [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:10] DEBUG[114314][C-000006f8] acl.c: For destination '216.115.69.144', our source address is '207.91.156.217'. [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Setting NAT on RTP to On [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: SIP call-id changed from '3c18d6ff0668c0120ecee87506cecf00@207.91.156.217:5060' to '076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060' [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: *** Our native formats are (ulaw) [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: *** Joint capabilities are (ulaw) [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: *** Our capabilities are (ulaw) [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: *** Our preferred formats from the incoming channel are (ulaw) [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: This channel will not be able to handle video. [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newchannel Privilege: call,all Channel: SIP/flowroute-0000006c ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-trunk Exten: s Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] channel_internal_api.c: Channel Call ID changing from [C-000006f8] to [C-000006f8] [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-trunk Exten: s Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: SIPCALLID Value: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: DIALEDPEERNUMBER Value: flowroute/18018309060 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Outgoing Call for 18018309060 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Updating call counter for outgoing call [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/flowroute-0000006c ChannelState: 0 ChannelStateDesc: Down CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: _1XXXXXX. Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Extension: _1XXXXXX. Application: AppDial AppData: (Outgoing Line) [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: NewCallerid Privilege: call,all Channel: SIP/flowroute-0000006c ChannelState: 0 ChannelStateDesc: Down CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: _1XXXXXX. Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: NewConnectedLine Privilege: call,all Channel: SIP/flowroute-0000006c ChannelState: 0 ChannelStateDesc: Down CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: _1XXXXXX. Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: ** Our capability: (ulaw) Video flag: False Text flag: False [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: ** Our prefcodec: (ulaw) [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] chan_sip.c: Audio is at 16066 [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] chan_sip.c: Adding codec ulaw to SDP [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: -- Done with adding codecs to SDP [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Initializing initreq for method INVITE - callid 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 0 [ 48]: INVITE sip:18018309060@sip.flowroute.com SIP/2.0 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK2f355779;rport [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 3 [ 67]: From: "Luke Apartment" ;tag=as76bc5363 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 4 [ 39]: To: [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 6 [ 61]: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:35:10 GMT [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 12 [101]: Remote-Party-ID: "Luke Apartment" ;party=calling;privacy=off;screen=no [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] chan_sip.c: Reliably Transmitting (NAT) to 216.115.69.144:5060: INVITE sip:18018309060@sip.flowroute.com SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK2f355779;rport Max-Forwards: 70 From: "Luke Apartment" ;tag=as76bc5363 To: Contact: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:35:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "Luke Apartment" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 244 v=0 o=root 1400167239 1400167239 IN IP4 207.91.156.217 s=Asterisk PBX 13.13.1 c=IN IP4 207.91.156.217 t=0 0 m=audio 16066 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #75 [Dec 9 19:35:10] DEBUG[114314][C-000006f8] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:35:10] VERBOSE[114314][C-000006f8] app_dial.c: Called SIP/flowroute/18018309060 [Dec 9 19:35:10] DEBUG[85138] threadpool.c: Increasing threadpool stasis-core's size by 1 [Dec 9 19:35:10] DEBUG[94779] manager.c: Examining AMI event: Event: DialBegin Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 DestChannel: SIP/flowroute-0000006c DestChannelState: 0 DestChannelStateDesc: Down DestCallerIDNum: _1XXXXXX. DestCallerIDName: DestConnectedLineNum: 9726001152 DestConnectedLineName: Luke Apartment DestLanguage: en DestAccountCode: DestContext: from-trunk DestExten: _1XXXXXX. DestPriority: 1 DestUniqueid: 1481312110.218 DestLinkedid: 1481312110.217 DialString: flowroute/18018309060 [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK2f355779;rport=5060 From: "Luke Apartment" ;tag=as76bc5363 To: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK2f355779;rport=5060 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 2 [ 67]: From: "Luke Apartment" ;tag=as76bc5363 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 3 [ 39]: To: [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: --- (7 headers 0 lines) --- [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: = Looking for Call ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 (Checking To) --From tag as76bc5363 --To-tag [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: *** SIP TIMER: Cancelling retransmission #75 - INVITE (got response) [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060' Request 102: Found [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: SIP response 100 to standard invite [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK2f355779;rport=5060 From: "Luke Apartment" ;tag=as76bc5363 To: ;tag=870a5b262384e4f9f82f59836d699db5.5857 Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="sip.flowroute.com", nonce="WEsImlhLB25P25UP37SLPSvA1iDH2/AT", qop="auth" Content-Length: 0 <-------------> [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 0 [ 41]: SIP/2.0 407 Proxy Authentication Required [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK2f355779;rport=5060 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 2 [ 67]: From: "Luke Apartment" ;tag=as76bc5363 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 3 [ 81]: To: ;tag=870a5b262384e4f9f82f59836d699db5.5857 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 6 [106]: Proxy-Authenticate: Digest realm="sip.flowroute.com", nonce="WEsImlhLB25P25UP37SLPSvA1iDH2/AT", qop="auth" [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: --- (8 headers 0 lines) --- [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: = Looking for Call ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 (Checking To) --From tag as76bc5363 --To-tag 870a5b262384e4f9f82f59836d699db5.5857 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Acked pending invite 102 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Stopping retransmission on '076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: SIP response 407 to standard invite [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Transmitting (NAT) to 216.115.69.144:5060: ACK sip:18018309060@sip.flowroute.com SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK2f355779;rport Max-Forwards: 70 From: "Luke Apartment" ;tag=as76bc5363 To: ;tag=870a5b262384e4f9f82f59836d699db5.5857 Contact: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 --- [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Trying to put 'ACK sip:180' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Auth attempt 1 on INVITE [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: ** Our capability: (ulaw) Video flag: False Text flag: False [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: ** Our prefcodec: (ulaw) [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Audio is at 16066 [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Adding codec ulaw to SDP [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: -- Done with adding codecs to SDP [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Dec 9 19:35:10] VERBOSE[85170][C-000006f8] chan_sip.c: Reliably Transmitting (NAT) to 216.115.69.144:5060: INVITE sip:18018309060@sip.flowroute.com SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK124f9a54;rport Max-Forwards: 70 From: "Luke Apartment" ;tag=as76bc5363 To: Contact: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 13.13.1 Proxy-Authorization: Digest username="23616016", realm="sip.flowroute.com", algorithm=MD5, uri="sip:18018309060@sip.flowroute.com", nonce="WEsImlhLB25P25UP37SLPSvA1iDH2/AT", response="6e59535fd699d5bef73b33d0ef1d27e9", qop=auth, cnonce="46e1f6f1", nc=00000001 Date: Fri, 09 Dec 2016 19:35:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "Luke Apartment" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 244 v=0 o=root 1400167239 1400167240 IN IP4 207.91.156.217 s=Asterisk PBX 13.13.1 c=IN IP4 207.91.156.217 t=0 0 m=audio 16066 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #19 [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK124f9a54;rport=5060 From: "Luke Apartment" ;tag=as76bc5363 To: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 CSeq: 103 INVITE Content-Length: 0 <-------------> [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK124f9a54;rport=5060 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 2 [ 67]: From: "Luke Apartment" ;tag=as76bc5363 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 3 [ 39]: To: [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Dec 9 19:35:10] VERBOSE[85170] chan_sip.c: --- (7 headers 0 lines) --- [Dec 9 19:35:10] DEBUG[85170] chan_sip.c: = Looking for Call ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 (Checking To) --From tag as76bc5363 --To-tag [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: *** SIP TIMER: Cancelling retransmission #19 - INVITE (got response) [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060' Request 103: Found [Dec 9 19:35:10] DEBUG[85170][C-000006f8] chan_sip.c: SIP response 100 to standard invite [Dec 9 19:35:11] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 180 Ringing From: "Luke Apartment" ;tag=as76bc5363 To: ;tag=gK00db684d Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK124f9a54;rport=5060 Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 CSeq: 103 INVITE Record-Route: Record-Route: Contact: Content-Length: 0 <-------------> [Dec 9 19:35:11] DEBUG[85170] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Dec 9 19:35:11] DEBUG[85170] chan_sip.c: Header 1 [ 67]: From: "Luke Apartment" ;tag=as76bc5363 [Dec 9 19:35:11] DEBUG[85170] chan_sip.c: Header 2 [ 54]: To: ;tag=gK00db684d [Dec 9 19:35:11] DEBUG[85170] chan_sip.c: Header 3 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK124f9a54;rport=5060 [Dec 9 19:35:11] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:11] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Dec 9 19:35:11] DEBUG[85170] chan_sip.c: Header 6 [ 42]: Record-Route: [Dec 9 19:35:11] DEBUG[85170] chan_sip.c: Header 7 [ 42]: Record-Route: [Dec 9 19:35:11] DEBUG[85170] chan_sip.c: Header 8 [ 45]: Contact: [Dec 9 19:35:11] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:35:11] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:35:11] DEBUG[85170] chan_sip.c: = Looking for Call ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 (Checking To) --From tag as76bc5363 --To-tag gK00db684d [Dec 9 19:35:11] DEBUG[85170][C-000006f8] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060' Request 103: Found [Dec 9 19:35:11] DEBUG[85170][C-000006f8] chan_sip.c: SIP response 180 to standard invite [Dec 9 19:35:11] DEBUG[85170][C-000006f8] sip/route.c: sip_route_process_header: [Dec 9 19:35:11] DEBUG[85170][C-000006f8] sip/route.c: sip_route_process_header: [Dec 9 19:35:11] VERBOSE[85170][C-000006f8] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:11] VERBOSE[85170][C-000006f8] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:11] VERBOSE[114314][C-000006f8] app_dial.c: SIP/flowroute-0000006c is ringing [Dec 9 19:35:11] DEBUG[114314][C-000006f8] rtp_engine.c: Setting early bridge SDP of 'SIP/100-0000006b' with that of 'SIP/flowroute-0000006c' [Dec 9 19:35:11] VERBOSE[114314][C-000006f8] chan_sip.c: <--- Transmitting (NAT) to 72.190.32.250:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK1641505246;received=72.190.32.250;rport=5060 From: "Local1" ;tag=2051421774 To: ;tag=as0deed62b Call-ID: 1453900646-5060-25@BJC.BGI.B.G CSeq: 220 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Dec 9 19:35:11] DEBUG[114314][C-000006f8] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:11] DEBUG[94779] manager.c: Examining AMI event: Event: Newstate Privilege: call,all Channel: SIP/flowroute-0000006c ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: _1XXXXXX. Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 [Dec 9 19:35:11] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - flowroute [Dec 9 19:35:11] DEBUG[85149] chan_sip.c: Checking device state for peer flowroute [Dec 9 19:35:11] DEBUG[85149] devicestate.c: Changing state for SIP/flowroute - state 1 (Not in use) [Dec 9 19:35:13] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 180 Ringing From: "Luke Apartment" ;tag=as76bc5363 To: ;tag=gK00db684d Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK124f9a54;rport=5060 Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 CSeq: 103 INVITE Record-Route: Record-Route: Contact: Content-Length: 172 Content-Type: application/sdp v=0 o=- 791026 917100 IN IP4 74.120.93.26 s=- c=IN IP4 74.120.93.26 t=0 0 m=audio 39732 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Header 1 [ 67]: From: "Luke Apartment" ;tag=as76bc5363 [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Header 2 [ 54]: To: ;tag=gK00db684d [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Header 3 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK124f9a54;rport=5060 [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Header 6 [ 42]: Record-Route: [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Header 7 [ 42]: Record-Route: [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Header 8 [ 45]: Contact: [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Header 9 [ 19]: Content-Length: 172 [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Header 11 [ 0]: [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Body 0 [ 3]: v=0 [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Body 1 [ 37]: o=- 791026 917100 IN IP4 74.120.93.26 [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Body 2 [ 3]: s=- [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Body 3 [ 21]: c=IN IP4 74.120.93.26 [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Body 5 [ 27]: m=audio 39732 RTP/AVP 0 101 [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Body 6 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Body 7 [ 15]: a=fmtp:101 0-15 [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: Body 8 [ 10]: a=ptime:20 [Dec 9 19:35:13] VERBOSE[85170] chan_sip.c: --- (11 headers 9 lines) --- [Dec 9 19:35:13] DEBUG[85170] chan_sip.c: = Looking for Call ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 (Checking To) --From tag as76bc5363 --To-tag gK00db684d [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060' Request 103: Found [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: SIP response 180 to standard invite [Dec 9 19:35:13] DEBUG[85170][C-000006f8] sip/route.c: sip_route_process_header: [Dec 9 19:35:13] DEBUG[85170][C-000006f8] sip/route.c: sip_route_process_header: [Dec 9 19:35:13] VERBOSE[85170][C-000006f8] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:13] VERBOSE[85170][C-000006f8] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: Processing session-level SDP o=- 791026 917100 IN IP4 74.120.93.26... OK. [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [Dec 9 19:35:13] DEBUG[85170][C-000006f8] netsock2.c: Splitting '74.120.93.26' into... [Dec 9 19:35:13] DEBUG[85170][C-000006f8] netsock2.c: ...host '74.120.93.26' and port ''. [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: Processing session-level SDP c=IN IP4 74.120.93.26... OK. [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Dec 9 19:35:13] VERBOSE[85170][C-000006f8] chan_sip.c: Found RTP audio format 0 [Dec 9 19:35:13] DEBUG[85170][C-000006f8] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f4e893623e0 [Dec 9 19:35:13] VERBOSE[85170][C-000006f8] chan_sip.c: Found RTP audio format 101 [Dec 9 19:35:13] DEBUG[85170][C-000006f8] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7f4e893623e0 [Dec 9 19:35:13] VERBOSE[85170][C-000006f8] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Dec 9 19:35:13] VERBOSE[85170][C-000006f8] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Dec 9 19:35:13] VERBOSE[85170][C-000006f8] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Dec 9 19:35:13] DEBUG[85170][C-000006f8] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Dec 9 19:35:13] DEBUG[85170][C-000006f8] acl.c: For destination '74.120.93.26', our source address is '207.91.156.217'. [Dec 9 19:35:13] DEBUG[85170][C-000006f8] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4ecc03fd90' [Dec 9 19:35:13] VERBOSE[85170][C-000006f8] chan_sip.c: Peer audio RTP is at port 74.120.93.26:39732 [Dec 9 19:35:13] DEBUG[85170][C-000006f8] rtp_engine.c: Copying payload 0 (0x307cf18) from 0x7f4e893623e0 to 0x7f4ecc03ff58 [Dec 9 19:35:13] DEBUG[85170][C-000006f8] rtp_engine.c: Copying payload 101 (0x7f4e8002a8e8) from 0x7f4e893623e0 to 0x7f4ecc03ff58 [Dec 9 19:35:13] DEBUG[85170][C-000006f8] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f4ecc03fd90' [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: We're settling with these formats: (ulaw) [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: We have an owner, now see if we need to change this call [Dec 9 19:35:13] DEBUG[85170][C-000006f8] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Dec 9 19:35:13] VERBOSE[114314][C-000006f8] app_dial.c: SIP/flowroute-0000006c is ringing [Dec 9 19:35:13] DEBUG[114314][C-000006f8] rtp_engine.c: Setting early bridge SDP of 'SIP/100-0000006b' with that of 'SIP/flowroute-0000006c' [Dec 9 19:35:13] VERBOSE[114314][C-000006f8] app_dial.c: SIP/flowroute-0000006c is making progress passing it to SIP/100-0000006b [Dec 9 19:35:13] DEBUG[114314][C-000006f8] rtp_engine.c: Setting early bridge SDP of 'SIP/100-0000006b' with that of 'SIP/flowroute-0000006c' [Dec 9 19:35:13] DEBUG[114314][C-000006f8] chan_sip.c: ** Our capability: (ulaw|ilbc|g729) Video flag: True Text flag: True [Dec 9 19:35:13] DEBUG[114314][C-000006f8] chan_sip.c: ** Our prefcodec: (nothing) [Dec 9 19:35:13] VERBOSE[114314][C-000006f8] chan_sip.c: Audio is at 16466 [Dec 9 19:35:13] VERBOSE[114314][C-000006f8] chan_sip.c: Adding codec ulaw to SDP [Dec 9 19:35:13] VERBOSE[114314][C-000006f8] chan_sip.c: Adding codec ilbc to SDP [Dec 9 19:35:13] VERBOSE[114314][C-000006f8] chan_sip.c: Adding codec g729 to SDP [Dec 9 19:35:13] VERBOSE[114314][C-000006f8] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 9 19:35:13] DEBUG[114314][C-000006f8] chan_sip.c: -- Done with adding codecs to SDP [Dec 9 19:35:13] DEBUG[114314][C-000006f8] chan_sip.c: Setting framing on incoming call: 30 [Dec 9 19:35:13] DEBUG[114314][C-000006f8] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|ilbc|g729) [Dec 9 19:35:13] VERBOSE[114314][C-000006f8] chan_sip.c: <--- Transmitting (NAT) to 72.190.32.250:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK1641505246;received=72.190.32.250;rport=5060 From: "Local1" ;tag=2051421774 To: ;tag=as0deed62b Call-ID: 1453900646-5060-25@BJC.BGI.B.G CSeq: 220 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 327 v=0 o=root 455117568 455117568 IN IP4 207.91.156.217 s=Asterisk PBX 13.13.1 c=IN IP4 207.91.156.217 t=0 0 m=audio 16466 RTP/AVP 0 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 a=maxptime:150 a=sendrecv <------------> [Dec 9 19:35:13] DEBUG[114314][C-000006f8] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:13] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: 0x7f4e800b2b70 -- Probation learning mode pass with source address 72.190.32.250:5006 [Dec 9 19:35:13] DEBUG[114314][C-000006f8] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Dec 9 19:35:13] DEBUG[114314][C-000006f8] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:35:13] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4e80014da0' [Dec 9 19:35:13] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Ooh, format changed from none to ulaw [Dec 9 19:35:13] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f4ecc03fd90' [Dec 9 19:35:13] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: 0x7f4ecc041f80 -- Probation learning mode pass with source address 74.120.93.26:39732 [Dec 9 19:35:13] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Ooh, format changed from none to ulaw [Dec 9 19:35:13] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: 0x7f4e800b2b70 -- Probation learning mode pass with source address 72.190.32.250:5006 [Dec 9 19:35:18] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:16467' into... [Dec 9 19:35:18] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '16467'. [Dec 9 19:35:18] DEBUG[85207] netsock2.c: Splitting '72.190.32.250:5007' into... [Dec 9 19:35:18] DEBUG[85207] netsock2.c: ...host '72.190.32.250' and port '5007'. [Dec 9 19:35:18] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPSent Privilege: reporting,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 To: 72.190.32.250:5007 From: 207.91.156.217:16467 SSRC: 0x0b2b967f PT: 200(SR) ReportCount: 1 SentNTP: 1481312118.3834408960 SentRTP: 39600 SentPackets: 166 SentOctets: 39840 Report0SourceSSRC: 0x501ee4db Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 38438 Report0SequenceNumberCycles: 0 Report0IAJitter: 30 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:35:18] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:16067' into... [Dec 9 19:35:18] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '16067'. [Dec 9 19:35:18] DEBUG[85207] netsock2.c: Splitting '74.120.93.26:39733' into... [Dec 9 19:35:18] DEBUG[85207] netsock2.c: ...host '74.120.93.26' and port '39733'. [Dec 9 19:35:18] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPSent Privilege: reporting,all Channel: SIP/flowroute-0000006c ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: _1XXXXXX. Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 To: 74.120.93.26:39733 From: 207.91.156.217:16067 SSRC: 0x3fa1dc58 PT: 200(SR) ReportCount: 1 SentNTP: 1481312118.3834732544 SentRTP: 58934080 SentPackets: 167 SentOctets: 40080 Report0SourceSSRC: 0xe409d998 Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 355 Report0SequenceNumberCycles: 0 Report0IAJitter: 0 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:35:18] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Dec 9 19:35:18] DEBUG[85207] netsock2.c: Splitting '72.190.32.250:5007' into... [Dec 9 19:35:18] DEBUG[85207] netsock2.c: ...host '72.190.32.250' and port '5007'. [Dec 9 19:35:18] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:16467' into... [Dec 9 19:35:18] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '16467'. [Dec 9 19:35:18] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPReceived Privilege: reporting,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 To: 207.91.156.217:16467 From: 72.190.32.250:5007 RTT: 0.0000 SSRC: 0x501ee4db PT: 200(SR) ReportCount: 1 SentNTP: 1481312118.17590784552960 SentRTP: 58934320 SentPackets: 167 SentOctets: 40080 Report0SourceSSRC: 0x0b2b967f Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 25445 Report0SequenceNumberCycles: 0 Report0IAJitter: 84 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 407d926766bba0734436b9ac5d8ff787@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:18] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: SIP call-id changed from '407d926766bba0734436b9ac5d8ff787@207.91.156.217:5060' to '5454a38d035ed41703ce37ac2779a680@207.91.156.217:5060' [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 5454a38d035ed41703ce37ac2779a680@207.91.156.217:5060 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 0 [ 40]: OPTIONS sip:100@192.168.1.6:5060 SIP/2.0 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK2486fa82;rport [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as5a97f2cd [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 4 [ 30]: To: [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 5454a38d035ed41703ce37ac2779a680@207.91.156.217:5060 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:35:18 GMT [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:35:18] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 72.190.32.250:5060: OPTIONS sip:100@192.168.1.6:5060 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK2486fa82;rport Max-Forwards: 70 From: "asterisk" ;tag=as5a97f2cd To: Contact: Call-ID: 5454a38d035ed41703ce37ac2779a680@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:35:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #79 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:18] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK2486fa82;rport=5060 From: "asterisk" ;tag=as5a97f2cd To: ;tag=1062141561 Call-ID: 5454a38d035ed41703ce37ac2779a680@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK2486fa82;rport=5060 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as5a97f2cd [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 3 [ 45]: To: ;tag=1062141561 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 5454a38d035ed41703ce37ac2779a680@207.91.156.217:5060 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:35:18] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: = Looking for Call ID: 5454a38d035ed41703ce37ac2779a680@207.91.156.217:5060 (Checking To) --From tag as5a97f2cd --To-tag 1062141561 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #79 [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Stopping retransmission on '5454a38d035ed41703ce37ac2779a680@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:18] DEBUG[85170] chan_sip.c: Destroying SIP dialog 5454a38d035ed41703ce37ac2779a680@207.91.156.217:5060 [Dec 9 19:35:18] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '5454a38d035ed41703ce37ac2779a680@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:35:19] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:35:19] VERBOSE[114314][C-000006f8] app_dial.c: SIP/flowroute-0000006c requested media update control 26, passing it to SIP/100-0000006b [Dec 9 19:35:19] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Changing ssrc from 187405951 to 1502779067 due to a source change [Dec 9 19:35:19] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 200 OK From: "Luke Apartment" ;tag=as76bc5363 To: ;tag=gK00db684d Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK124f9a54;rport=5060 Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 CSeq: 103 INVITE Record-Route: Record-Route: Contact: Session-Expires: 1800;refresher=uas Content-Length: 172 Content-Type: application/sdp Supported: timer v=0 o=- 791026 917100 IN IP4 74.120.93.26 s=- c=IN IP4 74.120.93.26 t=0 0 m=audio 39732 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 1 [ 67]: From: "Luke Apartment" ;tag=as76bc5363 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 2 [ 54]: To: ;tag=gK00db684d [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 3 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK124f9a54;rport=5060 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 6 [ 42]: Record-Route: [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 7 [ 42]: Record-Route: [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 8 [ 45]: Contact: [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 10 [ 19]: Content-Length: 172 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 12 [ 16]: Supported: timer [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 13 [ 0]: [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Body 0 [ 3]: v=0 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Body 1 [ 37]: o=- 791026 917100 IN IP4 74.120.93.26 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Body 2 [ 3]: s=- [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Body 3 [ 21]: c=IN IP4 74.120.93.26 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Body 5 [ 27]: m=audio 39732 RTP/AVP 0 101 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Body 6 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Body 7 [ 15]: a=fmtp:101 0-15 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Body 8 [ 10]: a=ptime:20 [Dec 9 19:35:19] VERBOSE[85170] chan_sip.c: --- (13 headers 9 lines) --- [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: = Looking for Call ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 (Checking To) --From tag as76bc5363 --To-tag gK00db684d [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: Acked pending invite 103 [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: Stopping retransmission on '076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060' of Request 103: Match Found [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: SIP response 200 to standard invite [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: Call 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 responded to our reinvite without changing SDP version; ignoring SDP. [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: Updating call counter for outgoing call [Dec 9 19:35:19] DEBUG[85170][C-000006f8] sip/route.c: sip_route_process_header: [Dec 9 19:35:19] DEBUG[85170][C-000006f8] sip/route.c: sip_route_process_header: [Dec 9 19:35:19] VERBOSE[85170][C-000006f8] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:19] VERBOSE[85170][C-000006f8] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: Session-Expires: 1800 [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: Refresher: UAS [Dec 9 19:35:19] VERBOSE[85170][C-000006f8] chan_sip.c: Transmitting (NAT) to 216.115.69.144:5060: ACK sip:+18018309060@74.120.93.24:5060 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK1381bfa3;rport Route: , Max-Forwards: 70 From: "Luke Apartment" ;tag=as76bc5363 To: ;tag=gK00db684d Contact: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 --- [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: Trying to put 'ACK sip:+18' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Session timer started: 62 - 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 1768000ms [Dec 9 19:35:19] VERBOSE[114314][C-000006f8] app_dial.c: SIP/flowroute-0000006c answered SIP/100-0000006b [Dec 9 19:35:19] DEBUG[114314][C-000006f8] rtp_engine.c: Setting early bridge SDP of 'SIP/100-0000006b' with that of 'SIP/flowroute-0000006c' [Dec 9 19:35:19] DEBUG[114314][C-000006f8] chan_sip.c: SIP answering channel: SIP/100-0000006b [Dec 9 19:35:19] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Setting the marker bit due to a source update [Dec 9 19:35:19] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - flowroute [Dec 9 19:35:19] DEBUG[114314][C-000006f8] chan_sip.c: ** Our capability: (ulaw|ilbc|g729) Video flag: True Text flag: True [Dec 9 19:35:19] DEBUG[114314][C-000006f8] chan_sip.c: ** Our prefcodec: (nothing) [Dec 9 19:35:19] DEBUG[85149] chan_sip.c: Checking device state for peer flowroute [Dec 9 19:35:19] DEBUG[85149] devicestate.c: Changing state for SIP/flowroute - state 1 (Not in use) [Dec 9 19:35:19] VERBOSE[114314][C-000006f8] chan_sip.c: Audio is at 16466 [Dec 9 19:35:19] VERBOSE[114314][C-000006f8] chan_sip.c: Adding codec ulaw to SDP [Dec 9 19:35:19] VERBOSE[114314][C-000006f8] chan_sip.c: Adding codec ilbc to SDP [Dec 9 19:35:19] VERBOSE[114314][C-000006f8] chan_sip.c: Adding codec g729 to SDP [Dec 9 19:35:19] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Dec 9 19:35:19] DEBUG[85149] chan_sip.c: Checking device state for peer 100 [Dec 9 19:35:19] VERBOSE[114314][C-000006f8] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 9 19:35:19] DEBUG[114314][C-000006f8] chan_sip.c: -- Done with adding codecs to SDP [Dec 9 19:35:19] DEBUG[114314][C-000006f8] chan_sip.c: Setting framing on incoming call: 30 [Dec 9 19:35:19] DEBUG[85149] devicestate.c: Changing state for SIP/100 - state 2 (In use) [Dec 9 19:35:19] DEBUG[114314][C-000006f8] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|ilbc|g729) [Dec 9 19:35:19] VERBOSE[114314][C-000006f8] chan_sip.c: <--- Reliably Transmitting (NAT) to 72.190.32.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK1641505246;received=72.190.32.250;rport=5060 From: "Local1" ;tag=2051421774 To: ;tag=as0deed62b Call-ID: 1453900646-5060-25@BJC.BGI.B.G CSeq: 220 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 327 v=0 o=root 455117568 455117568 IN IP4 207.91.156.217 s=Asterisk PBX 13.13.1 c=IN IP4 207.91.156.217 t=0 0 m=audio 16466 RTP/AVP 0 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 a=maxptime:150 a=sendrecv <------------> [Dec 9 19:35:19] DEBUG[114314][C-000006f8] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #57 [Dec 9 19:35:19] DEBUG[114314][C-000006f8] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: Newstate Privilege: call,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: _1XXXXXX. Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: DIALSTATUS Value: ANSWER [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: DIALEDPEERNAME Value: SIP/flowroute-0000006c [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: DIALEDPEERNUMBER Value: flowroute/18018309060 [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: DialEnd Privilege: call,all Channel: SIP/100-0000006b ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 DestChannel: SIP/flowroute-0000006c DestChannelState: 6 DestChannelStateDesc: Up DestCallerIDNum: _1XXXXXX. DestCallerIDName: DestConnectedLineNum: 9726001152 DestConnectedLineName: Luke Apartment DestLanguage: en DestAccountCode: DestContext: from-trunk DestExten: DestPriority: 1 DestUniqueid: 1481312110.218 DestLinkedid: 1481312110.217 DialStatus: ANSWER [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge_native_rtp.c: Bridge '3db015c8-89fc-453b-9a09-4dc61cd445a6' can not use native RTP bridge as two channels are required [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Chose bridge technology simple_bridge [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: calling simple_bridge technology constructor [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: calling simple_bridge technology start [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: Newstate Privilege: call,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: BridgeCreate Privilege: call,all BridgeUniqueid: 3db015c8-89fc-453b-9a09-4dc61cd445a6 BridgeType: basic BridgeTechnology: simple_bridge BridgeCreator: BridgeName: BridgeNumChannels: 0 BridgeVideoSourceMode: none [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Session timer started: 40 - 1453900646-5060-25@BJC.BGI.B.G 900000ms [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge_channel.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: 0x7f4ecc0455d0(SIP/flowroute-0000006c) is joining [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge_channel.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: pushing 0x7f4ecc0455d0(SIP/flowroute-0000006c) [Dec 9 19:35:19] VERBOSE[114372][C-000006f8] bridge_channel.c: Channel SIP/flowroute-0000006c joined 'simple_bridge' basic-bridge <3db015c8-89fc-453b-9a09-4dc61cd445a6> [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge_native_rtp.c: Bridge '3db015c8-89fc-453b-9a09-4dc61cd445a6' can not use native RTP bridge as two channels are required [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge.c: Bridge technology softmix does not have any capabilities we want. [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge.c: Chose bridge technology simple_bridge [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6 is already using the new technology. [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: 0x7f4ecc0455d0(SIP/flowroute-0000006c) is joining simple_bridge technology [Dec 9 19:35:19] DEBUG[114372][C-000006f8] res_rtp_asterisk.c: Changing ssrc from 1067572312 to 706815009 due to a source change [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Extension: Application: AppDial AppData: (Outgoing Line) [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge_channel.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: 0x7f4ecc027ad0(SIP/100-0000006b) is joining [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: BridgeEnter Privilege: call,all BridgeUniqueid: 3db015c8-89fc-453b-9a09-4dc61cd445a6 BridgeType: basic BridgeTechnology: simple_bridge BridgeCreator: BridgeName: BridgeNumChannels: 1 BridgeVideoSourceMode: none Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge_channel.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: pushing 0x7f4ecc027ad0(SIP/100-0000006b) [Dec 9 19:35:19] VERBOSE[114314][C-000006f8] bridge_channel.c: Channel SIP/100-0000006b joined 'simple_bridge' basic-bridge <3db015c8-89fc-453b-9a09-4dc61cd445a6> [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Dec 9 19:35:19] DEBUG[85150] cdr.c: Finalized CDR for SIP/flowroute-0000006c - start 1481312110.446108 answer 1481312119.177997 end 1481312119.178330 dispo ANSWERED [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge technology softmix does not have any capabilities we want. [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Chose bridge technology native_rtp [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: calling native_rtp technology constructor [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: moving 0x7f4ecc0455d0(SIP/flowroute-0000006c) to dummy bridge temporarily [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: BridgeEnter Privilege: call,all BridgeUniqueid: 3db015c8-89fc-453b-9a09-4dc61cd445a6 BridgeType: basic BridgeTechnology: simple_bridge BridgeCreator: BridgeName: BridgeNumChannels: 2 BridgeVideoSourceMode: none Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: 0x7f4ecc0455d0(SIP/flowroute-0000006c) is leaving simple_bridge technology (dummy) [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: calling simple_bridge technology stop [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: 0x7f4ecc027ad0(SIP/100-0000006b) is joining native_rtp technology [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: 0x7f4ecc0455d0(SIP/flowroute-0000006c) is joining native_rtp technology [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: calling native_rtp technology start [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: calling simple_bridge technology destructor [Dec 9 19:35:19] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Changing ssrc from 1502779067 to 928975833 due to a source change [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge technology softmix does not have any capabilities we want. [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Chose bridge technology native_rtp [Dec 9 19:35:19] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6 is already using the new technology. [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: BRIDGEPEER Value: SIP/flowroute-0000006c [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: BRIDGEPVTCALLID Value: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: BRIDGEPEER Value: SIP/100-0000006b [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: BRIDGEPVTCALLID Value: 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge.c: Bridge technology softmix does not have any capabilities we want. [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge.c: Chose bridge technology native_rtp [Dec 9 19:35:19] DEBUG[114372][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6 is already using the new technology. [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: BRIDGEPEER Value: SIP/flowroute-0000006c [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: BRIDGEPVTCALLID Value: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: BRIDGEPEER Value: SIP/100-0000006b [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: BRIDGEPVTCALLID Value: 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: BRIDGEPEER Value: SIP/flowroute-0000006c [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: BRIDGEPVTCALLID Value: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: BRIDGEPEER Value: SIP/100-0000006b [Dec 9 19:35:19] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: BRIDGEPVTCALLID Value: 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:19] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Got RTCP report of 12 bytes [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: SIP TIMER: Rescheduling retransmission #57 (1) SIP/2.0 - 1 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #57)) [Dec 9 19:35:19] VERBOSE[85170] chan_sip.c: Retransmitting #1 (NAT) to 72.190.32.250:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK1641505246;received=72.190.32.250;rport=5060 From: "Local1" ;tag=2051421774 To: ;tag=as0deed62b Call-ID: 1453900646-5060-25@BJC.BGI.B.G CSeq: 220 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 327 v=0 o=root 455117568 455117568 IN IP4 207.91.156.217 s=Asterisk PBX 13.13.1 c=IN IP4 207.91.156.217 t=0 0 m=audio 16466 RTP/AVP 0 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 a=maxptime:150 a=sendrecv --- [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:19] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> ACK sip:8018309060@207.91.156.217:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK1942679835;rport From: "Local1" ;tag=2051421774 To: ;tag=as0deed62b Call-ID: 1453900646-5060-25@BJC.BGI.B.G CSeq: 220 ACK Contact: Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 0 [ 46]: ACK sip:8018309060@207.91.156.217:5060 SIP/2.0 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK1942679835;rport [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "Local1" ;tag=2051421774 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 3 [ 57]: To: ;tag=as0deed62b [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 4 [ 39]: Call-ID: 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 5 [ 13]: CSeq: 220 ACK [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 6 [ 35]: Contact: [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 8 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 10 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Dec 9 19:35:19] VERBOSE[85170] chan_sip.c: --- (12 headers 0 lines) --- [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: = Looking for Call ID: 1453900646-5060-25@BJC.BGI.B.G (Checking From) --From tag 2051421774 --To-tag as0deed62b [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #57 [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: Stopping retransmission on '1453900646-5060-25@BJC.BGI.B.G' of Response 220: Match Found [Dec 9 19:35:19] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> ACK sip:8018309060@207.91.156.217:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK1942679835;rport From: "Local1" ;tag=2051421774 To: ;tag=as0deed62b Call-ID: 1453900646-5060-25@BJC.BGI.B.G CSeq: 220 ACK Contact: Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 0 [ 46]: ACK sip:8018309060@207.91.156.217:5060 SIP/2.0 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK1942679835;rport [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "Local1" ;tag=2051421774 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 3 [ 57]: To: ;tag=as0deed62b [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 4 [ 39]: Call-ID: 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 5 [ 13]: CSeq: 220 ACK [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 6 [ 35]: Contact: [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 8 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 10 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Dec 9 19:35:19] VERBOSE[85170] chan_sip.c: --- (12 headers 0 lines) --- [Dec 9 19:35:19] DEBUG[85170] chan_sip.c: = Looking for Call ID: 1453900646-5060-25@BJC.BGI.B.G (Checking From) --From tag 2051421774 --To-tag as0deed62b [Dec 9 19:35:19] DEBUG[85170][C-000006f8] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Dec 9 19:35:20] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> <-------------> [Dec 9 19:35:23] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:16467' into... [Dec 9 19:35:23] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '16467'. [Dec 9 19:35:23] DEBUG[85207] netsock2.c: Splitting '72.190.32.250:5007' into... [Dec 9 19:35:23] DEBUG[85207] netsock2.c: ...host '72.190.32.250' and port '5007'. [Dec 9 19:35:23] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPSent Privilege: reporting,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 To: 72.190.32.250:5007 From: 207.91.156.217:16467 SSRC: 0x375f0bd9 PT: 200(SR) ReportCount: 1 SentNTP: 1481312123.3839533056 SentRTP: 41680 SentPackets: 413 SentOctets: 80080 Report0SourceSSRC: 0x501ee4db Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 38446 Report0SequenceNumberCycles: 0 Report0IAJitter: 34 Report0LSR: 2247493877 Report0DLSR: 4.9910 [Dec 9 19:35:23] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:16067' into... [Dec 9 19:35:23] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '16067'. [Dec 9 19:35:23] DEBUG[85207] netsock2.c: Splitting '74.120.93.26:39733' into... [Dec 9 19:35:23] DEBUG[85207] netsock2.c: ...host '74.120.93.26' and port '39733'. [Dec 9 19:35:23] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPSent Privilege: reporting,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 To: 74.120.93.26:39733 From: 207.91.156.217:16067 SSRC: 0x2a212421 PT: 200(SR) ReportCount: 1 SentNTP: 1481312123.3839823872 SentRTP: 58936000 SentPackets: 325 SentOctets: 78000 Report0SourceSSRC: 0xe409d998 Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 277 Report0SequenceNumberCycles: 0 Report0IAJitter: 0 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:35:24] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Dec 9 19:35:24] DEBUG[85207] netsock2.c: Splitting '72.190.32.250:5007' into... [Dec 9 19:35:24] DEBUG[85207] netsock2.c: ...host '72.190.32.250' and port '5007'. [Dec 9 19:35:24] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:16467' into... [Dec 9 19:35:24] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '16467'. [Dec 9 19:35:24] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPReceived Privilege: reporting,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 To: 207.91.156.217:16467 From: 72.190.32.250:5007 RTT: 0.0000 SSRC: 0x48ef13c0 PT: 200(SR) ReportCount: 1 SentNTP: 1481312124.2476998656 SentRTP: 58977040 SentPackets: 167 SentOctets: 40080 Report0SourceSSRC: 0x0d410dac Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 545 Report0SequenceNumberCycles: 0 Report0IAJitter: 14 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 18c6f7b8306b688e748f292c03da57f4@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:25] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: SIP call-id changed from '18c6f7b8306b688e748f292c03da57f4@207.91.156.217:5060' to '6afb6aee60b6748a4611302039d9fcb6@207.91.156.217:5060' [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 6afb6aee60b6748a4611302039d9fcb6@207.91.156.217:5060 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 0 [ 44]: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK0449a809;rport [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as4a138925 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 4 [ 34]: To: [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 6afb6aee60b6748a4611302039d9fcb6@207.91.156.217:5060 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:35:25 GMT [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:35:25] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 97.77.119.234:47976: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK0449a809;rport Max-Forwards: 70 From: "asterisk" ;tag=as4a138925 To: Contact: Call-ID: 6afb6aee60b6748a4611302039d9fcb6@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:35:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #35 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:35:25] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK0449a809;rport=5060 From: "asterisk" ;tag=as4a138925 To: ;tag=12622694 Call-ID: 6afb6aee60b6748a4611302039d9fcb6@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK0449a809;rport=5060 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as4a138925 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 3 [ 47]: To: ;tag=12622694 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 6afb6aee60b6748a4611302039d9fcb6@207.91.156.217:5060 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:35:25] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: = Looking for Call ID: 6afb6aee60b6748a4611302039d9fcb6@207.91.156.217:5060 (Checking To) --From tag as4a138925 --To-tag 12622694 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #35 [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Stopping retransmission on '6afb6aee60b6748a4611302039d9fcb6@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:25] DEBUG[85170] chan_sip.c: Destroying SIP dialog 6afb6aee60b6748a4611302039d9fcb6@207.91.156.217:5060 [Dec 9 19:35:25] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '6afb6aee60b6748a4611302039d9fcb6@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> REGISTER sip:testbed2.primevox.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.247:47976;branch=z9hG4bK1300485936;rport From: ;tag=1885999639 To: Call-ID: 256831031-54249-15@BJC.BGI.C.CEH CSeq: 33490 REGISTER Contact: ;reg-id=4;+sip.instance="" Authorization: Digest username="1033", realm="asterisk", nonce="4268b1f1", uri="sip:testbed2.primevox.net", response="9db7e6897d2e7f49da69543e4b5b3b80", algorithm=MD5 Max-Forwards: 70 User-Agent: Grandstream GXP2140 1.0.7.97 Supported: path Expires: 600 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 0 [ 42]: REGISTER sip:testbed2.primevox.net SIP/2.0 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.2.247:47976;branch=z9hG4bK1300485936;rport [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 2 [ 53]: From: ;tag=1885999639 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 3 [ 36]: To: [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 4 [ 41]: Call-ID: 256831031-54249-15@BJC.BGI.C.CEH [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 5 [ 20]: CSeq: 33490 REGISTER [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 6 [112]: Contact: ;reg-id=4;+sip.instance="" [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 7 [166]: Authorization: Digest username="1033", realm="asterisk", nonce="4268b1f1", uri="sip:testbed2.primevox.net", response="9db7e6897d2e7f49da69543e4b5b3b80", algorithm=MD5 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 10 [ 15]: Supported: path [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 11 [ 12]: Expires: 600 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 12 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: --- (14 headers 0 lines) --- [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: = Looking for Call ID: 256831031-54249-15@BJC.BGI.C.CEH (Checking From) --From tag 1885999639 --To-tag [Dec 9 19:35:26] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:26] DEBUG[85170] netsock2.c: Splitting '192.168.2.247:47976' into... [Dec 9 19:35:26] DEBUG[85170] netsock2.c: ...host '192.168.2.247' and port '47976'. [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: NAT detected for 192.168.2.247 / 97.77.119.234 [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: Sending to 97.77.119.234:47976 (NAT) [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 256831031-54249-15@BJC.BGI.C.CEH - REGISTER (No RTP) [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Initializing initreq for method REGISTER - callid 256831031-54249-15@BJC.BGI.C.CEH [Dec 9 19:35:26] DEBUG[85170] netsock2.c: Splitting '192.168.2.247:47976' into... [Dec 9 19:35:26] DEBUG[85170] netsock2.c: ...host '192.168.2.247' and port '47976'. [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: NAT detected for 192.168.2.247 / 97.77.119.234 [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: Sending to 97.77.119.234:47976 (NAT) [Dec 9 19:35:26] DEBUG[85170] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:35:26] DEBUG[85170] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: <--- Transmitting (NAT) to 97.77.119.234:47976 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.247:47976;branch=z9hG4bK1300485936;received=97.77.119.234;rport=47976 From: ;tag=1885999639 To: ;tag=as7cb5c9bc Call-ID: 256831031-54249-15@BJC.BGI.C.CEH CSeq: 33490 REGISTER Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5511ad9c" Content-Length: 0 <------------> [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: Scheduling destruction of SIP dialog '256831031-54249-15@BJC.BGI.C.CEH' in 32000 ms (Method: REGISTER) [Dec 9 19:35:26] DEBUG[94779] manager.c: Examining AMI event: Event: ChallengeSent Privilege: security,all EventTV: 2016-12-09T19:35:26.190+0000 Severity: Informational Service: SIP EventVersion: 1 AccountID: 1033 SessionID: 0x7f4e800055d0 LocalAddress: IPV4/UDP/207.91.156.217/5060 RemoteAddress: IPV4/UDP/97.77.119.234/47976 Challenge: 5511ad9c [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> REGISTER sip:testbed2.primevox.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.247:47976;branch=z9hG4bK247705971;rport From: ;tag=1885999639 To: Call-ID: 256831031-54249-15@BJC.BGI.C.CEH CSeq: 33491 REGISTER Contact: ;reg-id=4;+sip.instance="" Authorization: Digest username="1033", realm="asterisk", nonce="5511ad9c", uri="sip:testbed2.primevox.net", response="cb85a433954f7ba51415da561450eea1", algorithm=MD5 Max-Forwards: 70 User-Agent: Grandstream GXP2140 1.0.7.97 Supported: path Expires: 600 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 0 [ 42]: REGISTER sip:testbed2.primevox.net SIP/2.0 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.2.247:47976;branch=z9hG4bK247705971;rport [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 2 [ 53]: From: ;tag=1885999639 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 3 [ 36]: To: [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 4 [ 41]: Call-ID: 256831031-54249-15@BJC.BGI.C.CEH [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 5 [ 20]: CSeq: 33491 REGISTER [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 6 [112]: Contact: ;reg-id=4;+sip.instance="" [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 7 [166]: Authorization: Digest username="1033", realm="asterisk", nonce="5511ad9c", uri="sip:testbed2.primevox.net", response="cb85a433954f7ba51415da561450eea1", algorithm=MD5 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 10 [ 15]: Supported: path [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 11 [ 12]: Expires: 600 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 12 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: --- (14 headers 0 lines) --- [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: = Looking for Call ID: 256831031-54249-15@BJC.BGI.C.CEH (Checking From) --From tag 1885999639 --To-tag [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Initializing initreq for method REGISTER - callid 256831031-54249-15@BJC.BGI.C.CEH [Dec 9 19:35:26] DEBUG[85170] netsock2.c: Splitting '192.168.2.247:47976' into... [Dec 9 19:35:26] DEBUG[85170] netsock2.c: ...host '192.168.2.247' and port '47976'. [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: NAT detected for 192.168.2.247 / 97.77.119.234 [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: Sending to 97.77.119.234:47976 (NAT) [Dec 9 19:35:26] DEBUG[85170] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:35:26] DEBUG[85170] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: build_path: do not use Path headers [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 7384df496042028f326d710d4f061f22@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:26] DEBUG[94779] manager.c: Examining AMI event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/1033 PeerStatus: Registered Address: 97.77.119.234:47976 [Dec 9 19:35:26] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: SIP call-id changed from '7384df496042028f326d710d4f061f22@207.91.156.217:5060' to '4c90992f5bfbb3630b9acd642b838aff@207.91.156.217:5060' [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 4c90992f5bfbb3630b9acd642b838aff@207.91.156.217:5060 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 0 [ 44]: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3e3b839c;rport [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as64bd515c [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 4 [ 34]: To: [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 4c90992f5bfbb3630b9acd642b838aff@207.91.156.217:5060 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:35:26 GMT [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 97.77.119.234:47976: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3e3b839c;rport Max-Forwards: 70 From: "asterisk" ;tag=as64bd515c To: Contact: Call-ID: 4c90992f5bfbb3630b9acd642b838aff@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:35:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #30 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: <--- Transmitting (NAT) to 97.77.119.234:47976 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.247:47976;branch=z9hG4bK247705971;received=97.77.119.234;rport=47976 From: ;tag=1885999639 To: ;tag=as7cb5c9bc Call-ID: 256831031-54249-15@BJC.BGI.C.CEH CSeq: 33491 REGISTER Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: ;expires=600 Date: Fri, 09 Dec 2016 19:35:26 GMT Content-Length: 0 <------------> [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 0709ce813a04ebd455e3078e458ec8ec@207.91.156.217:5060 - NOTIFY (No RTP) [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:26] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: SIP call-id changed from '0709ce813a04ebd455e3078e458ec8ec@207.91.156.217:5060' to '07990d096efc8e9a1e2ce7645d9debc3@207.91.156.217:5060' [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: Scheduling destruction of SIP dialog '07990d096efc8e9a1e2ce7645d9debc3@207.91.156.217:5060' in 6400 ms (Method: NOTIFY) [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Initializing initreq for method NOTIFY - callid 07990d096efc8e9a1e2ce7645d9debc3@207.91.156.217:5060 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 0 [ 43]: NOTIFY sip:1033@192.168.2.247:47976 SIP/2.0 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK387abce1;rport [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 3 [ 50]: From: ;tag=as090573b0 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 4 [ 34]: To: [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 07990d096efc8e9a1e2ce7645d9debc3@207.91.156.217:5060 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 7 [ 16]: CSeq: 102 NOTIFY [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 9 [ 22]: Event: message-summary [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 10 [ 48]: Content-Type: application/simple-message-summary [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 97.77.119.234:47976: NOTIFY sip:1033@192.168.2.247:47976 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK387abce1;rport Max-Forwards: 70 From: ;tag=as090573b0 To: Contact: Call-ID: 07990d096efc8e9a1e2ce7645d9debc3@207.91.156.217:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 13.13.1 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 94 Messages-Waiting: no Message-Account: sip:asterisk@207.91.156.217 Voice-Message: 0/0 (0/0) --- [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #34 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: Scheduling destruction of SIP dialog '256831031-54249-15@BJC.BGI.C.CEH' in 32000 ms (Method: REGISTER) [Dec 9 19:35:26] DEBUG[94779] manager.c: Examining AMI event: Event: SuccessfulAuth Privilege: security,all EventTV: 2016-12-09T19:35:26.212+0000 Severity: Informational Service: SIP EventVersion: 1 AccountID: 1033 SessionID: 0x7f4e800055d0 LocalAddress: IPV4/UDP/207.91.156.217/5060 RemoteAddress: IPV4/UDP/97.77.119.234/47976 UsingPassword: 1 [Dec 9 19:35:26] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - 1033 [Dec 9 19:35:26] DEBUG[85149] chan_sip.c: Checking device state for peer 1033 [Dec 9 19:35:26] DEBUG[85149] devicestate.c: Changing state for SIP/1033 - state 1 (Not in use) [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3e3b839c;rport=5060 From: "asterisk" ;tag=as64bd515c To: ;tag=566704741 Call-ID: 4c90992f5bfbb3630b9acd642b838aff@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3e3b839c;rport=5060 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as64bd515c [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 3 [ 48]: To: ;tag=566704741 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 4c90992f5bfbb3630b9acd642b838aff@207.91.156.217:5060 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: = Looking for Call ID: 4c90992f5bfbb3630b9acd642b838aff@207.91.156.217:5060 (Checking To) --From tag as64bd515c --To-tag 566704741 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #30 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Stopping retransmission on '4c90992f5bfbb3630b9acd642b838aff@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Destroying SIP dialog 4c90992f5bfbb3630b9acd642b838aff@207.91.156.217:5060 [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '4c90992f5bfbb3630b9acd642b838aff@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK387abce1;rport=5060 From: ;tag=as090573b0 To: ;tag=322496501 Call-ID: 07990d096efc8e9a1e2ce7645d9debc3@207.91.156.217:5060 CSeq: 102 NOTIFY Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK387abce1;rport=5060 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 2 [ 50]: From: ;tag=as090573b0 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 3 [ 48]: To: ;tag=322496501 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 07990d096efc8e9a1e2ce7645d9debc3@207.91.156.217:5060 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 102 NOTIFY [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: = Looking for Call ID: 07990d096efc8e9a1e2ce7645d9debc3@207.91.156.217:5060 (Checking To) --From tag as090573b0 --To-tag 322496501 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #34 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Stopping retransmission on '07990d096efc8e9a1e2ce7645d9debc3@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Got 200 accepted on NOTIFY 07990d096efc8e9a1e2ce7645d9debc3@207.91.156.217:5060 [Dec 9 19:35:26] DEBUG[85170] chan_sip.c: Destroying SIP dialog 07990d096efc8e9a1e2ce7645d9debc3@207.91.156.217:5060 [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '07990d096efc8e9a1e2ce7645d9debc3@207.91.156.217:5060' Method: NOTIFY [Dec 9 19:35:26] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> <-------------> [Dec 9 19:35:28] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:16467' into... [Dec 9 19:35:28] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '16467'. [Dec 9 19:35:28] DEBUG[85207] netsock2.c: Splitting '72.190.32.250:5007' into... [Dec 9 19:35:28] DEBUG[85207] netsock2.c: ...host '72.190.32.250' and port '5007'. [Dec 9 19:35:28] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPSent Privilege: reporting,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 To: 72.190.32.250:5007 From: 207.91.156.217:16467 SSRC: 0x375f0bd9 PT: 200(SR) ReportCount: 1 SentNTP: 1481312128.3838087168 SentRTP: 41680 SentPackets: 663 SentOctets: 120080 Report0SourceSSRC: 0x501ee4db Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 38446 Report0SequenceNumberCycles: 0 Report0IAJitter: 34 Report0LSR: 2247886831 Report0DLSR: 4.3730 [Dec 9 19:35:28] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:16067' into... [Dec 9 19:35:28] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '16067'. [Dec 9 19:35:28] DEBUG[85207] netsock2.c: Splitting '74.120.93.26:39733' into... [Dec 9 19:35:28] DEBUG[85207] netsock2.c: ...host '74.120.93.26' and port '39733'. [Dec 9 19:35:28] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPSent Privilege: reporting,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 To: 74.120.93.26:39733 From: 207.91.156.217:16067 SSRC: 0x2a212421 PT: 200(SR) ReportCount: 1 SentNTP: 1481312128.3838472192 SentRTP: 58936000 SentPackets: 491 SentOctets: 117840 Report0SourceSSRC: 0xe409d998 Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 277 Report0SequenceNumberCycles: 0 Report0IAJitter: 0 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:35:29] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Dec 9 19:35:29] DEBUG[85207] netsock2.c: Splitting '72.190.32.250:5007' into... [Dec 9 19:35:29] DEBUG[85207] netsock2.c: ...host '72.190.32.250' and port '5007'. [Dec 9 19:35:29] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:16467' into... [Dec 9 19:35:29] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '16467'. [Dec 9 19:35:29] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPReceived Privilege: reporting,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 To: 207.91.156.217:16467 From: 72.190.32.250:5007 RTT: 0.0000 SSRC: 0x48ef13c0 PT: 200(SR) ReportCount: 1 SentNTP: 1481312129.484675584 SentRTP: 59017120 SentPackets: 334 SentOctets: 80160 Report0SourceSSRC: 0x0d410dac Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 796 Report0SequenceNumberCycles: 0 Report0IAJitter: 15 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:35:30] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> BYE sip:8018309060@207.91.156.217:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK202510564;rport From: "Local1" ;tag=2051421774 To: ;tag=as0deed62b Call-ID: 1453900646-5060-25@BJC.BGI.B.G CSeq: 221 BYE Contact: Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 0 [ 46]: BYE sip:8018309060@207.91.156.217:5060 SIP/2.0 [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK202510564;rport [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "Local1" ;tag=2051421774 [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 3 [ 57]: To: ;tag=as0deed62b [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 4 [ 39]: Call-ID: 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 5 [ 13]: CSeq: 221 BYE [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 6 [ 35]: Contact: [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 8 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 10 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Dec 9 19:35:30] VERBOSE[85170] chan_sip.c: --- (12 headers 0 lines) --- [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: = Looking for Call ID: 1453900646-5060-25@BJC.BGI.B.G (Checking From) --From tag 2051421774 --To-tag as0deed62b [Dec 9 19:35:30] DEBUG[85170][C-000006f8] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Dec 9 19:35:30] DEBUG[85170][C-000006f8] chan_sip.c: Initializing initreq for method BYE - callid 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:30] DEBUG[85170][C-000006f8] netsock2.c: Splitting '192.168.1.6:5060' into... [Dec 9 19:35:30] DEBUG[85170][C-000006f8] netsock2.c: ...host '192.168.1.6' and port '5060'. [Dec 9 19:35:30] DEBUG[85170][C-000006f8] chan_sip.c: NAT detected for 192.168.1.6 / 72.190.32.250 [Dec 9 19:35:30] VERBOSE[85170][C-000006f8] chan_sip.c: Sending to 72.190.32.250:5060 (NAT) [Dec 9 19:35:30] DEBUG[85170][C-000006f8] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: RTPAUDIOQOS Value: ssrc=928975833;themssrc=1344201947;lp=0;rxjitter=0.000000;rxcount=536;txjitter=0.004360;txcount=729;rlp=0;rtt=0.000000 [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=928975833;themssrc=1344201947;lp=0;rxjitter=0.000000;rxcount=536;txjitter=0.004360;txcount=729;rlp=0;rtt=0.000000 [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; [Dec 9 19:35:30] DEBUG[85170][C-000006f8] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4e80014da0' [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; [Dec 9 19:35:30] VERBOSE[85170][C-000006f8] chan_sip.c: Scheduling destruction of SIP dialog '1453900646-5060-25@BJC.BGI.B.G' in 6400 ms (Method: BYE) [Dec 9 19:35:30] DEBUG[85170][C-000006f8] chan_sip.c: Received bye, issuing owner hangup [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; [Dec 9 19:35:30] VERBOSE[85170][C-000006f8] chan_sip.c: <--- Transmitting (NAT) to 72.190.32.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK202510564;received=72.190.32.250;rport=5060 From: "Local1" ;tag=2051421774 To: ;tag=as0deed62b Call-ID: 1453900646-5060-25@BJC.BGI.B.G CSeq: 221 BYE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Dec 9 19:35:30] DEBUG[85170][C-000006f8] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Session timer stopped: 40 - 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:30] DEBUG[114314][C-000006f8] bridge_channel.c: Setting 0x7f4ecc027ad0(SIP/100-0000006b) state from:0 to:1 [Dec 9 19:35:30] DEBUG[114314][C-000006f8] bridge_channel.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: pulling 0x7f4ecc027ad0(SIP/100-0000006b) [Dec 9 19:35:30] VERBOSE[114314][C-000006f8] bridge_channel.c: Channel SIP/100-0000006b left 'native_rtp' basic-bridge <3db015c8-89fc-453b-9a09-4dc61cd445a6> [Dec 9 19:35:30] DEBUG[114314][C-000006f8] bridge_channel.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: 0x7f4ecc027ad0(SIP/100-0000006b) is leaving native_rtp technology [Dec 9 19:35:30] DEBUG[114314][C-000006f8] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/100-0000006b' and 'SIP/flowroute-0000006c' - media will flow through Asterisk core [Dec 9 19:35:30] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: dissolving bridge with cause 16(Normal Clearing) [Dec 9 19:35:30] DEBUG[114314][C-000006f8] bridge_channel.c: Setting 0x7f4ecc0455d0(SIP/flowroute-0000006c) state from:0 to:2 [Dec 9 19:35:30] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: queueing action type:13 sub:1001 [Dec 9 19:35:30] DEBUG[114314][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6 is dissolved, not performing smart bridge operation. [Dec 9 19:35:30] DEBUG[114372][C-000006f8] bridge_channel.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: pulling 0x7f4ecc0455d0(SIP/flowroute-0000006c) [Dec 9 19:35:30] VERBOSE[114372][C-000006f8] bridge_channel.c: Channel SIP/flowroute-0000006c left 'native_rtp' basic-bridge <3db015c8-89fc-453b-9a09-4dc61cd445a6> [Dec 9 19:35:30] DEBUG[114372][C-000006f8] bridge_channel.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: 0x7f4ecc0455d0(SIP/flowroute-0000006c) is leaving native_rtp technology [Dec 9 19:35:30] DEBUG[114372][C-000006f8] bridge_channel.c: Channel SIP/flowroute-0000006c will survive this bridge; clearing outgoing (dialed) flag [Dec 9 19:35:30] DEBUG[114372][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6 is dissolved, not performing smart bridge operation. [Dec 9 19:35:30] DEBUG[114372][C-000006f8] res_rtp_asterisk.c: Changing ssrc from 706815009 to 1981595167 due to a source change [Dec 9 19:35:30] DEBUG[114314][C-000006f8] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Dec 9 19:35:30] DEBUG[114314][C-000006f8] pbx.c: Spawn extension (dial-out-100,_1XXXXXX.,9) exited non-zero on 'SIP/100-0000006b' [Dec 9 19:35:30] VERBOSE[114314][C-000006f8] pbx.c: Spawn extension (dial-out-100, _1XXXXXX., 9) exited non-zero on 'SIP/100-0000006b' [Dec 9 19:35:30] DEBUG[114314][C-000006f8] channel.c: Soft-Hanging (0x10) up channel 'SIP/100-0000006b' [Dec 9 19:35:30] DEBUG[114314][C-000006f8] channel.c: Hanging up channel 'SIP/100-0000006b' [Dec 9 19:35:30] DEBUG[114314][C-000006f8] chan_sip.c: Hangup call SIP/100-0000006b, SIP callid 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:30] DEBUG[114314][C-000006f8] chan_sip.c: update_call_counter(100) - decrement call limit counter on hangup [Dec 9 19:35:30] DEBUG[114314][C-000006f8] chan_sip.c: Updating call counter for incoming call [Dec 9 19:35:30] DEBUG[114314][C-000006f8] chan_sip.c: Call from peer '100' removed from call limit 2147483647 [Dec 9 19:35:30] DEBUG[85148][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: actually destroying basic bridge, nobody wants it anymore [Dec 9 19:35:30] DEBUG[114314][C-000006f8] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4e80014da0' [Dec 9 19:35:30] DEBUG[85148][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: calling basic bridge destructor [Dec 9 19:35:30] DEBUG[85148][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: calling native_rtp technology stop [Dec 9 19:35:30] DEBUG[85148][C-000006f8] bridge.c: Bridge 3db015c8-89fc-453b-9a09-4dc61cd445a6: calling native_rtp technology destructor [Dec 9 19:35:30] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Dec 9 19:35:30] DEBUG[85149] chan_sip.c: Checking device state for peer 100 [Dec 9 19:35:30] DEBUG[85149] devicestate.c: Changing state for SIP/100 - state 1 (Not in use) [Dec 9 19:35:30] DEBUG[114372][C-000006f8] channel.c: Hanging up channel 'SIP/flowroute-0000006c' [Dec 9 19:35:30] DEBUG[114372][C-000006f8] chan_sip.c: Hangup call SIP/flowroute-0000006c, SIP callid 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:30] DEBUG[114372][C-000006f8] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4ecc03fd90' [Dec 9 19:35:30] VERBOSE[114372][C-000006f8] chan_sip.c: Scheduling destruction of SIP dialog '076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060' in 6400 ms (Method: INVITE) [Dec 9 19:35:30] DEBUG[85210] app_queue.c: Device 'SIP/100' changed to state '1' (Not in use) [Dec 9 19:35:30] DEBUG[85210] app_queue.c: Device 'Queue:Technical Support_avail' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 9 19:35:30] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Dec 9 19:35:30] DEBUG[85149] chan_sip.c: Checking device state for peer 100 [Dec 9 19:35:30] DEBUG[85149] devicestate.c: Changing state for SIP/100 - state 1 (Not in use) [Dec 9 19:35:30] DEBUG[85150] cdr.c: Finalized CDR for SIP/100-0000006b - start 1481312110.438662 answer 1481312119.178010 end 1481312130.267000 dispo ANSWERED [Dec 9 19:35:30] DEBUG[85151] app_queue.c: Extension '100@from-internal' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: from-internal Hint: SIP/100 Status: 0 StatusText: Idle [Dec 9 19:35:30] VERBOSE[114372][C-000006f8] chan_sip.c: Reliably Transmitting (NAT) to 216.115.69.144:5060: BYE sip:+18018309060@74.120.93.24:5060 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3fad3c18;rport Route: , Max-Forwards: 70 From: "Luke Apartment" ;tag=as76bc5363 To: ;tag=gK00db684d Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 13.13.1 Proxy-Authorization: Digest username="23616016", realm="sip.flowroute.com", algorithm=MD5, uri="sip:+18018309060@74.120.93.24:5060", nonce="WEsImlhLB25P25UP37SLPSvA1iDH2/AT", response="9678d95a9277dcfdd6089b70cebad47a", qop=auth, cnonce="06f39241", nc=00000002 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Dec 9 19:35:30] DEBUG[114372][C-000006f8] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #23 [Dec 9 19:35:30] DEBUG[114372][C-000006f8] chan_sip.c: Trying to put 'BYE sip:+18' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(uniqueid) result is '1481312110.217' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(start) result is '2016-12-09 19:35:10' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(end) result is '2016-12-09 19:35:30' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(disposition) result is 'ANSWERED' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(cdr_direction) result is '2' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(cdr_application) result is 'Outbound Dial' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(cdr_inbound_did) result is '' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(cdr_inbound_from_phonenumber) result is '' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(cdr_inbound_from_callerid) result is '' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(cdr_inbound_to_extension) result is '' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(cdr_outbound_did) result is '9726001152' [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Session timer stopped: 62 - 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(cdr_outbound_dialed_phonenumber) result is '18018309060' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(cdr_outbound_from_extension) result is '100' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(cdr_internal_from_extension) result is '' [Dec 9 19:35:30] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - flowroute [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(cdr_internal_to_extension) result is '' [Dec 9 19:35:30] DEBUG[85150] pbx_variables.c: Function CDR(cdr_recording_path) result is '' [Dec 9 19:35:30] DEBUG[85149] chan_sip.c: Checking device state for peer flowroute [Dec 9 19:35:30] DEBUG[85149] devicestate.c: Changing state for SIP/flowroute - state 1 (Not in use) [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: Cdr Privilege: cdr,all AccountCode: Source: 9726001152 Destination: _1XXXXXX. DestinationContext: dial-out-100 CallerID: "Luke Apartment" <9726001152> Channel: SIP/100-0000006b DestinationChannel: SIP/flowroute-0000006c LastApplication: Dial LastData: SIP/flowroute/18018309060 StartTime: 2016-12-09 19:35:10 AnswerTime: 2016-12-09 19:35:19 EndTime: 2016-12-09 19:35:30 Duration: 19 BillableSeconds: 11 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1481312110.217 UserField: unique_id: 1481312110.217 tstamp_start: 2016-12-09 19:35:10 tstamp_end: 2016-12-09 19:35:30 answer_status: ANSWERED direction: 2 application: Outbound Dial inbound_did: inbound_from_phonenumber: inbound_from_callerid: inbound_to_extension: outbound_did: 9726001152 outbound_dialed_phonenumber: 18018309060 outbound_from_extension: 100 internal_from_extension: internal_to_extension: recording_path: [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOS Value: ssrc=706815009;themssrc=3825850776;lp=0;rxjitter=0.000000;rxcount=558;txjitter=0.000027;txcount=536;rlp=0;rtt=0.000000 [Dec 9 19:35:30] DEBUG[85150] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=706815009;themssrc=3825850776;lp=0;rxjitter=0.000000;rxcount=558;txjitter=0.000027;txcount=536;rlp=0;rtt=0.000000 [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; [Dec 9 19:35:30] DEBUG[85150] res_config_sqlite.c: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('"Luke Apartment" <9726001152>','9726001152','_1XXXXXX.','dial-out-100','SIP/100-0000006b','SIP/flowroute-0000006c','Dial','SIP/flowroute/18018309060','2016-12-09 19:35:10','2016-12-09 19:35:19','2016-12-09 19:35:30','19','11','ANSWERED','DOCUMENTATION','1481312110.217') [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: HangupRequest Privilege: call,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: BRIDGEPEER Value: [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: BRIDGEPVTCALLID Value: [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: BRIDGEPEER Value: [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: BRIDGEPVTCALLID Value: [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: BridgeLeave Privilege: call,all BridgeUniqueid: 3db015c8-89fc-453b-9a09-4dc61cd445a6 BridgeType: basic BridgeTechnology: native_rtp BridgeCreator: BridgeName: BridgeNumChannels: 1 BridgeVideoSourceMode: none Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: ANSWEREDTIME Value: 11 [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: BridgeLeave Privilege: call,all BridgeUniqueid: 3db015c8-89fc-453b-9a09-4dc61cd445a6 BridgeType: basic BridgeTechnology: native_rtp BridgeCreator: BridgeName: BridgeNumChannels: 0 BridgeVideoSourceMode: none Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: DIALEDTIME Value: 19 [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Variable: DIALSTATUS Value: ANSWER [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: SoftHangupRequest Privilege: call,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Cause: 16 [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: BridgeDestroy Privilege: call,all BridgeUniqueid: 3db015c8-89fc-453b-9a09-4dc61cd445a6 BridgeType: basic BridgeTechnology: native_rtp BridgeCreator: BridgeName: BridgeNumChannels: 0 BridgeVideoSourceMode: none [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: Hangup Privilege: call,all Channel: SIP/100-0000006b ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312110.217 Linkedid: 1481312110.217 Cause: 16 Cause-txt: Normal Clearing [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: DeviceStateChange Privilege: call,all Device: SIP/100 State: NOT_INUSE [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOS Value: ssrc=1981595167;themssrc=3825850776;lp=0;rxjitter=0.000000;rxcount=558;txjitter=0.000027;txcount=536;rlp=0;rtt=0.000000 [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: QueueMemberStatus Privilege: agent,all Ringinuse: 0 StateInterface: SIP/100 CallsTaken: 0 Queue: Technical Support Membership: static Status: 1 MemberName: SIP/100 Penalty: 0 Interface: SIP/100 LastCall: 0 InCall: 0 Paused: 0 PausedReason: [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: DeviceStateChange Privilege: call,all Device: Queue:Technical Support_avail State: NOT_INUSE [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Variable: RTPAUDIOQOS Value: ssrc=1981595167;themssrc=3825850776;lp=0;rxjitter=0.000000;rxcount=558;txjitter=0.000027;txcount=536;rlp=0;rtt=0.000000 [Dec 9 19:35:30] DEBUG[94779] manager.c: Examining AMI event: Event: Hangup Privilege: call,all Channel: SIP/flowroute-0000006c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312110.218 Linkedid: 1481312110.217 Cause: 16 Cause-txt: Normal Clearing [Dec 9 19:35:30] DEBUG[85150] cdr.c: CDR for SIP/flowroute-0000006c is dialed and has no Party B; discarding [Dec 9 19:35:30] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 200 OK From: "Luke Apartment" ;tag=as76bc5363 To: ;tag=gK00db684d Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3fad3c18;rport=5060 Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 CSeq: 104 BYE Record-Route: Record-Route: Content-Length: 0 <-------------> [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 1 [ 67]: From: "Luke Apartment" ;tag=as76bc5363 [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 2 [ 54]: To: ;tag=gK00db684d [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 3 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3fad3c18;rport=5060 [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 6 [ 42]: Record-Route: [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 7 [ 42]: Record-Route: [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Dec 9 19:35:30] VERBOSE[85170] chan_sip.c: --- (9 headers 0 lines) --- [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: = Looking for Call ID: 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 (Checking To) --From tag as76bc5363 --To-tag gK00db684d [Dec 9 19:35:30] DEBUG[85170][C-000006f8] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #23 [Dec 9 19:35:30] DEBUG[85170][C-000006f8] chan_sip.c: Stopping retransmission on '076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060' of Request 104: Match Found [Dec 9 19:35:30] DEBUG[85170] chan_sip.c: Destroying SIP dialog 076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060 [Dec 9 19:35:30] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '076b6e610f8be5a8696688646c4f9b85@207.91.156.217:5060' Method: INVITE [Dec 9 19:35:30] DEBUG[85170] rtp_engine.c: Destroyed RTP instance '0x7f4ecc03fd90' [Dec 9 19:35:31] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 5c3b4bbf7c77dcd523dcbcdd0289b44f@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:33] DEBUG[85170] acl.c: For destination '216.115.69.144', our source address is '207.91.156.217'. [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: SIP call-id changed from '5c3b4bbf7c77dcd523dcbcdd0289b44f@207.91.156.217:5060' to '1c8b6865313f384c2b93e51b14cbaa38@207.91.156.217:5060' [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 1c8b6865313f384c2b93e51b14cbaa38@207.91.156.217:5060 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 0 [ 37]: OPTIONS sip:sip.flowroute.com SIP/2.0 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3ca9062b;rport [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as7b4fd11b [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 4 [ 27]: To: [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 1c8b6865313f384c2b93e51b14cbaa38@207.91.156.217:5060 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:35:33 GMT [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:35:33] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 216.115.69.144:5060: OPTIONS sip:sip.flowroute.com SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3ca9062b;rport Max-Forwards: 70 From: "asterisk" ;tag=as7b4fd11b To: Contact: Call-ID: 1c8b6865313f384c2b93e51b14cbaa38@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:35:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #75 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:35:33] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3ca9062b;rport=5060 From: "asterisk" ;tag=as7b4fd11b To: ;tag=20e1698a3241cbcc6677a39fcb65a0aa.8400 Call-ID: 1c8b6865313f384c2b93e51b14cbaa38@207.91.156.217:5060 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Content-Length: 0 <-------------> [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3ca9062b;rport=5060 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as7b4fd11b [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 3 [ 69]: To: ;tag=20e1698a3241cbcc6677a39fcb65a0aa.8400 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 1c8b6865313f384c2b93e51b14cbaa38@207.91.156.217:5060 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 6 [ 11]: Accept: */* [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 7 [ 17]: Accept-Encoding: [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 9 [ 11]: Supported: [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Dec 9 19:35:33] VERBOSE[85170] chan_sip.c: --- (11 headers 0 lines) --- [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: = Looking for Call ID: 1c8b6865313f384c2b93e51b14cbaa38@207.91.156.217:5060 (Checking To) --From tag as7b4fd11b --To-tag 20e1698a3241cbcc6677a39fcb65a0aa.8400 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #75 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Stopping retransmission on '1c8b6865313f384c2b93e51b14cbaa38@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Destroying SIP dialog 1c8b6865313f384c2b93e51b14cbaa38@207.91.156.217:5060 [Dec 9 19:35:33] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '1c8b6865313f384c2b93e51b14cbaa38@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 62352b276376950014a21c6d76b3d9c0@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:33] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: SIP call-id changed from '62352b276376950014a21c6d76b3d9c0@207.91.156.217:5060' to '21f13ddc1e614c396d828bee6ecdbc6a@207.91.156.217:5060' [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 21f13ddc1e614c396d828bee6ecdbc6a@207.91.156.217:5060 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 0 [ 40]: OPTIONS sip:100@192.168.1.6:5060 SIP/2.0 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK72a589f6;rport [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as366afa7b [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 4 [ 30]: To: [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 21f13ddc1e614c396d828bee6ecdbc6a@207.91.156.217:5060 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:35:33 GMT [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:35:33] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 72.190.32.250:5060: OPTIONS sip:100@192.168.1.6:5060 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK72a589f6;rport Max-Forwards: 70 From: "asterisk" ;tag=as366afa7b To: Contact: Call-ID: 21f13ddc1e614c396d828bee6ecdbc6a@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:35:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #19 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:33] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK72a589f6;rport=5060 From: "asterisk" ;tag=as366afa7b To: ;tag=465606362 Call-ID: 21f13ddc1e614c396d828bee6ecdbc6a@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK72a589f6;rport=5060 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as366afa7b [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 3 [ 44]: To: ;tag=465606362 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 21f13ddc1e614c396d828bee6ecdbc6a@207.91.156.217:5060 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:35:33] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: = Looking for Call ID: 21f13ddc1e614c396d828bee6ecdbc6a@207.91.156.217:5060 (Checking To) --From tag as366afa7b --To-tag 465606362 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #19 [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Stopping retransmission on '21f13ddc1e614c396d828bee6ecdbc6a@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:33] DEBUG[85170] chan_sip.c: Destroying SIP dialog 21f13ddc1e614c396d828bee6ecdbc6a@207.91.156.217:5060 [Dec 9 19:35:33] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '21f13ddc1e614c396d828bee6ecdbc6a@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:35:34] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:40886 ---> <-------------> [Dec 9 19:35:36] DEBUG[85170] chan_sip.c: Auto destroying SIP dialog '1453900646-5060-25@BJC.BGI.B.G' [Dec 9 19:35:36] DEBUG[85170] chan_sip.c: Destroying SIP dialog 1453900646-5060-25@BJC.BGI.B.G [Dec 9 19:35:36] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '1453900646-5060-25@BJC.BGI.B.G' Method: BYE [Dec 9 19:35:36] DEBUG[85170] rtp_engine.c: Destroyed RTP instance '0x7f4e80014da0' [Dec 9 19:35:40] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> <-------------> [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 60103d1019284b5d1d3d16757cbda1a1@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:41] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: SIP call-id changed from '60103d1019284b5d1d3d16757cbda1a1@207.91.156.217:5060' to '0f90a694455b62bb1eb79df446434b6b@207.91.156.217:5060' [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 0f90a694455b62bb1eb79df446434b6b@207.91.156.217:5060 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 0 [ 44]: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK0733ae70;rport [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as3be5bbc7 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 4 [ 34]: To: [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 0f90a694455b62bb1eb79df446434b6b@207.91.156.217:5060 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:35:41 GMT [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:35:41] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 97.77.119.234:47976: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK0733ae70;rport Max-Forwards: 70 From: "asterisk" ;tag=as3be5bbc7 To: Contact: Call-ID: 0f90a694455b62bb1eb79df446434b6b@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:35:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #59 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:35:41] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK0733ae70;rport=5060 From: "asterisk" ;tag=as3be5bbc7 To: ;tag=1477247329 Call-ID: 0f90a694455b62bb1eb79df446434b6b@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK0733ae70;rport=5060 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as3be5bbc7 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 3 [ 49]: To: ;tag=1477247329 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 0f90a694455b62bb1eb79df446434b6b@207.91.156.217:5060 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:35:41] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: = Looking for Call ID: 0f90a694455b62bb1eb79df446434b6b@207.91.156.217:5060 (Checking To) --From tag as3be5bbc7 --To-tag 1477247329 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #59 [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Stopping retransmission on '0f90a694455b62bb1eb79df446434b6b@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:41] DEBUG[85170] chan_sip.c: Destroying SIP dialog 0f90a694455b62bb1eb79df446434b6b@207.91.156.217:5060 [Dec 9 19:35:41] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '0f90a694455b62bb1eb79df446434b6b@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:35:44] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:35:45] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> <-------------> [Dec 9 19:35:45] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> INVITE sip:8018309060@testbed2.primevox.net SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK548259106;rport From: "Local1" ;tag=1110989379 To: Call-ID: 1197569442-5060-26@BJC.BGI.B.G CSeq: 230 INVITE Contact: "Local1" Max-Forwards: 70 User-Agent: Grandstream GXP2140 1.0.7.97 Privacy: none P-Preferred-Identity: "Local1" Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 355 v=0 o=100 8001 8000 IN IP4 192.168.1.6 s=SIP Call c=IN IP4 192.168.1.6 t=0 0 m=audio 5006 RTP/AVP 125 0 97 18 9 101 a=sendrecv a=rtpmap:125 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 0 [ 51]: INVITE sip:8018309060@testbed2.primevox.net SIP/2.0 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK548259106;rport [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "Local1" ;tag=1110989379 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 3 [ 42]: To: [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 4 [ 39]: Call-ID: 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 230 INVITE [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 6 [ 44]: Contact: "Local1" [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 8 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 9 [ 13]: Privacy: none [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 10 [ 62]: P-Preferred-Identity: "Local1" [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 11 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 12 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 14 [ 47]: Accept: application/sdp, application/dtmf-relay [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 15 [ 19]: Content-Length: 355 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Header 16 [ 0]: [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 0 [ 3]: v=0 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 1 [ 34]: o=100 8001 8000 IN IP4 192.168.1.6 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 2 [ 10]: s=SIP Call [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 3 [ 20]: c=IN IP4 192.168.1.6 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 5 [ 38]: m=audio 5006 RTP/AVP 125 0 97 18 9 101 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 6 [ 10]: a=sendrecv [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 7 [ 25]: a=rtpmap:125 opus/48000/2 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 8 [ 10]: a=ptime:20 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 10 [ 21]: a=rtpmap:97 iLBC/8000 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 11 [ 17]: a=fmtp:97 mode=30 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 12 [ 21]: a=rtpmap:18 G729/8000 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 13 [ 19]: a=fmtp:18 annexb=no [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 14 [ 20]: a=rtpmap:9 G722/8000 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 15 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Body 16 [ 15]: a=fmtp:101 0-15 [Dec 9 19:35:45] VERBOSE[85170] chan_sip.c: --- (16 headers 17 lines) --- [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: = Looking for Call ID: 1197569442-5060-26@BJC.BGI.B.G (Checking From) --From tag 1110989379 --To-tag [Dec 9 19:35:45] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:45] DEBUG[85170] netsock2.c: Splitting '192.168.1.6:5060' into... [Dec 9 19:35:45] DEBUG[85170] netsock2.c: ...host '192.168.1.6' and port '5060'. [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: NAT detected for 192.168.1.6 / 72.190.32.250 [Dec 9 19:35:45] VERBOSE[85170] chan_sip.c: Sending to 72.190.32.250:5060 (NAT) [Dec 9 19:35:45] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 1197569442-5060-26@BJC.BGI.B.G - INVITE (No RTP) [Dec 9 19:35:45] DEBUG[85170][C-000006f9] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Dec 9 19:35:45] DEBUG[85170][C-000006f9] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, path, timer" [Dec 9 19:35:45] DEBUG[85170][C-000006f9] sip/reqresp_parser.c: Found SIP option: -replaces- [Dec 9 19:35:45] DEBUG[85170][C-000006f9] sip/reqresp_parser.c: Matched SIP option: replaces [Dec 9 19:35:45] DEBUG[85170][C-000006f9] sip/reqresp_parser.c: Found SIP option: -path- [Dec 9 19:35:45] DEBUG[85170][C-000006f9] sip/reqresp_parser.c: Matched SIP option: path [Dec 9 19:35:45] DEBUG[85170][C-000006f9] sip/reqresp_parser.c: Found SIP option: -timer- [Dec 9 19:35:45] DEBUG[85170][C-000006f9] sip/reqresp_parser.c: Matched SIP option: timer [Dec 9 19:35:45] DEBUG[85170][C-000006f9] netsock2.c: Splitting '192.168.1.6:5060' into... [Dec 9 19:35:45] DEBUG[85170][C-000006f9] netsock2.c: ...host '192.168.1.6' and port '5060'. [Dec 9 19:35:45] DEBUG[85170][C-000006f9] chan_sip.c: NAT detected for 192.168.1.6 / 72.190.32.250 [Dec 9 19:35:45] VERBOSE[85170][C-000006f9] chan_sip.c: Sending to 72.190.32.250:5060 (NAT) [Dec 9 19:35:45] DEBUG[85170][C-000006f9] chan_sip.c: Initializing initreq for method INVITE - callid 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:35:45] VERBOSE[85170][C-000006f9] chan_sip.c: Using INVITE request as basis request - 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:35:45] DEBUG[85170][C-000006f9] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:35:45] DEBUG[85170][C-000006f9] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:35:45] VERBOSE[85170][C-000006f9] chan_sip.c: Found peer '100' for '100' from 72.190.32.250:5060 [Dec 9 19:35:45] DEBUG[85170][C-000006f9] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f4e80002c00' [Dec 9 19:35:45] DEBUG[85170][C-000006f9] res_rtp_asterisk.c: Allocated port 14788 for RTP instance '0x7f4e80002c00' [Dec 9 19:35:45] DEBUG[85170][C-000006f9] rtp_engine.c: RTP instance '0x7f4e80002c00' is setup and ready to go [Dec 9 19:35:45] DEBUG[85170][C-000006f9] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Dec 9 19:35:45] DEBUG[85170][C-000006f9] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:35:45] DEBUG[85170][C-000006f9] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f4e80002c00' [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] netsock2.c: Using SIP RTP CoS mark 5 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Setting NAT on RTP to On [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing session-level SDP o=100 8001 8000 IN IP4 192.168.1.6... OK. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED OR FAILED. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] netsock2.c: Splitting '192.168.1.6' into... [Dec 9 19:35:46] DEBUG[85170][C-000006f9] netsock2.c: ...host '192.168.1.6' and port ''. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.6... OK. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Found RTP audio format 125 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] rtp_engine.c: Don't have a default tx payload type 125 format for m type on 0x7f4e89363180 [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Found RTP audio format 0 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f4e89363180 [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Found RTP audio format 97 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] rtp_engine.c: Setting tx payload type 97 based on m type on 0x7f4e89363180 [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Found RTP audio format 18 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] rtp_engine.c: Setting tx payload type 18 based on m type on 0x7f4e89363180 [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Found RTP audio format 9 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] rtp_engine.c: Setting tx payload type 9 based on m type on 0x7f4e89363180 [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Found RTP audio format 101 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7f4e89363180 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Found audio description format opus for ID 125 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:125 opus/48000/2... OK. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Found audio description format PCMU for ID 0 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Found audio description format iLBC for ID 97 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 iLBC/8000... OK. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=fmtp:97 mode=30... OK. [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Found audio description format G729 for ID 18 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK. [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Found audio description format G722 for ID 9 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Capabilities: us - (ulaw|ilbc|g729), peer - audio=(ulaw|g722|g729|ilbc|opus)/video=(nothing)/text=(nothing), combined - (ulaw|ilbc|g729) [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Dec 9 19:35:46] DEBUG[85170][C-000006f9] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] acl.c: For destination '192.168.1.6', our source address is '207.91.156.217'. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4e80002c00' [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Peer audio RTP is at port 192.168.1.6:5006 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] rtp_engine.c: Copying payload 0 (0x7f4e8001e608) from 0x7f4e89363180 to 0x7f4e80002dc8 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] rtp_engine.c: Copying payload 9 (0x7f4e80040118) from 0x7f4e89363180 to 0x7f4e80002dc8 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] rtp_engine.c: Copying payload 18 (0x7f4e800333e8) from 0x7f4e89363180 to 0x7f4e80002dc8 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] rtp_engine.c: Copying payload 97 (0x7f4e80035d68) from 0x7f4e89363180 to 0x7f4e80002dc8 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] rtp_engine.c: Copying payload 101 (0x7f4e80031738) from 0x7f4e89363180 to 0x7f4e80002dc8 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] rtp_engine.c: Copying payload 125 (0x7f4e80015d28) from 0x7f4e89363180 to 0x7f4e80002dc8 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f4e80002c00' [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: We're settling with these formats: (ulaw|ilbc|g729) [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Checking SIP call limits for device 100 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Updating call counter for incoming call [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Call from peer '100' is 1 out of 2147483647 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:35:46] DEBUG[85170][C-000006f9] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:35:46] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:35:46] DEBUG[85170][C-000006f9] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Looking for 8018309060 in from-internal (domain testbed2.primevox.net) [Dec 9 19:35:46] DEBUG[85149] chan_sip.c: Checking device state for peer 100 [Dec 9 19:35:46] DEBUG[85149] devicestate.c: Changing state for SIP/100 - state 2 (In use) [Dec 9 19:35:46] DEBUG[85210] app_queue.c: Device 'SIP/100' changed to state '2' (In use) [Dec 9 19:35:46] DEBUG[85210] app_queue.c: Device 'Queue:Technical Support_avail' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Incoming INVITE with 'timer' option supported [Dec 9 19:35:46] DEBUG[85151] app_queue.c: Extension '100@from-internal' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: DeviceStateChange Privilege: call,all Device: SIP/100 State: INUSE [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: from-internal Hint: SIP/100 Status: 1 StatusText: InUse [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: QueueMemberStatus Privilege: agent,all Ringinuse: 0 StateInterface: SIP/100 CallsTaken: 0 Queue: Technical Support Membership: static Status: 2 MemberName: SIP/100 Penalty: 0 Interface: SIP/100 LastCall: 0 InCall: 0 Paused: 0 PausedReason: [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: *** Our native formats are (ulaw) [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: *** Joint capabilities are (ulaw|ilbc|g729) [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: *** Our capabilities are (ulaw|ilbc|g729) [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: DeviceStateChange Privilege: call,all Device: Queue:Technical Support_avail State: INUSE [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: This channel will not be able to handle video. [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newchannel Privilege: call,all Channel: SIP/100-0000006d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-internal Exten: 8018309060 Priority: 1 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-internal Exten: 8018309060 Priority: 1 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: SIPURI Value: sip:100@192.168.1.6:5060 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: SIP/100-0000006d: New call is still down.... Trying... [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: <--- Transmitting (NAT) to 72.190.32.250:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK548259106;received=72.190.32.250;rport=5060 From: "Local1" ;tag=1110989379 To: Call-ID: 1197569442-5060-26@BJC.BGI.B.G CSeq: 230 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-internal Exten: 8018309060 Priority: 1 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: SIPDOMAIN Value: testbed2.primevox.net [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-internal Exten: 8018309060 Priority: 1 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: SIPCALLID Value: 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:35:46] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Dec 9 19:35:46] DEBUG[85149] chan_sip.c: Checking device state for peer 100 [Dec 9 19:35:46] DEBUG[85149] devicestate.c: Changing state for SIP/100 - state 2 (In use) [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newstate Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-internal Exten: 8018309060 Priority: 1 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx_variables.c: Result of 'EXTEN' is '8018309060' [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx.c: Launching 'Goto' [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: SuccessfulAuth Privilege: security,all EventTV: 2016-12-09T19:35:46.003+0000 Severity: Informational Service: SIP EventVersion: 1 AccountID: 8018309060 SessionID: 0x7f4e80023870 LocalAddress: IPV4/UDP/207.91.156.217/5060 RemoteAddress: IPV4/UDP/72.190.32.250/5060 UsingPassword: 1 [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx.c: Executing [8018309060@from-internal:1] Goto("SIP/100-0000006d", "dial-out-100,8018309060,1") in new stack [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx_builtins.c: Goto (dial-out-100,8018309060,1) [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-internal Exten: 8018309060 Priority: 1 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Extension: 8018309060 Application: Goto AppData: dial-out-100,8018309060,1 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx_variables.c: Result of 'EXTEN' is '8018309060' [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx.c: Launching 'Set' [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx.c: Executing [8018309060@dial-out-100:1] Set("SIP/100-0000006d", "temp=18018309060") in new stack [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx.c: Launching 'Goto' [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx.c: Executing [8018309060@dial-out-100:2] Goto("SIP/100-0000006d", "_1XXXXXX.,modified") in new stack [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: 8018309060 Priority: 1 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Extension: 8018309060 Application: Set AppData: temp=18018309060 [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx_builtins.c: Goto (dial-out-100,_1XXXXXX.,2) [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: 8018309060 Priority: 1 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: temp Value: 18018309060 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx_variables.c: Function CDR(cdr_direction) result is '' [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx_variables.c: Expression result is '0' [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx.c: Launching 'GotoIf' [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx.c: Executing [_1XXXXXX.@dial-out-100:2] GotoIf("SIP/100-0000006d", "0?keepDir") in new stack [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: 8018309060 Priority: 2 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Extension: 8018309060 Application: Goto AppData: _1XXXXXX.,modified [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx_builtins.c: Not taking any branch [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx.c: Launching 'Set' [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx.c: Executing [_1XXXXXX.@dial-out-100:3] Set("SIP/100-0000006d", "CDR(cdr_direction)=2") in new stack [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 2 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Extension: _1XXXXXX. Application: GotoIf AppData: 0?keepDir [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 3 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Extension: _1XXXXXX. Application: Set AppData: CDR(cdr_direction)=2 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx_variables.c: Result of 'temp' is '18018309060' [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx.c: Launching 'Set' [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx.c: Executing [_1XXXXXX.@dial-out-100:4] Set("SIP/100-0000006d", "CDR(cdr_outbound_dialed_phonenumber)=18018309060") in new stack [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx.c: Launching 'Set' [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx.c: Executing [_1XXXXXX.@dial-out-100:5] Set("SIP/100-0000006d", "CDR(cdr_outbound_from_extension)=100") in new stack [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 4 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Extension: _1XXXXXX. Application: Set AppData: CDR(cdr_outbound_dialed_phonenumber)=18018309060 [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 5 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Extension: _1XXXXXX. Application: Set AppData: CDR(cdr_outbound_from_extension)=100 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx.c: Launching 'Set' [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx.c: Executing [_1XXXXXX.@dial-out-100:6] Set("SIP/100-0000006d", "CDR(cdr_application)=Outbound Dial") in new stack [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx.c: Launching 'Set' [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx.c: Executing [_1XXXXXX.@dial-out-100:7] Set("SIP/100-0000006d", "CALLERID(all)="Luke Apartment" <9726001152>") in new stack [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 6 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Extension: _1XXXXXX. Application: Set AppData: CDR(cdr_application)=Outbound Dial [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx.c: Launching 'Set' [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx.c: Executing [_1XXXXXX.@dial-out-100:8] Set("SIP/100-0000006d", "CDR(cdr_outbound_did)=9726001152") in new stack [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx_variables.c: Result of 'temp' is '18018309060' [Dec 9 19:35:46] DEBUG[114537][C-000006f9] pbx.c: Launching 'Dial' [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] pbx.c: Executing [_1XXXXXX.@dial-out-100:9] Dial("SIP/100-0000006d", "SIP/flowroute/18018309060") in new stack [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 7 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Extension: _1XXXXXX. Application: Set AppData: CALLERID(all)="Luke Apartment" <9726001152> [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: NewCallerid Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 7 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Asked to create a SIP channel with formats: (ulaw) [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Allocating new SIP dialog for 3d938c5673b414434558e14063c1ab1c@207.91.156.217:5060 - INVITE (No RTP) [Dec 9 19:35:46] DEBUG[114537][C-000006f9] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f4e7c00a0f0' [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 8 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Extension: _1XXXXXX. Application: Set AppData: CDR(cdr_outbound_did)=9726001152 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Allocated port 14074 for RTP instance '0x7f4e7c00a0f0' [Dec 9 19:35:46] DEBUG[114537][C-000006f9] rtp_engine.c: RTP instance '0x7f4e7c00a0f0' is setup and ready to go [Dec 9 19:35:46] DEBUG[114537][C-000006f9] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Dec 9 19:35:46] DEBUG[114537][C-000006f9] netsock2.c: Splitting 'testbed2.primevox.net' into... [Dec 9 19:35:46] DEBUG[114537][C-000006f9] netsock2.c: ...host 'testbed2.primevox.net' and port ''. [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Extension: _1XXXXXX. Application: Dial AppData: SIP/flowroute/18018309060 [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: DIALSTATUS Value: [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: DIALEDPEERNUMBER Value: [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: DIALEDPEERNAME Value: [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: ANSWEREDTIME Value: [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: DIALEDTIME Value: [Dec 9 19:35:46] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f4e7c00a0f0' [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] netsock2.c: Using SIP RTP CoS mark 5 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Setting NAT on RTP to On [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:46] DEBUG[114537][C-000006f9] acl.c: For destination '216.115.69.144', our source address is '207.91.156.217'. [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Setting NAT on RTP to On [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: SIP call-id changed from '3d938c5673b414434558e14063c1ab1c@207.91.156.217:5060' to '4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060' [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: *** Our native formats are (ulaw) [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: *** Joint capabilities are (ulaw) [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: *** Our capabilities are (ulaw) [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: *** Our preferred formats from the incoming channel are (ulaw) [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: This channel will not be able to handle video. [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newchannel Privilege: call,all Channel: SIP/flowroute-0000006e ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-trunk Exten: s Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] channel_internal_api.c: Channel Call ID changing from [C-000006f9] to [C-000006f9] [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-trunk Exten: s Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: SIPCALLID Value: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Outgoing Call for 18018309060 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Updating call counter for outgoing call [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: DIALEDPEERNUMBER Value: flowroute/18018309060 [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/flowroute-0000006e ChannelState: 0 ChannelStateDesc: Down CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: _1XXXXXX. Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Extension: _1XXXXXX. Application: AppDial AppData: (Outgoing Line) [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: NewCallerid Privilege: call,all Channel: SIP/flowroute-0000006e ChannelState: 0 ChannelStateDesc: Down CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: _1XXXXXX. Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: NewConnectedLine Privilege: call,all Channel: SIP/flowroute-0000006e ChannelState: 0 ChannelStateDesc: Down CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: _1XXXXXX. Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: ** Our capability: (ulaw) Video flag: False Text flag: False [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: ** Our prefcodec: (ulaw) [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] chan_sip.c: Audio is at 14074 [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] chan_sip.c: Adding codec ulaw to SDP [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: -- Done with adding codecs to SDP [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Initializing initreq for method INVITE - callid 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 0 [ 48]: INVITE sip:18018309060@sip.flowroute.com SIP/2.0 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK261a0ddc;rport [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 3 [ 67]: From: "Luke Apartment" ;tag=as4389a784 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 4 [ 39]: To: [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 6 [ 61]: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:35:46 GMT [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 12 [101]: Remote-Party-ID: "Luke Apartment" ;party=calling;privacy=off;screen=no [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] chan_sip.c: Reliably Transmitting (NAT) to 216.115.69.144:5060: INVITE sip:18018309060@sip.flowroute.com SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK261a0ddc;rport Max-Forwards: 70 From: "Luke Apartment" ;tag=as4389a784 To: Contact: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:35:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "Luke Apartment" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 244 v=0 o=root 1879703968 1879703968 IN IP4 207.91.156.217 s=Asterisk PBX 13.13.1 c=IN IP4 207.91.156.217 t=0 0 m=audio 14074 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #76 [Dec 9 19:35:46] DEBUG[114537][C-000006f9] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:35:46] VERBOSE[114537][C-000006f9] app_dial.c: Called SIP/flowroute/18018309060 [Dec 9 19:35:46] DEBUG[94779] manager.c: Examining AMI event: Event: DialBegin Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 DestChannel: SIP/flowroute-0000006e DestChannelState: 0 DestChannelStateDesc: Down DestCallerIDNum: _1XXXXXX. DestCallerIDName: DestConnectedLineNum: 9726001152 DestConnectedLineName: Luke Apartment DestLanguage: en DestAccountCode: DestContext: from-trunk DestExten: _1XXXXXX. DestPriority: 1 DestUniqueid: 1481312146.222 DestLinkedid: 1481312146.221 DialString: flowroute/18018309060 [Dec 9 19:35:46] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK261a0ddc;rport=5060 From: "Luke Apartment" ;tag=as4389a784 To: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK261a0ddc;rport=5060 [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 2 [ 67]: From: "Luke Apartment" ;tag=as4389a784 [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 3 [ 39]: To: [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Dec 9 19:35:46] VERBOSE[85170] chan_sip.c: --- (7 headers 0 lines) --- [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: = Looking for Call ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 (Checking To) --From tag as4389a784 --To-tag [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: *** SIP TIMER: Cancelling retransmission #76 - INVITE (got response) [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060' Request 102: Found [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: SIP response 100 to standard invite [Dec 9 19:35:46] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK261a0ddc;rport=5060 From: "Luke Apartment" ;tag=as4389a784 To: ;tag=870a5b262384e4f9f82f59836d699db5.1ca8 Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="sip.flowroute.com", nonce="WEsIvlhLB5IB6GWiUlO/HmNmE+ziYT3s", qop="auth" Content-Length: 0 <-------------> [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 0 [ 41]: SIP/2.0 407 Proxy Authentication Required [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK261a0ddc;rport=5060 [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 2 [ 67]: From: "Luke Apartment" ;tag=as4389a784 [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 3 [ 81]: To: ;tag=870a5b262384e4f9f82f59836d699db5.1ca8 [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 6 [106]: Proxy-Authenticate: Digest realm="sip.flowroute.com", nonce="WEsIvlhLB5IB6GWiUlO/HmNmE+ziYT3s", qop="auth" [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 9 19:35:46] VERBOSE[85170] chan_sip.c: --- (8 headers 0 lines) --- [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: = Looking for Call ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 (Checking To) --From tag as4389a784 --To-tag 870a5b262384e4f9f82f59836d699db5.1ca8 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Acked pending invite 102 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Stopping retransmission on '4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: SIP response 407 to standard invite [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Transmitting (NAT) to 216.115.69.144:5060: ACK sip:18018309060@sip.flowroute.com SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK261a0ddc;rport Max-Forwards: 70 From: "Luke Apartment" ;tag=as4389a784 To: ;tag=870a5b262384e4f9f82f59836d699db5.1ca8 Contact: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 --- [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Trying to put 'ACK sip:180' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Auth attempt 1 on INVITE [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: ** Our capability: (ulaw) Video flag: False Text flag: False [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: ** Our prefcodec: (ulaw) [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Audio is at 14074 [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Adding codec ulaw to SDP [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: -- Done with adding codecs to SDP [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Dec 9 19:35:46] VERBOSE[85170][C-000006f9] chan_sip.c: Reliably Transmitting (NAT) to 216.115.69.144:5060: INVITE sip:18018309060@sip.flowroute.com SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3726f1b2;rport Max-Forwards: 70 From: "Luke Apartment" ;tag=as4389a784 To: Contact: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 13.13.1 Proxy-Authorization: Digest username="23616016", realm="sip.flowroute.com", algorithm=MD5, uri="sip:18018309060@sip.flowroute.com", nonce="WEsIvlhLB5IB6GWiUlO/HmNmE+ziYT3s", response="d686b623e8eed042c61f99816c01618a", qop=auth, cnonce="726b03e9", nc=00000001 Date: Fri, 09 Dec 2016 19:35:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "Luke Apartment" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 244 v=0 o=root 1879703968 1879703969 IN IP4 207.91.156.217 s=Asterisk PBX 13.13.1 c=IN IP4 207.91.156.217 t=0 0 m=audio 14074 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #11 [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:35:46] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3726f1b2;rport=5060 From: "Luke Apartment" ;tag=as4389a784 To: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 CSeq: 103 INVITE Content-Length: 0 <-------------> [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3726f1b2;rport=5060 [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 2 [ 67]: From: "Luke Apartment" ;tag=as4389a784 [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 3 [ 39]: To: [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Dec 9 19:35:46] VERBOSE[85170] chan_sip.c: --- (7 headers 0 lines) --- [Dec 9 19:35:46] DEBUG[85170] chan_sip.c: = Looking for Call ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 (Checking To) --From tag as4389a784 --To-tag [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: *** SIP TIMER: Cancelling retransmission #11 - INVITE (got response) [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060' Request 103: Found [Dec 9 19:35:46] DEBUG[85170][C-000006f9] chan_sip.c: SIP response 100 to standard invite [Dec 9 19:35:47] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 180 Ringing From: "Luke Apartment" ;tag=as4389a784 To: ;tag=gK08d946fd Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3726f1b2;rport=5060 Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 CSeq: 103 INVITE Record-Route: Record-Route: Contact: Content-Length: 0 <-------------> [Dec 9 19:35:47] DEBUG[85170] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Dec 9 19:35:47] DEBUG[85170] chan_sip.c: Header 1 [ 67]: From: "Luke Apartment" ;tag=as4389a784 [Dec 9 19:35:47] DEBUG[85170] chan_sip.c: Header 2 [ 54]: To: ;tag=gK08d946fd [Dec 9 19:35:47] DEBUG[85170] chan_sip.c: Header 3 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3726f1b2;rport=5060 [Dec 9 19:35:47] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:35:47] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Dec 9 19:35:47] DEBUG[85170] chan_sip.c: Header 6 [ 42]: Record-Route: [Dec 9 19:35:47] DEBUG[85170] chan_sip.c: Header 7 [ 42]: Record-Route: [Dec 9 19:35:47] DEBUG[85170] chan_sip.c: Header 8 [ 45]: Contact: [Dec 9 19:35:47] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:35:47] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:35:47] DEBUG[85170] chan_sip.c: = Looking for Call ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 (Checking To) --From tag as4389a784 --To-tag gK08d946fd [Dec 9 19:35:47] DEBUG[85170][C-000006f9] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060' Request 103: Found [Dec 9 19:35:47] DEBUG[85170][C-000006f9] chan_sip.c: SIP response 180 to standard invite [Dec 9 19:35:47] DEBUG[85170][C-000006f9] sip/route.c: sip_route_process_header: [Dec 9 19:35:47] DEBUG[85170][C-000006f9] sip/route.c: sip_route_process_header: [Dec 9 19:35:47] VERBOSE[85170][C-000006f9] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:47] VERBOSE[85170][C-000006f9] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:47] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - flowroute [Dec 9 19:35:47] DEBUG[85149] chan_sip.c: Checking device state for peer flowroute [Dec 9 19:35:47] VERBOSE[114537][C-000006f9] app_dial.c: SIP/flowroute-0000006e is ringing [Dec 9 19:35:47] DEBUG[85149] devicestate.c: Changing state for SIP/flowroute - state 1 (Not in use) [Dec 9 19:35:47] DEBUG[114537][C-000006f9] rtp_engine.c: Setting early bridge SDP of 'SIP/100-0000006d' with that of 'SIP/flowroute-0000006e' [Dec 9 19:35:47] DEBUG[94779] manager.c: Examining AMI event: Event: Newstate Privilege: call,all Channel: SIP/flowroute-0000006e ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: _1XXXXXX. Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 [Dec 9 19:35:47] VERBOSE[114537][C-000006f9] chan_sip.c: <--- Transmitting (NAT) to 72.190.32.250:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK548259106;received=72.190.32.250;rport=5060 From: "Local1" ;tag=1110989379 To: ;tag=as4325b685 Call-ID: 1197569442-5060-26@BJC.BGI.B.G CSeq: 230 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Dec 9 19:35:47] DEBUG[114537][C-000006f9] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 42a9ed4c17f760c219fcc487051295e1@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:48] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: SIP call-id changed from '42a9ed4c17f760c219fcc487051295e1@207.91.156.217:5060' to '091e3a0e3cc90f1879406a47569f5241@207.91.156.217:5060' [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 091e3a0e3cc90f1879406a47569f5241@207.91.156.217:5060 [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Header 0 [ 40]: OPTIONS sip:100@192.168.1.6:5060 SIP/2.0 [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK63e11cd9;rport [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as6953b7ad [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Header 4 [ 30]: To: [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 091e3a0e3cc90f1879406a47569f5241@207.91.156.217:5060 [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:35:48 GMT [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:35:48] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 72.190.32.250:5060: OPTIONS sip:100@192.168.1.6:5060 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK63e11cd9;rport Max-Forwards: 70 From: "asterisk" ;tag=as6953b7ad To: Contact: Call-ID: 091e3a0e3cc90f1879406a47569f5241@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:35:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Dec 9 19:35:48] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:49] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK63e11cd9;rport=5060 From: "asterisk" ;tag=as6953b7ad To: ;tag=80572475 Call-ID: 091e3a0e3cc90f1879406a47569f5241@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK63e11cd9;rport=5060 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as6953b7ad [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 3 [ 43]: To: ;tag=80572475 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 091e3a0e3cc90f1879406a47569f5241@207.91.156.217:5060 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:35:49] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: = Looking for Call ID: 091e3a0e3cc90f1879406a47569f5241@207.91.156.217:5060 (Checking To) --From tag as6953b7ad --To-tag 80572475 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Stopping retransmission on '091e3a0e3cc90f1879406a47569f5241@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Destroying SIP dialog 091e3a0e3cc90f1879406a47569f5241@207.91.156.217:5060 [Dec 9 19:35:49] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '091e3a0e3cc90f1879406a47569f5241@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:35:49] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 180 Ringing From: "Luke Apartment" ;tag=as4389a784 To: ;tag=gK08d946fd Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3726f1b2;rport=5060 Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 CSeq: 103 INVITE Record-Route: Record-Route: Contact: Content-Length: 172 Content-Type: application/sdp v=0 o=- 329610 688916 IN IP4 74.120.93.27 s=- c=IN IP4 74.120.93.27 t=0 0 m=audio 53072 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 1 [ 67]: From: "Luke Apartment" ;tag=as4389a784 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 2 [ 54]: To: ;tag=gK08d946fd [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 3 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3726f1b2;rport=5060 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 6 [ 42]: Record-Route: [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 7 [ 42]: Record-Route: [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 8 [ 45]: Contact: [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 9 [ 19]: Content-Length: 172 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Header 11 [ 0]: [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Body 0 [ 3]: v=0 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Body 1 [ 37]: o=- 329610 688916 IN IP4 74.120.93.27 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Body 2 [ 3]: s=- [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Body 3 [ 21]: c=IN IP4 74.120.93.27 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Body 5 [ 27]: m=audio 53072 RTP/AVP 0 101 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Body 6 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Body 7 [ 15]: a=fmtp:101 0-15 [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: Body 8 [ 10]: a=ptime:20 [Dec 9 19:35:49] VERBOSE[85170] chan_sip.c: --- (11 headers 9 lines) --- [Dec 9 19:35:49] DEBUG[85170] chan_sip.c: = Looking for Call ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 (Checking To) --From tag as4389a784 --To-tag gK08d946fd [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060' Request 103: Found [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: SIP response 180 to standard invite [Dec 9 19:35:49] DEBUG[85170][C-000006f9] sip/route.c: sip_route_process_header: [Dec 9 19:35:49] DEBUG[85170][C-000006f9] sip/route.c: sip_route_process_header: [Dec 9 19:35:49] VERBOSE[85170][C-000006f9] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:49] VERBOSE[85170][C-000006f9] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: Processing session-level SDP o=- 329610 688916 IN IP4 74.120.93.27... OK. [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [Dec 9 19:35:49] DEBUG[85170][C-000006f9] netsock2.c: Splitting '74.120.93.27' into... [Dec 9 19:35:49] DEBUG[85170][C-000006f9] netsock2.c: ...host '74.120.93.27' and port ''. [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: Processing session-level SDP c=IN IP4 74.120.93.27... OK. [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Dec 9 19:35:49] VERBOSE[85170][C-000006f9] chan_sip.c: Found RTP audio format 0 [Dec 9 19:35:49] DEBUG[85170][C-000006f9] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f4e893623e0 [Dec 9 19:35:49] VERBOSE[85170][C-000006f9] chan_sip.c: Found RTP audio format 101 [Dec 9 19:35:49] DEBUG[85170][C-000006f9] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7f4e893623e0 [Dec 9 19:35:49] VERBOSE[85170][C-000006f9] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Dec 9 19:35:49] VERBOSE[85170][C-000006f9] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Dec 9 19:35:49] VERBOSE[85170][C-000006f9] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Dec 9 19:35:49] DEBUG[85170][C-000006f9] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Dec 9 19:35:49] DEBUG[85170][C-000006f9] acl.c: For destination '74.120.93.27', our source address is '207.91.156.217'. [Dec 9 19:35:49] DEBUG[85170][C-000006f9] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4e7c00a0f0' [Dec 9 19:35:49] VERBOSE[85170][C-000006f9] chan_sip.c: Peer audio RTP is at port 74.120.93.27:53072 [Dec 9 19:35:49] DEBUG[85170][C-000006f9] rtp_engine.c: Copying payload 0 (0x307cf18) from 0x7f4e893623e0 to 0x7f4e7c00a2b8 [Dec 9 19:35:49] DEBUG[85170][C-000006f9] rtp_engine.c: Copying payload 101 (0x7f4e800d8a28) from 0x7f4e893623e0 to 0x7f4e7c00a2b8 [Dec 9 19:35:49] DEBUG[85170][C-000006f9] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f4e7c00a0f0' [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: We're settling with these formats: (ulaw) [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: We have an owner, now see if we need to change this call [Dec 9 19:35:49] DEBUG[85170][C-000006f9] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Dec 9 19:35:49] VERBOSE[114537][C-000006f9] app_dial.c: SIP/flowroute-0000006e is ringing [Dec 9 19:35:49] DEBUG[114537][C-000006f9] rtp_engine.c: Setting early bridge SDP of 'SIP/100-0000006d' with that of 'SIP/flowroute-0000006e' [Dec 9 19:35:49] VERBOSE[114537][C-000006f9] app_dial.c: SIP/flowroute-0000006e is making progress passing it to SIP/100-0000006d [Dec 9 19:35:49] DEBUG[114537][C-000006f9] rtp_engine.c: Setting early bridge SDP of 'SIP/100-0000006d' with that of 'SIP/flowroute-0000006e' [Dec 9 19:35:49] DEBUG[114537][C-000006f9] chan_sip.c: ** Our capability: (ulaw|ilbc|g729) Video flag: True Text flag: True [Dec 9 19:35:49] DEBUG[114537][C-000006f9] chan_sip.c: ** Our prefcodec: (nothing) [Dec 9 19:35:49] VERBOSE[114537][C-000006f9] chan_sip.c: Audio is at 14788 [Dec 9 19:35:49] VERBOSE[114537][C-000006f9] chan_sip.c: Adding codec ulaw to SDP [Dec 9 19:35:49] VERBOSE[114537][C-000006f9] chan_sip.c: Adding codec ilbc to SDP [Dec 9 19:35:49] VERBOSE[114537][C-000006f9] chan_sip.c: Adding codec g729 to SDP [Dec 9 19:35:49] VERBOSE[114537][C-000006f9] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 9 19:35:49] DEBUG[114537][C-000006f9] chan_sip.c: -- Done with adding codecs to SDP [Dec 9 19:35:49] DEBUG[114537][C-000006f9] chan_sip.c: Setting framing on incoming call: 30 [Dec 9 19:35:49] DEBUG[114537][C-000006f9] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|ilbc|g729) [Dec 9 19:35:49] VERBOSE[114537][C-000006f9] chan_sip.c: <--- Transmitting (NAT) to 72.190.32.250:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK548259106;received=72.190.32.250;rport=5060 From: "Local1" ;tag=1110989379 To: ;tag=as4325b685 Call-ID: 1197569442-5060-26@BJC.BGI.B.G CSeq: 230 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 327 v=0 o=root 795096357 795096357 IN IP4 207.91.156.217 s=Asterisk PBX 13.13.1 c=IN IP4 207.91.156.217 t=0 0 m=audio 14788 RTP/AVP 0 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 a=maxptime:150 a=sendrecv <------------> [Dec 9 19:35:49] DEBUG[114537][C-000006f9] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:49] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: 0x7f4e8009fc40 -- Probation learning mode pass with source address 72.190.32.250:5006 [Dec 9 19:35:49] DEBUG[114537][C-000006f9] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Dec 9 19:35:49] DEBUG[114537][C-000006f9] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:35:49] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4e80002c00' [Dec 9 19:35:49] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Ooh, format changed from none to ulaw [Dec 9 19:35:49] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f4e7c00a0f0' [Dec 9 19:35:49] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: 0x7f4e7c00abf0 -- Probation learning mode pass with source address 74.120.93.27:53072 [Dec 9 19:35:49] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Ooh, format changed from none to ulaw [Dec 9 19:35:49] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: 0x7f4e8009fc40 -- Probation learning mode pass with source address 72.190.32.250:5006 [Dec 9 19:35:50] DEBUG[114315] threadpool.c: Worker thread idle timeout reached. Dying. [Dec 9 19:35:50] DEBUG[85138] threadpool.c: Destroying worker thread 588 [Dec 9 19:35:54] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:14789' into... [Dec 9 19:35:54] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '14789'. [Dec 9 19:35:54] DEBUG[85207] netsock2.c: Splitting '72.190.32.250:5007' into... [Dec 9 19:35:54] DEBUG[85207] netsock2.c: ...host '72.190.32.250' and port '5007'. [Dec 9 19:35:54] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPSent Privilege: reporting,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 To: 72.190.32.250:5007 From: 207.91.156.217:14789 SSRC: 0x545c8bf5 PT: 200(SR) ReportCount: 1 SentNTP: 1481312154.2221035520 SentRTP: 39600 SentPackets: 166 SentOctets: 39840 Report0SourceSSRC: 0x35670490 Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 42208 Report0SequenceNumberCycles: 0 Report0IAJitter: 20 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:35:54] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPSent Privilege: reporting,all Channel: SIP/flowroute-0000006e ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: _1XXXXXX. Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 To: 74.120.93.27:53073 From: 207.91.156.217:14075 SSRC: 0x38039036 PT: 200(SR) ReportCount: 1 SentNTP: 1481312154.2221330432 SentRTP: 59064880 SentPackets: 167 SentOctets: 40080 Report0SourceSSRC: 0xd3444ceb Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 363 Report0SequenceNumberCycles: 0 Report0IAJitter: 0 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:35:54] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:14075' into... [Dec 9 19:35:54] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '14075'. [Dec 9 19:35:54] DEBUG[85207] netsock2.c: Splitting '74.120.93.27:53073' into... [Dec 9 19:35:54] DEBUG[85207] netsock2.c: ...host '74.120.93.27' and port '53073'. [Dec 9 19:35:54] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Dec 9 19:35:54] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPReceived Privilege: reporting,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 To: 207.91.156.217:14789 From: 72.190.32.250:5007 RTT: 0.0000 SSRC: 0x35670490 PT: 200(SR) ReportCount: 1 SentNTP: 1481312154.17590784552960 SentRTP: 59065120 SentPackets: 167 SentOctets: 40080 Report0SourceSSRC: 0x545c8bf5 Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 15329 Report0SequenceNumberCycles: 0 Report0IAJitter: 79 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:35:54] DEBUG[85207] netsock2.c: Splitting '72.190.32.250:5007' into... [Dec 9 19:35:54] DEBUG[85207] netsock2.c: ...host '72.190.32.250' and port '5007'. [Dec 9 19:35:54] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:14789' into... [Dec 9 19:35:54] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '14789'. [Dec 9 19:35:55] VERBOSE[114537][C-000006f9] app_dial.c: SIP/flowroute-0000006e requested media update control 26, passing it to SIP/100-0000006d [Dec 9 19:35:55] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Changing ssrc from 1415351285 to 1308312189 due to a source change [Dec 9 19:35:55] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 200 OK From: "Luke Apartment" ;tag=as4389a784 To: ;tag=gK08d946fd Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3726f1b2;rport=5060 Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 CSeq: 103 INVITE Record-Route: Record-Route: Contact: Session-Expires: 1800;refresher=uas Content-Length: 172 Content-Type: application/sdp Supported: timer v=0 o=- 329610 688916 IN IP4 74.120.93.27 s=- c=IN IP4 74.120.93.27 t=0 0 m=audio 53072 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 1 [ 67]: From: "Luke Apartment" ;tag=as4389a784 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 2 [ 54]: To: ;tag=gK08d946fd [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 3 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3726f1b2;rport=5060 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 6 [ 42]: Record-Route: [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 7 [ 42]: Record-Route: [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 8 [ 45]: Contact: [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 10 [ 19]: Content-Length: 172 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 12 [ 16]: Supported: timer [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 13 [ 0]: [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Body 0 [ 3]: v=0 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Body 1 [ 37]: o=- 329610 688916 IN IP4 74.120.93.27 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Body 2 [ 3]: s=- [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Body 3 [ 21]: c=IN IP4 74.120.93.27 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Body 5 [ 27]: m=audio 53072 RTP/AVP 0 101 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Body 6 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Body 7 [ 15]: a=fmtp:101 0-15 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Body 8 [ 10]: a=ptime:20 [Dec 9 19:35:55] VERBOSE[85170] chan_sip.c: --- (13 headers 9 lines) --- [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: = Looking for Call ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 (Checking To) --From tag as4389a784 --To-tag gK08d946fd [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: Acked pending invite 103 [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: Stopping retransmission on '4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060' of Request 103: Match Found [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: SIP response 200 to standard invite [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: Call 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 responded to our reinvite without changing SDP version; ignoring SDP. [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: Updating call counter for outgoing call [Dec 9 19:35:55] DEBUG[85170][C-000006f9] sip/route.c: sip_route_process_header: [Dec 9 19:35:55] DEBUG[85170][C-000006f9] sip/route.c: sip_route_process_header: [Dec 9 19:35:55] VERBOSE[85170][C-000006f9] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:55] VERBOSE[85170][C-000006f9] sip/route.c: sip_route_dump: route/path hop: [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: Session-Expires: 1800 [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: Refresher: UAS [Dec 9 19:35:55] VERBOSE[85170][C-000006f9] chan_sip.c: Transmitting (NAT) to 216.115.69.144:5060: ACK sip:+18018309060@74.120.93.24:5060 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK05e7e522;rport Route: , Max-Forwards: 70 From: "Luke Apartment" ;tag=as4389a784 To: ;tag=gK08d946fd Contact: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 --- [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: Trying to put 'ACK sip:+18' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Session timer started: 10 - 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 1768000ms [Dec 9 19:35:55] VERBOSE[114537][C-000006f9] app_dial.c: SIP/flowroute-0000006e answered SIP/100-0000006d [Dec 9 19:35:55] DEBUG[114537][C-000006f9] rtp_engine.c: Setting early bridge SDP of 'SIP/100-0000006d' with that of 'SIP/flowroute-0000006e' [Dec 9 19:35:55] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - flowroute [Dec 9 19:35:55] DEBUG[85149] chan_sip.c: Checking device state for peer flowroute [Dec 9 19:35:55] DEBUG[85149] devicestate.c: Changing state for SIP/flowroute - state 1 (Not in use) [Dec 9 19:35:55] DEBUG[85138] threadpool.c: Increasing threadpool stasis-core's size by 1 [Dec 9 19:35:55] DEBUG[114537][C-000006f9] chan_sip.c: SIP answering channel: SIP/100-0000006d [Dec 9 19:35:55] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Setting the marker bit due to a source update [Dec 9 19:35:55] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Dec 9 19:35:55] DEBUG[85149] chan_sip.c: Checking device state for peer 100 [Dec 9 19:35:55] DEBUG[114537][C-000006f9] chan_sip.c: ** Our capability: (ulaw|ilbc|g729) Video flag: True Text flag: True [Dec 9 19:35:55] DEBUG[85149] devicestate.c: Changing state for SIP/100 - state 2 (In use) [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: Newstate Privilege: call,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: _1XXXXXX. Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 [Dec 9 19:35:55] DEBUG[114537][C-000006f9] chan_sip.c: ** Our prefcodec: (nothing) [Dec 9 19:35:55] VERBOSE[114537][C-000006f9] chan_sip.c: Audio is at 14788 [Dec 9 19:35:55] VERBOSE[114537][C-000006f9] chan_sip.c: Adding codec ulaw to SDP [Dec 9 19:35:55] VERBOSE[114537][C-000006f9] chan_sip.c: Adding codec ilbc to SDP [Dec 9 19:35:55] VERBOSE[114537][C-000006f9] chan_sip.c: Adding codec g729 to SDP [Dec 9 19:35:55] VERBOSE[114537][C-000006f9] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 9 19:35:55] DEBUG[114537][C-000006f9] chan_sip.c: -- Done with adding codecs to SDP [Dec 9 19:35:55] DEBUG[114537][C-000006f9] chan_sip.c: Setting framing on incoming call: 30 [Dec 9 19:35:55] DEBUG[114537][C-000006f9] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|ilbc|g729) [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: DIALSTATUS Value: ANSWER [Dec 9 19:35:55] VERBOSE[114537][C-000006f9] chan_sip.c: <--- Reliably Transmitting (NAT) to 72.190.32.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK548259106;received=72.190.32.250;rport=5060 From: "Local1" ;tag=1110989379 To: ;tag=as4325b685 Call-ID: 1197569442-5060-26@BJC.BGI.B.G CSeq: 230 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 327 v=0 o=root 795096357 795096357 IN IP4 207.91.156.217 s=Asterisk PBX 13.13.1 c=IN IP4 207.91.156.217 t=0 0 m=audio 14788 RTP/AVP 0 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 a=maxptime:150 a=sendrecv <------------> [Dec 9 19:35:55] DEBUG[114537][C-000006f9] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #38 [Dec 9 19:35:55] DEBUG[114537][C-000006f9] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Session timer started: 37 - 1197569442-5060-26@BJC.BGI.B.G 900000ms [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: DIALEDPEERNAME Value: SIP/flowroute-0000006e [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge_native_rtp.c: Bridge '22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba' can not use native RTP bridge as two channels are required [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Chose bridge technology simple_bridge [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: calling simple_bridge technology constructor [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: calling simple_bridge technology start [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: DIALEDPEERNUMBER Value: flowroute/18018309060 [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: DialEnd Privilege: call,all Channel: SIP/100-0000006d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 DestChannel: SIP/flowroute-0000006e DestChannelState: 6 DestChannelStateDesc: Up DestCallerIDNum: _1XXXXXX. DestCallerIDName: DestConnectedLineNum: 9726001152 DestConnectedLineName: Luke Apartment DestLanguage: en DestAccountCode: DestContext: from-trunk DestExten: DestPriority: 1 DestUniqueid: 1481312146.222 DestLinkedid: 1481312146.221 DialStatus: ANSWER [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: Newstate Privilege: call,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge_channel.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: 0x7f4e7c01bfa0(SIP/flowroute-0000006e) is joining [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: BridgeCreate Privilege: call,all BridgeUniqueid: 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba BridgeType: basic BridgeTechnology: simple_bridge BridgeCreator: BridgeName: BridgeNumChannels: 0 BridgeVideoSourceMode: none [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge_channel.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: pushing 0x7f4e7c01bfa0(SIP/flowroute-0000006e) [Dec 9 19:35:55] VERBOSE[114595][C-000006f9] bridge_channel.c: Channel SIP/flowroute-0000006e joined 'simple_bridge' basic-bridge <22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba> [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge_native_rtp.c: Bridge '22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba' can not use native RTP bridge as two channels are required [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge.c: Bridge technology softmix does not have any capabilities we want. [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge.c: Chose bridge technology simple_bridge [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba is already using the new technology. [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: 0x7f4e7c01bfa0(SIP/flowroute-0000006e) is joining simple_bridge technology [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: Newexten Privilege: call,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Extension: Application: AppDial AppData: (Outgoing Line) [Dec 9 19:35:55] DEBUG[114595][C-000006f9] res_rtp_asterisk.c: Changing ssrc from 939757622 to 41559051 due to a source change [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: BridgeEnter Privilege: call,all BridgeUniqueid: 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba BridgeType: basic BridgeTechnology: simple_bridge BridgeCreator: BridgeName: BridgeNumChannels: 1 BridgeVideoSourceMode: none Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge_channel.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: 0x7f4e7c013020(SIP/100-0000006d) is joining [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge_channel.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: pushing 0x7f4e7c013020(SIP/100-0000006d) [Dec 9 19:35:55] VERBOSE[114537][C-000006f9] bridge_channel.c: Channel SIP/100-0000006d joined 'simple_bridge' basic-bridge <22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba> [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge technology softmix does not have any capabilities we want. [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Chose bridge technology native_rtp [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: calling native_rtp technology constructor [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: moving 0x7f4e7c01bfa0(SIP/flowroute-0000006e) to dummy bridge temporarily [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: 0x7f4e7c01bfa0(SIP/flowroute-0000006e) is leaving simple_bridge technology (dummy) [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: BridgeEnter Privilege: call,all BridgeUniqueid: 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba BridgeType: basic BridgeTechnology: simple_bridge BridgeCreator: BridgeName: BridgeNumChannels: 2 BridgeVideoSourceMode: none Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: calling simple_bridge technology stop [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: 0x7f4e7c013020(SIP/100-0000006d) is joining native_rtp technology [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: 0x7f4e7c01bfa0(SIP/flowroute-0000006e) is joining native_rtp technology [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: calling native_rtp technology start [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: calling simple_bridge technology destructor [Dec 9 19:35:55] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Changing ssrc from 1308312189 to 1963630971 due to a source change [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge technology softmix does not have any capabilities we want. [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Chose bridge technology native_rtp [Dec 9 19:35:55] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba is already using the new technology. [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: BRIDGEPEER Value: SIP/flowroute-0000006e [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: BRIDGEPVTCALLID Value: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: BRIDGEPEER Value: SIP/100-0000006d [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: BRIDGEPVTCALLID Value: 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:35:55] DEBUG[85150] cdr.c: Finalized CDR for SIP/flowroute-0000006e - start 1481312146.009673 answer 1481312155.192600 end 1481312155.193410 dispo ANSWERED [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge.c: Bridge technology softmix does not have any capabilities we want. [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge.c: Chose bridge technology native_rtp [Dec 9 19:35:55] DEBUG[114595][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba is already using the new technology. [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: BRIDGEPEER Value: SIP/flowroute-0000006e [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: BRIDGEPVTCALLID Value: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: BRIDGEPEER Value: SIP/100-0000006d [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: BRIDGEPVTCALLID Value: 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: BRIDGEPEER Value: SIP/flowroute-0000006e [Dec 9 19:35:55] DEBUG[85138] threadpool.c: Increasing threadpool stasis-core's size by 1 [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: BRIDGEPVTCALLID Value: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: BRIDGEPEER Value: SIP/100-0000006d [Dec 9 19:35:55] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: BRIDGEPVTCALLID Value: 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:35:55] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Got RTCP report of 12 bytes [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: SIP TIMER: Rescheduling retransmission #38 (1) SIP/2.0 - 1 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #38)) [Dec 9 19:35:55] VERBOSE[85170] chan_sip.c: Retransmitting #1 (NAT) to 72.190.32.250:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK548259106;received=72.190.32.250;rport=5060 From: "Local1" ;tag=1110989379 To: ;tag=as4325b685 Call-ID: 1197569442-5060-26@BJC.BGI.B.G CSeq: 230 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 327 v=0 o=root 795096357 795096357 IN IP4 207.91.156.217 s=Asterisk PBX 13.13.1 c=IN IP4 207.91.156.217 t=0 0 m=audio 14788 RTP/AVP 0 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 a=maxptime:150 a=sendrecv --- [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:35:55] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> ACK sip:8018309060@207.91.156.217:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK836167500;rport From: "Local1" ;tag=1110989379 To: ;tag=as4325b685 Call-ID: 1197569442-5060-26@BJC.BGI.B.G CSeq: 230 ACK Contact: Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 0 [ 46]: ACK sip:8018309060@207.91.156.217:5060 SIP/2.0 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK836167500;rport [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "Local1" ;tag=1110989379 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 3 [ 57]: To: ;tag=as4325b685 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 4 [ 39]: Call-ID: 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 5 [ 13]: CSeq: 230 ACK [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 6 [ 35]: Contact: [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 8 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 10 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Dec 9 19:35:55] VERBOSE[85170] chan_sip.c: --- (12 headers 0 lines) --- [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: = Looking for Call ID: 1197569442-5060-26@BJC.BGI.B.G (Checking From) --From tag 1110989379 --To-tag as4325b685 [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #38 [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: Stopping retransmission on '1197569442-5060-26@BJC.BGI.B.G' of Response 230: Match Found [Dec 9 19:35:55] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> ACK sip:8018309060@207.91.156.217:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK836167500;rport From: "Local1" ;tag=1110989379 To: ;tag=as4325b685 Call-ID: 1197569442-5060-26@BJC.BGI.B.G CSeq: 230 ACK Contact: Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 0 [ 46]: ACK sip:8018309060@207.91.156.217:5060 SIP/2.0 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK836167500;rport [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "Local1" ;tag=1110989379 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 3 [ 57]: To: ;tag=as4325b685 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 4 [ 39]: Call-ID: 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 5 [ 13]: CSeq: 230 ACK [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 6 [ 35]: Contact: [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 8 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 10 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Dec 9 19:35:55] VERBOSE[85170] chan_sip.c: --- (12 headers 0 lines) --- [Dec 9 19:35:55] DEBUG[85170] chan_sip.c: = Looking for Call ID: 1197569442-5060-26@BJC.BGI.B.G (Checking From) --From tag 1110989379 --To-tag as4325b685 [Dec 9 19:35:55] DEBUG[85170][C-000006f9] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 170086b87dacc4165606795a622bf135@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:35:56] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: SIP call-id changed from '170086b87dacc4165606795a622bf135@207.91.156.217:5060' to '6ee47b4e12fdd6b402b011de1c823f3c@207.91.156.217:5060' [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 6ee47b4e12fdd6b402b011de1c823f3c@207.91.156.217:5060 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 0 [ 44]: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3fc3c780;rport [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as14bc7dc0 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 4 [ 34]: To: [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 6ee47b4e12fdd6b402b011de1c823f3c@207.91.156.217:5060 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:35:56 GMT [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:35:56] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 97.77.119.234:47976: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3fc3c780;rport Max-Forwards: 70 From: "asterisk" ;tag=as14bc7dc0 To: Contact: Call-ID: 6ee47b4e12fdd6b402b011de1c823f3c@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:35:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:35:56] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3fc3c780;rport=5060 From: "asterisk" ;tag=as14bc7dc0 To: ;tag=1307965825 Call-ID: 6ee47b4e12fdd6b402b011de1c823f3c@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3fc3c780;rport=5060 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as14bc7dc0 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 3 [ 49]: To: ;tag=1307965825 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 6ee47b4e12fdd6b402b011de1c823f3c@207.91.156.217:5060 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:35:56] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: = Looking for Call ID: 6ee47b4e12fdd6b402b011de1c823f3c@207.91.156.217:5060 (Checking To) --From tag as14bc7dc0 --To-tag 1307965825 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16 [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Stopping retransmission on '6ee47b4e12fdd6b402b011de1c823f3c@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:35:56] DEBUG[85170] chan_sip.c: Destroying SIP dialog 6ee47b4e12fdd6b402b011de1c823f3c@207.91.156.217:5060 [Dec 9 19:35:56] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '6ee47b4e12fdd6b402b011de1c823f3c@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:35:57] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:35:58] DEBUG[85170] chan_sip.c: Auto destroying SIP dialog '256831031-54249-15@BJC.BGI.C.CEH' [Dec 9 19:35:58] DEBUG[85170] chan_sip.c: Destroying SIP dialog 256831031-54249-15@BJC.BGI.C.CEH [Dec 9 19:35:58] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '256831031-54249-15@BJC.BGI.C.CEH' Method: REGISTER [Dec 9 19:35:59] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:14789' into... [Dec 9 19:35:59] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '14789'. [Dec 9 19:35:59] DEBUG[85207] netsock2.c: Splitting '72.190.32.250:5007' into... [Dec 9 19:35:59] DEBUG[85207] netsock2.c: ...host '72.190.32.250' and port '5007'. [Dec 9 19:35:59] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPSent Privilege: reporting,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 To: 72.190.32.250:5007 From: 207.91.156.217:14789 SSRC: 0x750aa17b PT: 200(SR) ReportCount: 1 SentNTP: 1481312159.2223054848 SentRTP: 44920 SentPackets: 405 SentOctets: 79840 Report0SourceSSRC: 0x35670490 Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 42230 Report0SequenceNumberCycles: 0 Report0IAJitter: 18 Report0LSR: 2249853173 Report0DLSR: 4.9930 [Dec 9 19:35:59] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:14075' into... [Dec 9 19:35:59] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '14075'. [Dec 9 19:35:59] DEBUG[85207] netsock2.c: Splitting '74.120.93.27:53073' into... [Dec 9 19:35:59] DEBUG[85207] netsock2.c: ...host '74.120.93.27' and port '53073'. [Dec 9 19:35:59] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPSent Privilege: reporting,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 To: 74.120.93.27:53073 From: 207.91.156.217:14075 SSRC: 0x027a240b PT: 200(SR) ReportCount: 1 SentNTP: 1481312159.2223370240 SentRTP: 59070160 SentPackets: 325 SentOctets: 78000 Report0SourceSSRC: 0xd3444ceb Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 298 Report0SequenceNumberCycles: 0 Report0IAJitter: 0 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:36:00] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> <-------------> [Dec 9 19:36:00] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Dec 9 19:36:00] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPReceived Privilege: reporting,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 To: 207.91.156.217:14789 From: 72.190.32.250:5007 RTT: 0.0000 SSRC: 0x17a7139a PT: 200(SR) ReportCount: 1 SentNTP: 1481312160.17590784552960 SentRTP: 59111200 SentPackets: 167 SentOctets: 40080 Report0SourceSSRC: 0x09410c9a Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 565 Report0SequenceNumberCycles: 0 Report0IAJitter: 19 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:36:00] DEBUG[85207] netsock2.c: Splitting '72.190.32.250:5007' into... [Dec 9 19:36:00] DEBUG[85207] netsock2.c: ...host '72.190.32.250' and port '5007'. [Dec 9 19:36:00] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:14789' into... [Dec 9 19:36:00] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '14789'. [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 5d4f01b073d431fa1dc44f6e1562dbf0@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:36:04] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: SIP call-id changed from '5d4f01b073d431fa1dc44f6e1562dbf0@207.91.156.217:5060' to '7a6e391f21058be80f4b686f4fd4d262@207.91.156.217:5060' [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 7a6e391f21058be80f4b686f4fd4d262@207.91.156.217:5060 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 0 [ 40]: OPTIONS sip:100@192.168.1.6:5060 SIP/2.0 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK20b2b334;rport [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as01ee3c5e [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 4 [ 30]: To: [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 7a6e391f21058be80f4b686f4fd4d262@207.91.156.217:5060 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:36:04 GMT [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:36:04] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 72.190.32.250:5060: OPTIONS sip:100@192.168.1.6:5060 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK20b2b334;rport Max-Forwards: 70 From: "asterisk" ;tag=as01ee3c5e To: Contact: Call-ID: 7a6e391f21058be80f4b686f4fd4d262@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:36:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #62 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:36:04] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK20b2b334;rport=5060 From: "asterisk" ;tag=as01ee3c5e To: ;tag=443788884 Call-ID: 7a6e391f21058be80f4b686f4fd4d262@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK20b2b334;rport=5060 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as01ee3c5e [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 3 [ 44]: To: ;tag=443788884 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 7a6e391f21058be80f4b686f4fd4d262@207.91.156.217:5060 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:36:04] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: = Looking for Call ID: 7a6e391f21058be80f4b686f4fd4d262@207.91.156.217:5060 (Checking To) --From tag as01ee3c5e --To-tag 443788884 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #62 [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Stopping retransmission on '7a6e391f21058be80f4b686f4fd4d262@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:36:04] DEBUG[85170] chan_sip.c: Destroying SIP dialog 7a6e391f21058be80f4b686f4fd4d262@207.91.156.217:5060 [Dec 9 19:36:04] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '7a6e391f21058be80f4b686f4fd4d262@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:36:04] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:14789' into... [Dec 9 19:36:04] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPSent Privilege: reporting,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 To: 72.190.32.250:5007 From: 207.91.156.217:14789 SSRC: 0x750aa17b PT: 200(SR) ReportCount: 1 SentNTP: 1481312164.2224762880 SentRTP: 44920 SentPackets: 655 SentOctets: 119840 Report0SourceSSRC: 0x35670490 Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 42230 Report0SequenceNumberCycles: 0 Report0IAJitter: 18 Report0LSR: 2250246389 Report0DLSR: 3.9820 [Dec 9 19:36:04] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '14789'. [Dec 9 19:36:04] DEBUG[85207] netsock2.c: Splitting '72.190.32.250:5007' into... [Dec 9 19:36:04] DEBUG[85207] netsock2.c: ...host '72.190.32.250' and port '5007'. [Dec 9 19:36:04] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPSent Privilege: reporting,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 To: 74.120.93.27:53073 From: 207.91.156.217:14075 SSRC: 0x027a240b PT: 200(SR) ReportCount: 1 SentNTP: 1481312164.2225037312 SentRTP: 59070160 SentPackets: 492 SentOctets: 118080 Report0SourceSSRC: 0xd3444ceb Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 298 Report0SequenceNumberCycles: 0 Report0IAJitter: 0 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:36:04] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:14075' into... [Dec 9 19:36:04] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '14075'. [Dec 9 19:36:04] DEBUG[85207] netsock2.c: Splitting '74.120.93.27:53073' into... [Dec 9 19:36:04] DEBUG[85207] netsock2.c: ...host '74.120.93.27' and port '53073'. [Dec 9 19:36:05] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> <-------------> [Dec 9 19:36:05] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Dec 9 19:36:05] DEBUG[94779] manager.c: Examining AMI event: Event: RTCPReceived Privilege: reporting,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 To: 207.91.156.217:14789 From: 72.190.32.250:5007 RTT: 0.0000 SSRC: 0x17a7139a PT: 200(SR) ReportCount: 1 SentNTP: 1481312165.484675584 SentRTP: 59151280 SentPackets: 334 SentOctets: 80160 Report0SourceSSRC: 0x09410c9a Report0FractionLost: 0 Report0CumulativeLost: 0 Report0HighestSequence: 816 Report0SequenceNumberCycles: 0 Report0IAJitter: 16 Report0LSR: 0 Report0DLSR: 0.0000 [Dec 9 19:36:05] DEBUG[85207] netsock2.c: Splitting '72.190.32.250:5007' into... [Dec 9 19:36:05] DEBUG[85207] netsock2.c: ...host '72.190.32.250' and port '5007'. [Dec 9 19:36:05] DEBUG[85207] netsock2.c: Splitting '207.91.156.217:14789' into... [Dec 9 19:36:05] DEBUG[85207] netsock2.c: ...host '207.91.156.217' and port '14789'. [Dec 9 19:36:06] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> BYE sip:8018309060@207.91.156.217:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK858553058;rport From: "Local1" ;tag=1110989379 To: ;tag=as4325b685 Call-ID: 1197569442-5060-26@BJC.BGI.B.G CSeq: 231 BYE Contact: Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 0 [ 46]: BYE sip:8018309060@207.91.156.217:5060 SIP/2.0 [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK858553058;rport [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "Local1" ;tag=1110989379 [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 3 [ 57]: To: ;tag=as4325b685 [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 4 [ 39]: Call-ID: 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 5 [ 13]: CSeq: 231 BYE [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 6 [ 35]: Contact: [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 8 [ 32]: Supported: replaces, path, timer [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 10 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Dec 9 19:36:06] VERBOSE[85170] chan_sip.c: --- (12 headers 0 lines) --- [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: = Looking for Call ID: 1197569442-5060-26@BJC.BGI.B.G (Checking From) --From tag 1110989379 --To-tag as4325b685 [Dec 9 19:36:06] DEBUG[85170][C-000006f9] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Dec 9 19:36:06] DEBUG[85170][C-000006f9] chan_sip.c: Initializing initreq for method BYE - callid 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:36:06] DEBUG[85170][C-000006f9] netsock2.c: Splitting '192.168.1.6:5060' into... [Dec 9 19:36:06] DEBUG[85170][C-000006f9] netsock2.c: ...host '192.168.1.6' and port '5060'. [Dec 9 19:36:06] DEBUG[85170][C-000006f9] chan_sip.c: NAT detected for 192.168.1.6 / 72.190.32.250 [Dec 9 19:36:06] VERBOSE[85170][C-000006f9] chan_sip.c: Sending to 72.190.32.250:5060 (NAT) [Dec 9 19:36:06] DEBUG[85170][C-000006f9] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:36:06] DEBUG[85170][C-000006f9] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4e80002c00' [Dec 9 19:36:06] VERBOSE[85170][C-000006f9] chan_sip.c: Scheduling destruction of SIP dialog '1197569442-5060-26@BJC.BGI.B.G' in 6400 ms (Method: BYE) [Dec 9 19:36:06] DEBUG[85170][C-000006f9] chan_sip.c: Received bye, issuing owner hangup [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: RTPAUDIOQOS Value: ssrc=1963630971;themssrc=895943824;lp=0;rxjitter=0.000000;rxcount=565;txjitter=0.002300;txcount=765;rlp=0;rtt=0.000000 [Dec 9 19:36:06] VERBOSE[85170][C-000006f9] chan_sip.c: <--- Transmitting (NAT) to 72.190.32.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK858553058;received=72.190.32.250;rport=5060 From: "Local1" ;tag=1110989379 To: ;tag=as4325b685 Call-ID: 1197569442-5060-26@BJC.BGI.B.G CSeq: 231 BYE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Dec 9 19:36:06] DEBUG[85170][C-000006f9] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Session timer stopped: 37 - 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=1963630971;themssrc=895943824;lp=0;rxjitter=0.000000;rxcount=565;txjitter=0.002300;txcount=765;rlp=0;rtt=0.000000 [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; [Dec 9 19:36:06] DEBUG[114537][C-000006f9] bridge_channel.c: Setting 0x7f4e7c013020(SIP/100-0000006d) state from:0 to:1 [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; [Dec 9 19:36:06] DEBUG[114537][C-000006f9] bridge_channel.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: pulling 0x7f4e7c013020(SIP/100-0000006d) [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; [Dec 9 19:36:06] VERBOSE[114537][C-000006f9] bridge_channel.c: Channel SIP/100-0000006d left 'native_rtp' basic-bridge <22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba> [Dec 9 19:36:06] DEBUG[114537][C-000006f9] bridge_channel.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: 0x7f4e7c013020(SIP/100-0000006d) is leaving native_rtp technology [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; [Dec 9 19:36:06] DEBUG[114537][C-000006f9] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/100-0000006d' and 'SIP/flowroute-0000006e' - media will flow through Asterisk core [Dec 9 19:36:06] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: dissolving bridge with cause 16(Normal Clearing) [Dec 9 19:36:06] DEBUG[114537][C-000006f9] bridge_channel.c: Setting 0x7f4e7c01bfa0(SIP/flowroute-0000006e) state from:0 to:2 [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOS Value: ssrc=41559051;themssrc=3544468715;lp=0;rxjitter=0.000000;rxcount=581;txjitter=0.000032;txcount=565;rlp=0;rtt=0.000000 [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=41559051;themssrc=3544468715;lp=0;rxjitter=0.000000;rxcount=581;txjitter=0.000032;txcount=565;rlp=0;rtt=0.000000 [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; [Dec 9 19:36:06] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: queueing action type:13 sub:1001 [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; [Dec 9 19:36:06] DEBUG[114537][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba is dissolved, not performing smart bridge operation. [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; [Dec 9 19:36:06] DEBUG[114595][C-000006f9] bridge_channel.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: pulling 0x7f4e7c01bfa0(SIP/flowroute-0000006e) [Dec 9 19:36:06] VERBOSE[114595][C-000006f9] bridge_channel.c: Channel SIP/flowroute-0000006e left 'native_rtp' basic-bridge <22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba> [Dec 9 19:36:06] DEBUG[114595][C-000006f9] bridge_channel.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: 0x7f4e7c01bfa0(SIP/flowroute-0000006e) is leaving native_rtp technology [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; [Dec 9 19:36:06] DEBUG[114595][C-000006f9] bridge_channel.c: Channel SIP/flowroute-0000006e will survive this bridge; clearing outgoing (dialed) flag [Dec 9 19:36:06] DEBUG[114595][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba is dissolved, not performing smart bridge operation. [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; [Dec 9 19:36:06] DEBUG[114595][C-000006f9] res_rtp_asterisk.c: Changing ssrc from 41559051 to 2104103439 due to a source change [Dec 9 19:36:06] DEBUG[114537][C-000006f9] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; [Dec 9 19:36:06] DEBUG[114537][C-000006f9] pbx.c: Spawn extension (dial-out-100,_1XXXXXX.,9) exited non-zero on 'SIP/100-0000006d' [Dec 9 19:36:06] VERBOSE[114537][C-000006f9] pbx.c: Spawn extension (dial-out-100, _1XXXXXX., 9) exited non-zero on 'SIP/100-0000006d' [Dec 9 19:36:06] DEBUG[114537][C-000006f9] channel.c: Soft-Hanging (0x10) up channel 'SIP/100-0000006d' [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: HangupRequest Privilege: call,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 [Dec 9 19:36:06] DEBUG[114595][C-000006f9] channel.c: Hanging up channel 'SIP/flowroute-0000006e' [Dec 9 19:36:06] DEBUG[114595][C-000006f9] chan_sip.c: Hangup call SIP/flowroute-0000006e, SIP callid 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:36:06] DEBUG[114595][C-000006f9] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4e7c00a0f0' [Dec 9 19:36:06] VERBOSE[114595][C-000006f9] chan_sip.c: Scheduling destruction of SIP dialog '4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060' in 6400 ms (Method: INVITE) [Dec 9 19:36:06] DEBUG[114537][C-000006f9] channel.c: Hanging up channel 'SIP/100-0000006d' [Dec 9 19:36:06] DEBUG[114537][C-000006f9] chan_sip.c: Hangup call SIP/100-0000006d, SIP callid 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:36:06] DEBUG[114537][C-000006f9] chan_sip.c: update_call_counter(100) - decrement call limit counter on hangup [Dec 9 19:36:06] DEBUG[114537][C-000006f9] chan_sip.c: Updating call counter for incoming call [Dec 9 19:36:06] DEBUG[114537][C-000006f9] chan_sip.c: Call from peer '100' removed from call limit 2147483647 [Dec 9 19:36:06] DEBUG[114537][C-000006f9] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4e80002c00' [Dec 9 19:36:06] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Dec 9 19:36:06] VERBOSE[114595][C-000006f9] chan_sip.c: Reliably Transmitting (NAT) to 216.115.69.144:5060: BYE sip:+18018309060@74.120.93.24:5060 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK07d76d90;rport Route: , Max-Forwards: 70 From: "Luke Apartment" ;tag=as4389a784 To: ;tag=gK08d946fd Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 13.13.1 Proxy-Authorization: Digest username="23616016", realm="sip.flowroute.com", algorithm=MD5, uri="sip:+18018309060@74.120.93.24:5060", nonce="WEsIvlhLB5IB6GWiUlO/HmNmE+ziYT3s", response="34d76a7f1534643a0965090a45b10360", qop=auth, cnonce="47b895a6", nc=00000002 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Dec 9 19:36:06] DEBUG[85149] chan_sip.c: Checking device state for peer 100 [Dec 9 19:36:06] DEBUG[114595][C-000006f9] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Dec 9 19:36:06] DEBUG[85149] devicestate.c: Changing state for SIP/100 - state 1 (Not in use) [Dec 9 19:36:06] DEBUG[114595][C-000006f9] chan_sip.c: Trying to put 'BYE sip:+18' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:36:06] DEBUG[85150] cdr.c: Finalized CDR for SIP/100-0000006d - start 1481312146.003065 answer 1481312155.193022 end 1481312166.736521 dispo ANSWERED [Dec 9 19:36:06] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - 100 [Dec 9 19:36:06] DEBUG[85149] chan_sip.c: Checking device state for peer 100 [Dec 9 19:36:06] DEBUG[85149] devicestate.c: Changing state for SIP/100 - state 1 (Not in use) [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(uniqueid) result is '1481312146.221' [Dec 9 19:36:06] DEBUG[85148][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: actually destroying basic bridge, nobody wants it anymore [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(start) result is '2016-12-09 19:35:46' [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(end) result is '2016-12-09 19:36:06' [Dec 9 19:36:06] DEBUG[85148][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: calling basic bridge destructor [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(disposition) result is 'ANSWERED' [Dec 9 19:36:06] DEBUG[85148][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: calling native_rtp technology stop [Dec 9 19:36:06] DEBUG[85148][C-000006f9] bridge.c: Bridge 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba: calling native_rtp technology destructor [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(cdr_direction) result is '2' [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(cdr_application) result is 'Outbound Dial' [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(cdr_inbound_did) result is '' [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(cdr_inbound_from_phonenumber) result is '' [Dec 9 19:36:06] DEBUG[85151] app_queue.c: Extension '100@from-internal' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(cdr_inbound_from_callerid) result is '' [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(cdr_inbound_to_extension) result is '' [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(cdr_outbound_did) result is '9726001152' [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(cdr_outbound_dialed_phonenumber) result is '18018309060' [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(cdr_outbound_from_extension) result is '100' [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(cdr_internal_from_extension) result is '' [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(cdr_internal_to_extension) result is '' [Dec 9 19:36:06] DEBUG[85150] pbx_variables.c: Function CDR(cdr_recording_path) result is '' [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Session timer stopped: 10 - 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:36:06] DEBUG[85210] app_queue.c: Device 'SIP/100' changed to state '1' (Not in use) [Dec 9 19:36:06] DEBUG[85210] app_queue.c: Device 'Queue:Technical Support_avail' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 9 19:36:06] DEBUG[85149] devicestate.c: No provider found, checking channel drivers for SIP - flowroute [Dec 9 19:36:06] DEBUG[85149] chan_sip.c: Checking device state for peer flowroute [Dec 9 19:36:06] DEBUG[85149] devicestate.c: Changing state for SIP/flowroute - state 1 (Not in use) [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: BRIDGEPEER Value: [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: BRIDGEPVTCALLID Value: [Dec 9 19:36:06] DEBUG[85150] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: BRIDGEPEER Value: [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: BRIDGEPVTCALLID Value: [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: BridgeLeave Privilege: call,all BridgeUniqueid: 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba BridgeType: basic BridgeTechnology: native_rtp BridgeCreator: BridgeName: BridgeNumChannels: 1 BridgeVideoSourceMode: none Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 [Dec 9 19:36:06] DEBUG[85150] res_config_sqlite.c: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('"Luke Apartment" <9726001152>','9726001152','_1XXXXXX.','dial-out-100','SIP/100-0000006d','SIP/flowroute-0000006e','Dial','SIP/flowroute/18018309060','2016-12-09 19:35:46','2016-12-09 19:35:55','2016-12-09 19:36:06','20','11','ANSWERED','DOCUMENTATION','1481312146.221') [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: ANSWEREDTIME Value: 11 [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: BridgeLeave Privilege: call,all BridgeUniqueid: 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba BridgeType: basic BridgeTechnology: native_rtp BridgeCreator: BridgeName: BridgeNumChannels: 0 BridgeVideoSourceMode: none Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: DIALEDTIME Value: 20 [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Variable: DIALSTATUS Value: ANSWER [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: SoftHangupRequest Privilege: call,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Cause: 16 [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOS Value: ssrc=2104103439;themssrc=3544468715;lp=0;rxjitter=0.000000;rxcount=581;txjitter=0.000032;txcount=565;rlp=0;rtt=0.000000 [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: Hangup Privilege: call,all Channel: SIP/100-0000006d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 9726001152 CallerIDName: Luke Apartment ConnectedLineNum: ConnectedLineName: Language: en AccountCode: Context: dial-out-100 Exten: _1XXXXXX. Priority: 9 Uniqueid: 1481312146.221 Linkedid: 1481312146.221 Cause: 16 Cause-txt: Normal Clearing [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Variable: RTPAUDIOQOS Value: ssrc=2104103439;themssrc=3544468715;lp=0;rxjitter=0.000000;rxcount=581;txjitter=0.000032;txcount=565;rlp=0;rtt=0.000000 [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: from-internal Hint: SIP/100 Status: 0 StatusText: Idle [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: Cdr Privilege: cdr,all AccountCode: Source: 9726001152 Destination: _1XXXXXX. DestinationContext: dial-out-100 CallerID: "Luke Apartment" <9726001152> Channel: SIP/100-0000006d DestinationChannel: SIP/flowroute-0000006e LastApplication: Dial LastData: SIP/flowroute/18018309060 StartTime: 2016-12-09 19:35:46 AnswerTime: 2016-12-09 19:35:55 EndTime: 2016-12-09 19:36:06 Duration: 20 BillableSeconds: 11 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1481312146.221 UserField: unique_id: 1481312146.221 tstamp_start: 2016-12-09 19:35:46 tstamp_end: 2016-12-09 19:36:06 answer_status: ANSWERED direction: 2 application: Outbound Dial inbound_did: inbound_from_phonenumber: inbound_from_callerid: inbound_to_extension: outbound_did: 9726001152 outbound_dialed_phonenumber: 18018309060 outbound_from_extension: 100 internal_from_extension: internal_to_extension: recording_path: [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: DeviceStateChange Privilege: call,all Device: SIP/100 State: NOT_INUSE [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: Hangup Privilege: call,all Channel: SIP/flowroute-0000006e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: _1XXXXXX. CallerIDName: ConnectedLineNum: 9726001152 ConnectedLineName: Luke Apartment Language: en AccountCode: Context: from-trunk Exten: Priority: 1 Uniqueid: 1481312146.222 Linkedid: 1481312146.221 Cause: 16 Cause-txt: Normal Clearing [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: BridgeDestroy Privilege: call,all BridgeUniqueid: 22c8bb19-61a4-4d54-b0a1-5e1bbabf69ba BridgeType: basic BridgeTechnology: native_rtp BridgeCreator: BridgeName: BridgeNumChannels: 0 BridgeVideoSourceMode: none [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: QueueMemberStatus Privilege: agent,all Ringinuse: 0 StateInterface: SIP/100 CallsTaken: 0 Queue: Technical Support Membership: static Status: 1 MemberName: SIP/100 Penalty: 0 Interface: SIP/100 LastCall: 0 InCall: 0 Paused: 0 PausedReason: [Dec 9 19:36:06] DEBUG[94779] manager.c: Examining AMI event: Event: DeviceStateChange Privilege: call,all Device: Queue:Technical Support_avail State: NOT_INUSE [Dec 9 19:36:06] DEBUG[85150] cdr.c: CDR for SIP/flowroute-0000006e is dialed and has no Party B; discarding [Dec 9 19:36:06] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:216.115.69.144:5060 ---> SIP/2.0 200 OK From: "Luke Apartment" ;tag=as4389a784 To: ;tag=gK08d946fd Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK07d76d90;rport=5060 Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 CSeq: 104 BYE Record-Route: Record-Route: Content-Length: 0 <-------------> [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 1 [ 67]: From: "Luke Apartment" ;tag=as4389a784 [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 2 [ 54]: To: ;tag=gK08d946fd [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 3 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK07d76d90;rport=5060 [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 6 [ 42]: Record-Route: [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 7 [ 42]: Record-Route: [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Dec 9 19:36:06] VERBOSE[85170] chan_sip.c: --- (9 headers 0 lines) --- [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: = Looking for Call ID: 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 (Checking To) --From tag as4389a784 --To-tag gK08d946fd [Dec 9 19:36:06] DEBUG[85170][C-000006f9] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #4 [Dec 9 19:36:06] DEBUG[85170][C-000006f9] chan_sip.c: Stopping retransmission on '4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060' of Request 104: Match Found [Dec 9 19:36:06] DEBUG[85170] chan_sip.c: Destroying SIP dialog 4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060 [Dec 9 19:36:06] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '4c6165662bce96383d38c24c0a6ab8de@207.91.156.217:5060' Method: INVITE [Dec 9 19:36:06] DEBUG[85170] rtp_engine.c: Destroyed RTP instance '0x7f4e7c00a0f0' [Dec 9 19:36:10] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 0a5398971b428e4348953a9950bb4efd@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:36:11] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: SIP call-id changed from '0a5398971b428e4348953a9950bb4efd@207.91.156.217:5060' to '69cab6362fecd64d074a64c37625fb71@207.91.156.217:5060' [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 69cab6362fecd64d074a64c37625fb71@207.91.156.217:5060 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 0 [ 44]: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3bca0122;rport [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as5f824a9f [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 4 [ 34]: To: [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 69cab6362fecd64d074a64c37625fb71@207.91.156.217:5060 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:36:11 GMT [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:36:11] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 97.77.119.234:47976: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3bca0122;rport Max-Forwards: 70 From: "asterisk" ;tag=as5f824a9f To: Contact: Call-ID: 69cab6362fecd64d074a64c37625fb71@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:36:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #3 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:36:11] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3bca0122;rport=5060 From: "asterisk" ;tag=as5f824a9f To: ;tag=1298517433 Call-ID: 69cab6362fecd64d074a64c37625fb71@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3bca0122;rport=5060 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as5f824a9f [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 3 [ 49]: To: ;tag=1298517433 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 69cab6362fecd64d074a64c37625fb71@207.91.156.217:5060 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:36:11] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: = Looking for Call ID: 69cab6362fecd64d074a64c37625fb71@207.91.156.217:5060 (Checking To) --From tag as5f824a9f --To-tag 1298517433 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #3 [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Stopping retransmission on '69cab6362fecd64d074a64c37625fb71@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:36:11] DEBUG[85170] chan_sip.c: Destroying SIP dialog 69cab6362fecd64d074a64c37625fb71@207.91.156.217:5060 [Dec 9 19:36:11] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '69cab6362fecd64d074a64c37625fb71@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:36:13] DEBUG[85170] chan_sip.c: Auto destroying SIP dialog '1197569442-5060-26@BJC.BGI.B.G' [Dec 9 19:36:13] DEBUG[85170] chan_sip.c: Destroying SIP dialog 1197569442-5060-26@BJC.BGI.B.G [Dec 9 19:36:13] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '1197569442-5060-26@BJC.BGI.B.G' Method: BYE [Dec 9 19:36:13] DEBUG[85170] rtp_engine.c: Destroyed RTP instance '0x7f4e80002c00' [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 7fae6d665e3ac8b264e3072937eb5212@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:36:19] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: SIP call-id changed from '7fae6d665e3ac8b264e3072937eb5212@207.91.156.217:5060' to '328794e90a9d03fe18f302a11344c347@207.91.156.217:5060' [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 328794e90a9d03fe18f302a11344c347@207.91.156.217:5060 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 0 [ 40]: OPTIONS sip:100@192.168.1.6:5060 SIP/2.0 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK7d1ccda1;rport [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as2587c059 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 4 [ 30]: To: [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 328794e90a9d03fe18f302a11344c347@207.91.156.217:5060 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:36:19 GMT [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:36:19] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 72.190.32.250:5060: OPTIONS sip:100@192.168.1.6:5060 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK7d1ccda1;rport Max-Forwards: 70 From: "asterisk" ;tag=as2587c059 To: Contact: Call-ID: 328794e90a9d03fe18f302a11344c347@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:36:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:36:19] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK7d1ccda1;rport=5060 From: "asterisk" ;tag=as2587c059 To: ;tag=1645774908 Call-ID: 328794e90a9d03fe18f302a11344c347@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK7d1ccda1;rport=5060 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as2587c059 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 3 [ 45]: To: ;tag=1645774908 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 328794e90a9d03fe18f302a11344c347@207.91.156.217:5060 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:36:19] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: = Looking for Call ID: 328794e90a9d03fe18f302a11344c347@207.91.156.217:5060 (Checking To) --From tag as2587c059 --To-tag 1645774908 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #39 [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Stopping retransmission on '328794e90a9d03fe18f302a11344c347@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:36:19] DEBUG[85170] chan_sip.c: Destroying SIP dialog 328794e90a9d03fe18f302a11344c347@207.91.156.217:5060 [Dec 9 19:36:19] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '328794e90a9d03fe18f302a11344c347@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:36:20] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:72.190.32.250:5060 ---> <-------------> [Dec 9 19:36:23] DEBUG[94779] manager.c: Running action 'SIPpeerstatus' [Dec 9 19:36:24] DEBUG[85184] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Dec 9 19:36:25] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> <-------------> [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 3f068c92325dfed523b257da4259d979@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 9 19:36:26] DEBUG[85170] acl.c: For destination '97.77.119.234', our source address is '207.91.156.217'. [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: SIP call-id changed from '3f068c92325dfed523b257da4259d979@207.91.156.217:5060' to '3aea622e1c5fec896c649796524dbbb7@207.91.156.217:5060' [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 3aea622e1c5fec896c649796524dbbb7@207.91.156.217:5060 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 0 [ 44]: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3741b6c0;rport [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 3 [ 61]: From: "asterisk" ;tag=as7377f647 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 4 [ 34]: To: [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 5 [ 43]: Contact: [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 6 [ 61]: Call-ID: 3aea622e1c5fec896c649796524dbbb7@207.91.156.217:5060 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 13.13.1 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 9 [ 35]: Date: Fri, 09 Dec 2016 19:36:26 GMT [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 9 19:36:26] VERBOSE[85170] chan_sip.c: Reliably Transmitting (NAT) to 97.77.119.234:47976: OPTIONS sip:1033@192.168.2.247:47976 SIP/2.0 Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3741b6c0;rport Max-Forwards: 70 From: "asterisk" ;tag=as7377f647 To: Contact: Call-ID: 3aea622e1c5fec896c649796524dbbb7@207.91.156.217:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.13.1 Date: Fri, 09 Dec 2016 19:36:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #43 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 97.77.119.234:47976 [Dec 9 19:36:26] VERBOSE[85170] chan_sip.c: <--- SIP read from UDP:97.77.119.234:47976 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3741b6c0;rport=5060 From: "asterisk" ;tag=as7377f647 To: ;tag=8347425 Call-ID: 3aea622e1c5fec896c649796524dbbb7@207.91.156.217:5060 CSeq: 102 OPTIONS Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.7.97 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 207.91.156.217:5060;branch=z9hG4bK3741b6c0;rport=5060 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 2 [ 61]: From: "asterisk" ;tag=as7377f647 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 3 [ 46]: To: ;tag=8347425 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 4 [ 61]: Call-ID: 3aea622e1c5fec896c649796524dbbb7@207.91.156.217:5060 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 6 [ 32]: Supported: replaces, path, timer [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 7 [ 40]: User-Agent: Grandstream GXP2140 1.0.7.97 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 8 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 9 19:36:26] VERBOSE[85170] chan_sip.c: --- (10 headers 0 lines) --- [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: = Looking for Call ID: 3aea622e1c5fec896c649796524dbbb7@207.91.156.217:5060 (Checking To) --From tag as7377f647 --To-tag 8347425 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #43 [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Stopping retransmission on '3aea622e1c5fec896c649796524dbbb7@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:36:26] DEBUG[85170] chan_sip.c: Destroying SIP dialog 3aea622e1c5fec896c649796524dbbb7@207.91.156.217:5060 [Dec 9 19:36:26] VERBOSE[85170] chan_sip.c: Really destroying SIP dialog '3aea622e1c5fec896c649796524dbbb7@207.91.156.217:5060' Method: OPTIONS [Dec 9 19:36:26] DEBUG[114594] threadpool.c: Worker thread idle timeout reached. Dying. [Dec 9 19:36:26] DEBUG[85138] threadpool.c: Destroying worker thread 589 [Dec 9 19:36:26] DEBUG[114596] threadpool.c: Worker thread idle timeout reached. Dying. [Dec 9 19:36:26] DEBUG[85138] threadpool.c: Destroying worker thread 590 [Dec 9 19:36:26] DEBUG[114313] threadpool.c: Worker thread idle timeout reached. Dying. [Dec 9 19:36:26] DEBUG[85138] threadpool.c: Destroying worker thread 587 [Dec 9 19:36:33] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 6d0a0aa1222788a67a2cd1d17d22a9f3@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:36:33] DEBUG[85170] acl.c: For destination '216.115.69.144', our source address is '207.91.156.217'. [Dec 9 19:36:33] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:36:33] DEBUG[85170] chan_sip.c: SIP call-id changed from '6d0a0aa1222788a67a2cd1d17d22a9f3@207.91.156.217:5060' to '5294c0a536a9840732c1c54106344154@207.91.156.217:5060' [Dec 9 19:36:33] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 5294c0a536a9840732c1c54106344154@207.91.156.217:5060 [Dec 9 19:36:33] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 216.115.69.144:5060 [Dec 9 19:36:33] DEBUG[85170] chan_sip.c: = Looking for Call ID: 5294c0a536a9840732c1c54106344154@207.91.156.217:5060 (Checking To) --From tag as6f9edf2d --To-tag 20e1698a3241cbcc6677a39fcb65a0aa.eacd [Dec 9 19:36:33] DEBUG[85170] chan_sip.c: Stopping retransmission on '5294c0a536a9840732c1c54106344154@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:36:33] DEBUG[85170] chan_sip.c: Destroying SIP dialog 5294c0a536a9840732c1c54106344154@207.91.156.217:5060 [Dec 9 19:36:34] DEBUG[85170] chan_sip.c: Allocating new SIP dialog for 0e0caf862cb6f8615a2fc37d4ec86c2c@207.91.156.217:5060 - OPTIONS (No RTP) [Dec 9 19:36:34] DEBUG[85170] acl.c: For destination '72.190.32.250', our source address is '207.91.156.217'. [Dec 9 19:36:34] DEBUG[85170] chan_sip.c: Setting AST_TRANSPORT_UDP with address 207.91.156.217:5060 [Dec 9 19:36:34] DEBUG[85170] chan_sip.c: SIP call-id changed from '0e0caf862cb6f8615a2fc37d4ec86c2c@207.91.156.217:5060' to '2dc1433526942adc410ecffa5cc9ebed@207.91.156.217:5060' [Dec 9 19:36:34] DEBUG[85170] chan_sip.c: Initializing initreq for method OPTIONS - callid 2dc1433526942adc410ecffa5cc9ebed@207.91.156.217:5060 [Dec 9 19:36:34] DEBUG[85170] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 72.190.32.250:5060 [Dec 9 19:36:34] DEBUG[85170] chan_sip.c: = Looking for Call ID: 2dc1433526942adc410ecffa5cc9ebed@207.91.156.217:5060 (Checking To) --From tag as4e3454c7 --To-tag 1936294800 [Dec 9 19:36:34] DEBUG[85170] chan_sip.c: Stopping retransmission on '2dc1433526942adc410ecffa5cc9ebed@207.91.156.217:5060' of Request 102: Match Found [Dec 9 19:36:34] DEBUG[85170] chan_sip.c: Destroying SIP dialog 2dc1433526942adc410ecffa5cc9ebed@207.91.156.217:5060 [Dec 9 19:36:36] Asterisk 13.13.1 built by root @ testbed2.primevox.net on a x86_64 running Linux on 2016-12-09 00:01:26 UTC [Dec 9 19:36:36] VERBOSE[113564] config.c: Parsing '/etc/asterisk/logger.conf': Found [Dec 9 19:36:38] VERBOSE[113564] asterisk.c: Remote UNIX connection disconnected