[May 27 17:35:33] DEBUG[28657]: chan_sip.c:4412 __sip_autodestruct: Auto destroying SIP dialog '1295208163' [May 27 17:35:33] DEBUG[28657]: chan_sip.c:6817 sip_destroy: Destroying SIP dialog 1295208163 Really destroying SIP dialog '1295208163' Method: REGISTER <--- SIP read from UDP:78.26.144.10:5060 ---> INVITE sip:6000333@209.239.114.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1272360582 From: ;tag=896355678 To: "Doctor" Call-ID: 1500096782 CSeq: 20 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Subject: Phone call Content-Length: 230 v=0 o=6000444 1786 3812 IN IP4 78.26.144.10 s=Talk c=IN IP4 78.26.144.10 b=AS:380 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7077 IN IP4 192.168.1.107 <-------------> [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 41]: INVITE sip:6000333@209.239.114.51 SIP/2.0 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1272360582 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 55]: From: ;tag=896355678 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 41]: To: "Doctor" [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 15]: CSeq: 20 INVITE [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 44]: Contact: [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 29]: Content-Type: application/sdp [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 16]: Max-Forwards: 70 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 11 [ 19]: Subject: Phone call [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 12 [ 19]: Content-Length: 230 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 13 [ 0]: [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000444 1786 3812 IN IP4 78.26.144.10 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 3 [ 21]: c=IN IP4 78.26.144.10 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 6 [ 26]: m=audio 7076 RTP/AVP 9 101 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 7 [ 20]: a=rtpmap:9 G722/8000 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 9 [ 15]: a=fmtp:101 0-11 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9605 parse_request: Body 10 [ 32]: a=rtcp:7077 IN IP4 192.168.1.107 --- (13 headers 11 lines) --- [May 27 17:35:36] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking From) --From tag 896355678 --To-tag [May 27 17:35:36] DEBUG[28657]: logger.c:1294 ast_create_callid: CALL_ID [C-0000019a] created by thread. [May 27 17:35:36] DEBUG[28657]: acl.c:979 ast_ouraddrfor: For destination '78.26.144.10', our source address is '209.239.114.51'. [May 27 17:35:36] DEBUG[28657]: chan_sip.c:4021 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 209.239.114.51:5060 [May 27 17:35:36] DEBUG[28657]: chan_sip.c:8721 sip_alloc: Allocating new SIP dialog for 1500096782 - INVITE (No RTP) [May 27 17:35:36] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:36] DEBUG[28657][C-0000019a]: chan_sip.c:27866 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [May 27 17:35:36] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107:5060' into... [May 27 17:35:36] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port '5060'. [May 27 17:35:36] DEBUG[28657][C-0000019a]: chan_sip.c:17845 check_via: NAT detected for 192.168.1.107:5060 / 78.26.144.10:5060 Sending to 78.26.144.10:5060 (NAT) [May 27 17:35:36] DEBUG[28657][C-0000019a]: chan_sip.c:24999 handle_request_invite: Initializing initreq for method INVITE - callid 1500096782 Using INVITE request as basis request - 1500096782 [May 27 17:35:36] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:40412' into... [May 27 17:35:36] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port ''. Found peer '6000444' for '6000444' from 78.26.144.10:5060 <--- Reliably Transmitting (NAT) to 78.26.144.10:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK1272360582;received=78.26.144.10;rport=5060 From: ;tag=896355678 To: "Doctor" ;tag=as6f611f91 Call-ID: 1500096782 CSeq: 20 INVITE Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="vm.intersog.com", nonce="7b87fadb" Content-Length: 0 <------------> [May 27 17:35:36] DEBUG[28657][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432319 [May 27 17:35:36] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 78.26.144.10:5060 Scheduling destruction of SIP dialog '1500096782' in 32000 ms (Method: INVITE) [May 27 17:35:36] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:78.26.144.10:5060 ---> ACK sip:6000333@209.239.114.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1272360582 From: ;tag=896355678 To: "Doctor" ;tag=as6f611f91 Call-ID: 1500096782 CSeq: 20 ACK Content-Length: 0 <-------------> [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 38]: ACK sip:6000333@209.239.114.51 SIP/2.0 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1272360582 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 55]: From: ;tag=896355678 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 56]: To: "Doctor" ;tag=as6f611f91 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 12]: CSeq: 20 ACK [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 17]: Content-Length: 0 --- (7 headers 0 lines) --- [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking From) --From tag 896355678 --To-tag as6f611f91 [May 27 17:35:37] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:27866 handle_incoming: **** Received ACK (6) - Command in SIP ACK [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:4523 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #432319 [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '1500096782' of Response 20: Match Found [May 27 17:35:37] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:78.26.144.10:5060 ---> INVITE sip:6000333@209.239.114.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK594604029 From: ;tag=896355678 To: "Doctor" Call-ID: 1500096782 CSeq: 21 INVITE Contact: Authorization: Digest username="6000444", realm="vm.intersog.com", nonce="7b87fadb", uri="sip:6000333@209.239.114.51", response="e8069e52f42913de65a89af6331afbf9", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Subject: Phone call Content-Length: 230 v=0 o=6000444 1786 3812 IN IP4 78.26.144.10 s=Talk c=IN IP4 78.26.144.10 b=AS:380 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7077 IN IP4 192.168.1.107 <-------------> [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 41]: INVITE sip:6000333@209.239.114.51 SIP/2.0 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK594604029 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 55]: From: ;tag=896355678 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 41]: To: "Doctor" [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 15]: CSeq: 21 INVITE [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 44]: Contact: [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [177]: Authorization: Digest username="6000444", realm="vm.intersog.com", nonce="7b87fadb", uri="sip:6000333@209.239.114.51", response="e8069e52f42913de65a89af6331afbf9", algorithm=MD5 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 29]: Content-Type: application/sdp [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 16]: Max-Forwards: 70 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 11 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 12 [ 19]: Subject: Phone call [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 13 [ 19]: Content-Length: 230 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 14 [ 0]: [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000444 1786 3812 IN IP4 78.26.144.10 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 3 [ 21]: c=IN IP4 78.26.144.10 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 6 [ 26]: m=audio 7076 RTP/AVP 9 101 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 7 [ 20]: a=rtpmap:9 G722/8000 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 9 [ 15]: a=fmtp:101 0-11 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9605 parse_request: Body 10 [ 32]: a=rtcp:7077 IN IP4 192.168.1.107 --- (14 headers 11 lines) --- [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking From) --From tag 896355678 --To-tag [May 27 17:35:37] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:27866 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107:5060' into... [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port '5060'. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:17845 check_via: NAT detected for 192.168.1.107:5060 / 78.26.144.10:5060 Sending to 78.26.144.10:5060 (NAT) [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:24999 handle_request_invite: Initializing initreq for method INVITE - callid 1500096782 Using INVITE request as basis request - 1500096782 [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:40412' into... [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port ''. Found peer '6000444' for '6000444' from 78.26.144.10:5060 [May 27 17:35:37] DEBUG[28657][C-0000019a]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7fa8701bc498' [May 27 17:35:37] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:1732 ast_rtp_new: Allocated port 11376 for RTP instance '0x7fa8701bc498' [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[28657][C-0000019a]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x7fa8701bc498' is setup and ready to go [May 27 17:35:37] DEBUG[28657][C-0000019a]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7fa8701cbfc8' [May 27 17:35:37] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:1732 ast_rtp_new: Allocated port 13754 for RTP instance '0x7fa8701cbfc8' [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[28657][C-0000019a]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x7fa8701cbfc8' is setup and ready to go [May 27 17:35:37] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa8701cbfc8' == Using SIP VIDEO CoS mark 6 [May 27 17:35:37] DEBUG[28657][C-0000019a]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7fa870202818' [May 27 17:35:37] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:1732 ast_rtp_new: Allocated port 19710 for RTP instance '0x7fa870202818' [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[28657][C-0000019a]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x7fa870202818' is setup and ready to go [May 27 17:35:37] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa870202818' [May 27 17:35:37] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa8701bc498' == Using SIP RTP CoS mark 5 [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:5725 do_setnat: Setting NAT on RTP to On [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:5729 do_setnat: Setting NAT on VRTP to On [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:5737 do_setnat: Setting NAT on TRTP to On [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=6000444 1786 3812 IN IP4 78.26.144.10... OK. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '78.26.144.10' into... [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '78.26.144.10' and port ''. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 78.26.144.10... OK. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. Found RTP audio format 9 [May 27 17:35:37] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa8980c5630 Found RTP audio format 101 [May 27 17:35:37] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa8980c5630 Found audio description format G722 for ID 9 [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. Found audio description format telephone-event for ID 101 [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtcp:7077 IN IP4 192.168.1.107... UNSUPPORTED OR FAILED. Capabilities: us - (ulaw|g722|h264|vp8), peer - audio=(g722)/video=(nothing)/text=(nothing), combined - (g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 27 17:35:37] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa8701bc498' Peer audio RTP is at port 78.26.144.10:7076 [May 27 17:35:37] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa8980c5630 to 0x7fa8701bc660 [May 27 17:35:37] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa8980c5630 to 0x7fa8701bc660 [May 27 17:35:37] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3817 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7fa8701bc498' [May 27 17:35:37] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa8701cbfc8' Peer doesn't provide video [May 27 17:35:37] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa870202818' Peer doesn't provide T.140 [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:10668 process_sdp: We're settling with these formats: (g722) [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:25137 handle_request_invite: Checking SIP call limits for device 6000444 [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:6669 update_call_counter: Updating call counter for incoming call [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:6774 update_call_counter: Call from peer '6000444' is 1 out of 2147483647 [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:40412' into... [May 27 17:35:37] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port ''. Looking for 6000333 in webrtc (domain 209.239.114.51) [May 27 17:35:37] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000444 [May 27 17:35:37] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000444 [May 27 17:35:37] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000444 - state 2 (In use) [May 27 17:35:37] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000444' state '2' [May 27 17:35:37] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000444' changed to state '2' (In use) but we don't care because they're not a member of any queue. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:7906 sip_new: *** Our native formats are (g722) [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:7907 sip_new: *** Joint capabilities are (g722) [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:7908 sip_new: *** Our capabilities are (ulaw|g722|h264|vp8) [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:7909 sip_new: *** AST_CODEC_CHOOSE formats are g722 [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:7935 sip_new: This channel can handle video! HOLLYWOOD next! [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:16109 build_route: build_route: Contact hop: list_route: hop: [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:25447 handle_request_invite: SIP/6000444-000000af: New call is still down.... Trying... <--- Transmitting (NAT) to 78.26.144.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK594604029;received=78.26.144.10;rport=5060 From: ;tag=896355678 To: "Doctor" Call-ID: 1500096782 CSeq: 21 INVITE Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:37] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000444 [May 27 17:35:37] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000444 [May 27 17:35:37] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000444 - state 2 (In use) [May 27 17:35:37] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000444' state '2' [May 27 17:35:37] DEBUG[20587][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:37] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. [May 27 17:35:37] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000444' changed to state '2' (In use) but we don't care because they're not a member of any queue. [May 27 17:35:37] DEBUG[20587][C-0000019a]: pbx.c:3642 ast_str_retrieve_variable: Result of 'EXTEN' is '6000333' [May 27 17:35:37] DEBUG[20587][C-0000019a]: pbx.c:4633 pbx_extension_helper: Launching 'NoOp' -- Executing [6000333@webrtc:1] NoOp("SIP/6000444-000000af", "Call Start to: 6000333") in new stack [May 27 17:35:37] DEBUG[20587][C-0000019a]: pbx.c:3642 ast_str_retrieve_variable: Result of 'EXTEN' is '6000333' [May 27 17:35:37] DEBUG[20587][C-0000019a]: pbx.c:4633 pbx_extension_helper: Launching 'Dial' -- Executing [6000333@webrtc:2] Dial("SIP/6000444-000000af", "SIP/6000333,20") in new stack [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:29402 sip_request_call: Asked to create a SIP channel with formats: (g722) [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:8721 sip_alloc: Allocating new SIP dialog for 0d1b688d77d1accb404d447869c85352@209.239.114.51:5060 - INVITE (No RTP) [May 27 17:35:37] DEBUG[20587][C-0000019a]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7fa8740264e8' [May 27 17:35:37] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:1732 ast_rtp_new: Allocated port 13830 for RTP instance '0x7fa8740264e8' [May 27 17:35:37] DEBUG[20587][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[20587][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[20587][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[20587][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[20587][C-0000019a]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x7fa8740264e8' is setup and ready to go [May 27 17:35:37] DEBUG[20587][C-0000019a]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7fa87402aa88' [May 27 17:35:37] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:1732 ast_rtp_new: Allocated port 13112 for RTP instance '0x7fa87402aa88' [May 27 17:35:37] DEBUG[20587][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[20587][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[20587][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[20587][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[20587][C-0000019a]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x7fa87402aa88' is setup and ready to go [May 27 17:35:37] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa87402aa88' == Using SIP VIDEO CoS mark 6 [May 27 17:35:37] DEBUG[20587][C-0000019a]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7fa874063918' [May 27 17:35:37] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:1732 ast_rtp_new: Allocated port 13138 for RTP instance '0x7fa874063918' [May 27 17:35:37] DEBUG[20587][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[20587][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[20587][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into... [May 27 17:35:37] DEBUG[20587][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''. [May 27 17:35:37] DEBUG[20587][C-0000019a]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x7fa874063918' is setup and ready to go [May 27 17:35:37] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa874063918' [May 27 17:35:37] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa8740264e8' == Using SIP RTP CoS mark 5 [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:5725 do_setnat: Setting NAT on RTP to On [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:5729 do_setnat: Setting NAT on VRTP to On [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:5737 do_setnat: Setting NAT on TRTP to On [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:3631 obproxy_get: OBPROXY: Not applying OBproxy to this call [May 27 17:35:37] DEBUG[20587][C-0000019a]: acl.c:979 ast_ouraddrfor: For destination '89.209.100.91', our source address is '209.239.114.51'. [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:4021 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 209.239.114.51:5060 [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:7906 sip_new: *** Our native formats are (g722) [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:7907 sip_new: *** Joint capabilities are (g722) [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:7908 sip_new: *** Our capabilities are (ulaw|g722|h264|vp8) [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:7909 sip_new: *** AST_CODEC_CHOOSE formats are g722 [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:7911 sip_new: *** Our preferred formats from the incoming channel are (g722) [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:7935 sip_new: This channel can handle video! HOLLYWOOD next! [May 27 17:35:37] DEBUG[20587][C-0000019a]: channel_internal_api.c:882 ast_channel_callid_set: Channel Call ID changing from [C-0000019a] to [C-0000019a] [May 27 17:35:37] DEBUG[20587][C-0000019a]: rtp_engine.c:1685 ast_rtp_instance_early_bridge_make_compatible: Seeded SDP of 'SIP/6000333-000000b0' with that of 'SIP/6000444-000000af' [May 27 17:35:37] DEBUG[20587][C-0000019a]: channel.c:6501 ast_channel_inherit_variables: Not copying variable DIALEDTIME. [May 27 17:35:37] DEBUG[20587][C-0000019a]: channel.c:6501 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. [May 27 17:35:37] DEBUG[20587][C-0000019a]: channel.c:6501 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. [May 27 17:35:37] DEBUG[20587][C-0000019a]: channel.c:6501 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [May 27 17:35:37] DEBUG[20587][C-0000019a]: channel.c:6501 ast_channel_inherit_variables: Not copying variable DIALSTATUS. [May 27 17:35:37] DEBUG[20587][C-0000019a]: channel.c:6501 ast_channel_inherit_variables: Not copying variable SIPCALLID. [May 27 17:35:37] DEBUG[20587][C-0000019a]: channel.c:6501 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [May 27 17:35:37] DEBUG[20587][C-0000019a]: channel.c:6501 ast_channel_inherit_variables: Not copying variable SIPURI. [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:6345 sip_call: Outgoing Call for 6000333 [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:6669 update_call_counter: Updating call counter for outgoing call [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:6774 update_call_counter: Call to peer '6000333' is 1 out of 2147483647 [May 27 17:35:37] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000333 [May 27 17:35:37] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000333 [May 27 17:35:37] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000333 - state 6 (Ringing) [May 27 17:35:37] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000333' state '6' [May 27 17:35:37] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000333' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:13004 add_sdp: This call needs video offers! [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:13054 add_sdp: ** Our capability: (ulaw|g722|h264|vp8) Video flag: False Text flag: False [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:13055 add_sdp: ** Our prefcodec: (g722) Audio is at 13830 Video is at 209.239.114.51:13112 Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding video codec 200006 (vp8) to SDP Adding video codec 200004 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:13192 add_sdp: -- Done with adding codecs to SDP [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:13390 add_sdp: Done building SDP. Settling with this capability: (ulaw|g722|h264|vp8) [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:3507 initialize_initreq: Initializing initreq for method INVITE - callid 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 0 [ 61]: INVITE sip:6000333@89.209.100.91;line=740fe18218f43a4 SIP/2.0 [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK2fa3e871;rport [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 2 [ 16]: Max-Forwards: 70 [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 3 [ 59]: From: "Patient" ;tag=as3d2643ca [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 4 [ 52]: To: [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 5 [ 42]: Contact: [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 6 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 7 [ 16]: CSeq: 102 INVITE [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 8 [ 30]: User-Agent: Video Medicine PBX [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 9 [ 35]: Date: Mon, 27 May 2013 17:35:37 GMT [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 11 [ 26]: Supported: replaces, timer [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 12 [ 29]: Content-Type: application/sdp Reliably Transmitting (NAT) to 89.209.100.91:5060: INVITE sip:6000333@89.209.100.91;line=740fe18218f43a4 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK2fa3e871;rport Max-Forwards: 70 From: "Patient" ;tag=as3d2643ca To: Contact: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 102 INVITE User-Agent: Video Medicine PBX Date: Mon, 27 May 2013 17:35:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 482 v=0 o=root 209909186 209909186 IN IP4 209.239.114.51 s=Asterisk PBX 11.2.1 c=IN IP4 209.239.114.51 b=CT:384 t=0 0 m=audio 13830 RTP/AVP 9 0 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 13112 RTP/AVP 100 99 a=rtpmap:100 VP8/90000 a=rtpmap:99 H264/90000 a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0 a=sendrecv --- [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432322 [May 27 17:35:37] DEBUG[20587][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 89.209.100.91:5060 -- Called SIP/6000333 <--- SIP read from UDP:89.209.100.91:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK2fa3e871;rport=5060 From: "Patient" ;tag=as3d2643ca To: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 102 INVITE User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK2fa3e871;rport=5060 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 59]: From: "Patient" ;tag=as3d2643ca [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 52]: To: [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 17]: Content-Length: 0 --- (8 headers 0 lines) --- [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (Checking To) --From tag as3d2643ca --To-tag [May 27 17:35:37] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:4590 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #432322 - INVITE (got response) [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:4597 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' Request 102: Found [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:22344 handle_response_invite: SIP response 100 to standard invite [May 27 17:35:37] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:89.209.100.91:5060 ---> SIP/2.0 101 Dialog Establishement Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK2fa3e871;rport=5060 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 102 INVITE Contact: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 33]: SIP/2.0 101 Dialog Establishement [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK2fa3e871;rport=5060 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 59]: From: "Patient" ;tag=as3d2643ca [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 66]: To: ;tag=525491751 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 40]: Contact: [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 17]: Content-Length: 0 --- (9 headers 0 lines) --- [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (Checking To) --From tag as3d2643ca --To-tag 525491751 [May 27 17:35:37] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:4597 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' Request 102: Found [May 27 17:35:37] DEBUG[28657][C-0000019a]: chan_sip.c:22344 handle_response_invite: SIP response 101 to standard invite [May 27 17:35:37] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:89.209.100.91:5060 ---> jaK <-------------> [May 27 17:35:37] DEBUG[28657]: chan_sip.c:9605 parse_request: Header 0 [ 3]: jaK <--- SIP read from UDP:89.209.100.91:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK2fa3e871;rport=5060 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 102 INVITE Contact: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:38] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 19]: SIP/2.0 180 Ringing [May 27 17:35:38] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK2fa3e871;rport=5060 [May 27 17:35:38] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 59]: From: "Patient" ;tag=as3d2643ca [May 27 17:35:38] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 66]: To: ;tag=525491751 [May 27 17:35:38] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:38] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [May 27 17:35:38] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 40]: Contact: [May 27 17:35:38] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:38] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 17]: Content-Length: 0 --- (9 headers 0 lines) --- [May 27 17:35:38] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (Checking To) --From tag as3d2643ca --To-tag 525491751 [May 27 17:35:38] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:38] DEBUG[28657][C-0000019a]: chan_sip.c:4597 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' Request 102: Found [May 27 17:35:38] DEBUG[28657][C-0000019a]: chan_sip.c:22344 handle_response_invite: SIP response 180 to standard invite [May 27 17:35:38] DEBUG[28657][C-0000019a]: chan_sip.c:16109 build_route: build_route: Contact hop: list_route: hop: [May 27 17:35:38] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. [May 27 17:35:38] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000333 [May 27 17:35:38] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000333 [May 27 17:35:38] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000333 - state 6 (Ringing) [May 27 17:35:38] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000333' state '6' -- SIP/6000333-000000b0 is ringing [May 27 17:35:38] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000333' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [May 27 17:35:38] DEBUG[20587][C-0000019a]: rtp_engine.c:1792 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/6000444-000000af' with that of 'SIP/6000333-000000b0' <--- Transmitting (NAT) to 78.26.144.10:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK594604029;received=78.26.144.10;rport=5060 From: ;tag=896355678 To: "Doctor" ;tag=as025a484a Call-ID: 1500096782 CSeq: 21 INVITE Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [May 27 17:35:38] DEBUG[20587][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 180' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:8721 sip_alloc: Allocating new SIP dialog for 2d06611a4c7945e75fdd749f2f670e33@209.239.114.51:5060 - OPTIONS (No RTP) [May 27 17:35:39] DEBUG[28657]: acl.c:979 ast_ouraddrfor: For destination '92.113.50.205', our source address is '209.239.114.51'. [May 27 17:35:39] DEBUG[28657]: chan_sip.c:4021 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 209.239.114.51:5060 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:3507 initialize_initreq: Initializing initreq for method OPTIONS - callid 61abb47f2f9188dc71e38d6d6c3e288f@209.239.114.51:5060 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 71]: OPTIONS sip:5000000248@92.113.50.205:36857;line=6666ca59cc8426b SIP/2.0 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK13cfd4af;rport [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 16]: Max-Forwards: 70 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 61]: From: "asterisk" ;tag=as503b4680 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: To: [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 43]: Contact: [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 61]: Call-ID: 61abb47f2f9188dc71e38d6d6c3e288f@209.239.114.51:5060 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 17]: CSeq: 102 OPTIONS [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 30]: User-Agent: Video Medicine PBX [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 35]: Date: Mon, 27 May 2013 17:35:39 GMT [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 11 [ 26]: Supported: replaces, timer Reliably Transmitting (NAT) to 92.113.50.205:36857: OPTIONS sip:5000000248@92.113.50.205:36857;line=6666ca59cc8426b SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK13cfd4af;rport Max-Forwards: 70 From: "asterisk" ;tag=as503b4680 To: Contact: Call-ID: 61abb47f2f9188dc71e38d6d6c3e288f@209.239.114.51:5060 CSeq: 102 OPTIONS User-Agent: Video Medicine PBX Date: Mon, 27 May 2013 17:35:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [May 27 17:35:39] DEBUG[28657]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432325 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 92.113.50.205:36857 <--- SIP read from UDP:89.209.100.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK2fa3e871;rport=5060 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 287 v=0 o=6000333 622 2564 IN IP4 89.209.100.91 s=Talk c=IN IP4 89.209.100.91 b=AS:380 t=0 0 m=audio 7076 RTP/AVP 9 0 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7077 IN IP4 192.168.3.93 m=video 0 RTP/AVP 0 a=inactive <-------------> [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK2fa3e871;rport=5060 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 59]: From: "Patient" ;tag=as3d2643ca [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 66]: To: ;tag=525491751 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 43]: Contact: [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 29]: Content-Type: application/sdp [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 19]: Content-Length: 287 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 0]: [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000333 622 2564 IN IP4 89.209.100.91 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 3 [ 22]: c=IN IP4 89.209.100.91 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 6 [ 28]: m=audio 7076 RTP/AVP 9 0 101 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 7 [ 20]: a=rtpmap:9 G722/8000 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 10 [ 15]: a=fmtp:101 0-11 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 11 [ 31]: a=rtcp:7077 IN IP4 192.168.3.93 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 12 [ 19]: m=video 0 RTP/AVP 0 [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9605 parse_request: Body 13 [ 10]: a=inactive --- (10 headers 14 lines) --- [May 27 17:35:39] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (Checking To) --From tag as3d2643ca --To-tag 525491751 [May 27 17:35:39] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:4518 __sip_ack: Acked pending invite 102 [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' of Request 102: Match Found [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:22344 handle_response_invite: SIP response 200 to standard invite [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=6000333 622 2564 IN IP4 89.209.100.91... OK. [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [May 27 17:35:39] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '89.209.100.91' into... [May 27 17:35:39] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '89.209.100.91' and port ''. [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 89.209.100.91... OK. [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED. [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. Found RTP audio format 9 [May 27 17:35:39] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa8980c4ad0 Found RTP audio format 0 [May 27 17:35:39] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7fa8980c4ad0 Found RTP audio format 101 [May 27 17:35:39] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa8980c4ad0 Found audio description format G722 for ID 9 [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. Found audio description format PCMU for ID 0 [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. Found audio description format telephone-event for ID 101 [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtcp:7077 IN IP4 192.168.3.93... UNSUPPORTED OR FAILED. [May 27 17:35:39] WARNING[28657][C-0000019a]: chan_sip.c:10131 process_sdp: Ignoring video stream offer because port number is zero Capabilities: us - (ulaw|g722|h264|vp8), peer - audio=(ulaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 27 17:35:39] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa8740264e8' Peer audio RTP is at port 89.209.100.91:7076 [May 27 17:35:39] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0x7fa8980c4ad0 to 0x7fa8740266b0 [May 27 17:35:39] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa8980c4ad0 to 0x7fa8740266b0 [May 27 17:35:39] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa8980c4ad0 to 0x7fa8740266b0 [May 27 17:35:39] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3817 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7fa8740264e8' [May 27 17:35:39] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa87402aa88' Peer doesn't provide video [May 27 17:35:39] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa874063918' Peer doesn't provide T.140 [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10668 process_sdp: We're settling with these formats: (ulaw|g722) [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10675 process_sdp: We have an owner, now see if we need to change this call [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:10681 process_sdp: Setting native formats after processing SDP. peer joint formats (ulaw|g722), old nativeformats (g722) [May 27 17:35:39] DEBUG[28657][C-0000019a]: channel.c:5355 set_format: Set channel SIP/6000333-000000b0 to read format g722 [May 27 17:35:39] DEBUG[28657][C-0000019a]: channel.c:5355 set_format: Set channel SIP/6000333-000000b0 to write format g722 [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:6669 update_call_counter: Updating call counter for outgoing call [May 27 17:35:39] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000333 [May 27 17:35:39] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000333 [May 27 17:35:39] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000333 - state 2 (In use) [May 27 17:35:39] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000333' state '2' [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:16109 build_route: build_route: Contact hop: list_route: hop: [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 set_destination: Parsing for address/port to send to [May 27 17:35:39] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.93' into... [May 27 17:35:39] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.93' and port ''. set_destination: set destination to 192.168.3.93:5060 Transmitting (NAT) to 89.209.100.91:5060: ACK sip:linphone.iphone@192.168.3.93 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK0c5a6b50;rport Max-Forwards: 70 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Contact: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 102 ACK User-Agent: Video Medicine PBX Content-Length: 0 --- [May 27 17:35:39] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'ACK sip:lin' onto UDP socket destined for 89.209.100.91:5060 [May 27 17:35:39] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000333' changed to state '2' (In use) but we don't care because they're not a member of any queue. [May 27 17:35:39] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. [May 27 17:35:39] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000333 [May 27 17:35:39] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000333 [May 27 17:35:39] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000333 - state 2 (In use) [May 27 17:35:39] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000333' state '2' [May 27 17:35:39] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000333' changed to state '2' (In use) but we don't care because they're not a member of any queue. -- SIP/6000333-000000b0 answered SIP/6000444-000000af [May 27 17:35:39] DEBUG[20587][C-0000019a]: rtp_engine.c:1792 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/6000444-000000af' with that of 'SIP/6000333-000000b0' [May 27 17:35:39] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000444 [May 27 17:35:39] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000444 [May 27 17:35:39] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000444 - state 2 (In use) [May 27 17:35:39] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000444' state '2' [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:7256 sip_answer: SIP answering channel: SIP/6000444-000000af [May 27 17:35:39] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:2085 ast_rtp_update_source: Setting the marker bit due to a source update [May 27 17:35:39] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000444' changed to state '2' (In use) but we don't care because they're not a member of any queue. [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:13504 transmit_response_with_sdp: Setting framing from config on incoming call [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:13054 add_sdp: ** Our capability: (g722) Video flag: True Text flag: True [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:13055 add_sdp: ** Our prefcodec: (nothing) Audio is at 11376 Adding codec 100012 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:13192 add_sdp: -- Done with adding codecs to SDP [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:13390 add_sdp: Done building SDP. Settling with this capability: (g722) <--- Reliably Transmitting (NAT) to 78.26.144.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK594604029;received=78.26.144.10;rport=5060 From: ;tag=896355678 To: "Doctor" ;tag=as025a484a Call-ID: 1500096782 CSeq: 21 INVITE Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 426475338 426475338 IN IP4 209.239.114.51 s=Asterisk PBX 11.2.1 c=IN IP4 209.239.114.51 t=0 0 m=audio 11376 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432327 [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:39] DEBUG[20587][C-0000019a]: features.c:4410 ast_bridge_call: bridge answer set, chan answer set [May 27 17:35:39] DEBUG[20587][C-0000019a]: features.c:4234 clear_dialed_interfaces: Removing dialed interfaces datastore on SIP/6000333-000000b0 since we're bridging [May 27 17:35:39] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:2085 ast_rtp_update_source: Setting the marker bit due to a source update [May 27 17:35:39] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:2085 ast_rtp_update_source: Setting the marker bit due to a source update -- Remotely bridging SIP/6000444-000000af and SIP/6000333-000000b0 [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:32435 sip_set_rtp_peer: Deferring reinvite on SIP '1500096782' - It's audio will be redirected to IP 89.209.100.91:7076 [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:32432 sip_set_rtp_peer: Sending reinvite on SIP '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' - It's audio soon redirected to IP 78.26.144.10:7076 [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 set_destination: Parsing for address/port to send to [May 27 17:35:39] DEBUG[20587][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.93' into... [May 27 17:35:39] DEBUG[20587][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.93' and port ''. set_destination: set destination to 192.168.3.93:5060 [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:13054 add_sdp: ** Our capability: (ulaw|g722) Video flag: True Text flag: True [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:13055 add_sdp: ** Our prefcodec: (g722) [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:13059 add_sdp: ** Our native-bridge filtered capablity: (g722) Audio is at 13830 Adding codec 100012 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:13192 add_sdp: -- Done with adding codecs to SDP [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:13390 add_sdp: Done building SDP. Settling with this capability: (g722) [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:3505 initialize_initreq: Initializing already initialized SIP dialog 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (presumably reinvite) [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 0 [ 47]: INVITE sip:linphone.iphone@192.168.3.93 SIP/2.0 [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK07cf60ab;rport [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 2 [ 16]: Max-Forwards: 70 [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 3 [ 59]: From: "Patient" ;tag=as3d2643ca [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 4 [ 66]: To: ;tag=525491751 [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 5 [ 42]: Contact: [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 6 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 7 [ 16]: CSeq: 103 INVITE [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 8 [ 30]: User-Agent: Video Medicine PBX [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 10 [ 26]: Supported: replaces, timer [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 12 [ 29]: Content-Type: application/sdp Reliably Transmitting (NAT) to 89.209.100.91:5060: INVITE sip:linphone.iphone@192.168.3.93 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK07cf60ab;rport Max-Forwards: 70 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Contact: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 103 INVITE User-Agent: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 259 v=0 o=root 209909186 209909187 IN IP4 78.26.144.10 s=Asterisk PBX 11.2.1 c=IN IP4 78.26.144.10 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432328 [May 27 17:35:39] DEBUG[20587][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 89.209.100.91:5060 [May 27 17:35:39] DEBUG[20587][C-0000019a]: rtp_engine.c:1278 remote_bridge_loop: Oooh, 'SIP/6000444-000000af' changed end address to 78.26.144.10:7076 (format (g722)) [May 27 17:35:39] DEBUG[20587][C-0000019a]: rtp_engine.c:1281 remote_bridge_loop: Oooh, 'SIP/6000444-000000af' was 78.26.144.10:7076/(format (g722)) <--- SIP read from UDP:89.209.100.91:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK07cf60ab;rport=5060 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 103 INVITE User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK07cf60ab;rport=5060 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 59]: From: "Patient" ;tag=as3d2643ca [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 66]: To: ;tag=525491751 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 103 INVITE [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 17]: Content-Length: 0 --- (8 headers 0 lines) --- [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (Checking To) --From tag as3d2643ca --To-tag 525491751 [May 27 17:35:40] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:4590 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #432328 - INVITE (got response) [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:4597 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' Request 103: Found [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:22342 handle_response_invite: SIP response 100 to RE-invite on outgoing call 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:40] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:78.26.144.10:5060 ---> ACK sip:6000333@209.239.114.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1382005936 From: ;tag=896355678 To: "Doctor" ;tag=as025a484a Call-ID: 1500096782 CSeq: 21 ACK Contact: Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 43]: ACK sip:6000333@209.239.114.51:5060 SIP/2.0 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1382005936 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 55]: From: ;tag=896355678 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 56]: To: "Doctor" ;tag=as025a484a [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 12]: CSeq: 21 ACK [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 43]: Contact: [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 16]: Max-Forwards: 70 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 17]: Content-Length: 0 --- (10 headers 0 lines) --- [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking From) --From tag 896355678 --To-tag as025a484a [May 27 17:35:40] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:27866 handle_incoming: **** Received ACK (6) - Command in SIP ACK [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:4523 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #432327 [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '1500096782' of Response 21: Match Found [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:22145 check_pendings: Sending pending reinvite on '1500096782' [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 1500096782 set_destination: Parsing for address/port to send to [May 27 17:35:40] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107' into... [May 27 17:35:40] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port ''. set_destination: set destination to 192.168.1.107:5060 [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:13054 add_sdp: ** Our capability: (g722) Video flag: True Text flag: True [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:13055 add_sdp: ** Our prefcodec: (nothing) [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:13059 add_sdp: ** Our native-bridge filtered capablity: (g722) Audio is at 11376 Adding codec 100012 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:13192 add_sdp: -- Done with adding codecs to SDP [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:13390 add_sdp: Done building SDP. Settling with this capability: (g722) [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:3505 initialize_initreq: Initializing already initialized SIP dialog 1500096782 (presumably reinvite) [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 0 [ 48]: INVITE sip:linphone.iphone@192.168.1.107 SIP/2.0 [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK10c0defb;rport [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 2 [ 16]: Max-Forwards: 70 [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 3 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 4 [ 53]: To: ;tag=896355678 [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 5 [ 42]: Contact: [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 6 [ 19]: Call-ID: 1500096782 [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 7 [ 16]: CSeq: 102 INVITE [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 8 [ 30]: User-Agent: Video Medicine PBX [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 10 [ 26]: Supported: replaces, timer [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:9568 parse_request: Header 12 [ 29]: Content-Type: application/sdp Reliably Transmitting (NAT) to 78.26.144.10:5060: INVITE sip:linphone.iphone@192.168.1.107 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK10c0defb;rport Max-Forwards: 70 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Contact: Call-ID: 1500096782 CSeq: 102 INVITE User-Agent: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 261 v=0 o=root 426475338 426475339 IN IP4 89.209.100.91 s=Asterisk PBX 11.2.1 c=IN IP4 89.209.100.91 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432329 [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:40] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. [May 27 17:35:40] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3502 ast_rtp_read: 0x7fa8701c93a0 -- start learning mode pass with addr = 78.26.144.10:7076 [May 27 17:35:40] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:1605 rtp_learning_rtp_seq_update: 0x7fa8701c93a0 -- probation = 4, seq = 0 [May 27 17:35:40] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3508 ast_rtp_read: 0x7fa8701c93a0 -- Condition for learning hasn't exited, so reject the frame. [May 27 17:35:40] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3502 ast_rtp_read: 0x7fa8701c93a0 -- start learning mode pass with addr = 78.26.144.10:7076 [May 27 17:35:40] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:1605 rtp_learning_rtp_seq_update: 0x7fa8701c93a0 -- probation = 3, seq = 1 [May 27 17:35:40] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3508 ast_rtp_read: 0x7fa8701c93a0 -- Condition for learning hasn't exited, so reject the frame. [May 27 17:35:40] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3502 ast_rtp_read: 0x7fa8701c93a0 -- start learning mode pass with addr = 78.26.144.10:7076 [May 27 17:35:40] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:1605 rtp_learning_rtp_seq_update: 0x7fa8701c93a0 -- probation = 2, seq = 2 [May 27 17:35:40] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3508 ast_rtp_read: 0x7fa8701c93a0 -- Condition for learning hasn't exited, so reject the frame. [May 27 17:35:40] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3502 ast_rtp_read: 0x7fa8701c93a0 -- start learning mode pass with addr = 78.26.144.10:7076 [May 27 17:35:40] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:1605 rtp_learning_rtp_seq_update: 0x7fa8701c93a0 -- probation = 1, seq = 3 [May 27 17:35:40] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3512 ast_rtp_read: 0x7fa8701c93a0 -- Probation Ended. Set strict_rtp_state to STRICT_RTP_CLOSED with address 78.26.144.10:7076 <--- SIP read from UDP:78.26.144.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK10c0defb;rport=5060 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Call-ID: 1500096782 CSeq: 102 INVITE User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK10c0defb;rport=5060 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 53]: To: ;tag=896355678 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 17]: Content-Length: 0 --- (8 headers 0 lines) --- [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking To) --From tag as025a484a --To-tag 896355678 [May 27 17:35:40] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:4590 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #432329 - INVITE (got response) [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:4597 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1500096782' Request 102: Found [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:22342 handle_response_invite: SIP response 100 to RE-invite on outgoing call 1500096782 [May 27 17:35:40] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:78.26.144.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK10c0defb;rport=5060 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Call-ID: 1500096782 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 230 v=0 o=6000444 1786 3813 IN IP4 78.26.144.10 s=Talk c=IN IP4 78.26.144.10 b=AS:380 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7077 IN IP4 192.168.1.107 <-------------> [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK10c0defb;rport=5060 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 53]: To: ;tag=896355678 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 43]: Contact: [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 29]: Content-Type: application/sdp [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 19]: Content-Length: 230 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 0]: [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000444 1786 3813 IN IP4 78.26.144.10 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 3 [ 21]: c=IN IP4 78.26.144.10 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 6 [ 26]: m=audio 7076 RTP/AVP 9 101 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 7 [ 20]: a=rtpmap:9 G722/8000 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 9 [ 15]: a=fmtp:101 0-11 [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9605 parse_request: Body 10 [ 32]: a=rtcp:7077 IN IP4 192.168.1.107 --- (10 headers 11 lines) --- [May 27 17:35:40] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking To) --From tag as025a484a --To-tag 896355678 [May 27 17:35:40] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:4518 __sip_ack: Acked pending invite 102 [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '1500096782' of Request 102: Match Found [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:22342 handle_response_invite: SIP response 200 to RE-invite on outgoing call 1500096782 [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=6000444 1786 3813 IN IP4 78.26.144.10... OK. [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [May 27 17:35:40] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '78.26.144.10' into... [May 27 17:35:40] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '78.26.144.10' and port ''. [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 78.26.144.10... OK. [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED. [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. Found RTP audio format 9 [May 27 17:35:40] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa8980c4ad0 Found RTP audio format 101 [May 27 17:35:40] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa8980c4ad0 Found audio description format G722 for ID 9 [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. Found audio description format telephone-event for ID 101 [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtcp:7077 IN IP4 192.168.1.107... UNSUPPORTED OR FAILED. Capabilities: us - (ulaw|g722|h264|vp8), peer - audio=(g722)/video=(nothing)/text=(nothing), combined - (g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 78.26.144.10:7076 [May 27 17:35:40] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa8980c4ad0 to 0x7fa8701bc660 [May 27 17:35:40] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa8980c4ad0 to 0x7fa8701bc660 [May 27 17:35:40] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa8701bc498' Peer doesn't provide video [May 27 17:35:40] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa870202818' Peer doesn't provide T.140 [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10668 process_sdp: We're settling with these formats: (g722) [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10675 process_sdp: We have an owner, now see if we need to change this call [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:10681 process_sdp: Setting native formats after processing SDP. peer joint formats (g722), old nativeformats (g722) [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:6669 update_call_counter: Updating call counter for incoming call [May 27 17:35:40] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000444 [May 27 17:35:40] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000444 [May 27 17:35:40] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000444 - state 2 (In use) [May 27 17:35:40] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000444' state '2' [May 27 17:35:40] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000444' changed to state '2' (In use) but we don't care because they're not a member of any queue. [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 1500096782 set_destination: Parsing for address/port to send to [May 27 17:35:40] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107' into... [May 27 17:35:40] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port ''. set_destination: set destination to 192.168.1.107:5060 Transmitting (NAT) to 78.26.144.10:5060: ACK sip:linphone.iphone@192.168.1.107 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK0ca166c8;rport Max-Forwards: 70 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Contact: Call-ID: 1500096782 CSeq: 102 ACK User-Agent: Video Medicine PBX Content-Length: 0 --- [May 27 17:35:40] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'ACK sip:lin' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:40] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. [May 27 17:35:40] DEBUG[28657]: chan_sip.c:4098 retrans_pkt: SIP TIMER: Not rescheduling id #432325:OPTIONS (Method 3) (No timer T1) Retransmitting #1 (NAT) to 92.113.50.205:36857: OPTIONS sip:5000000248@92.113.50.205:36857;line=6666ca59cc8426b SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK13cfd4af;rport Max-Forwards: 70 From: "asterisk" ;tag=as503b4680 To: Contact: Call-ID: 61abb47f2f9188dc71e38d6d6c3e288f@209.239.114.51:5060 CSeq: 102 OPTIONS User-Agent: Video Medicine PBX Date: Mon, 27 May 2013 17:35:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [May 27 17:35:40] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 92.113.50.205:36857 <--- SIP read from UDP:89.209.100.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK07cf60ab;rport=5060 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 103 INVITE Contact: Content-Type: application/sdp User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 230 v=0 o=6000333 622 2565 IN IP4 89.209.100.91 s=Talk c=IN IP4 89.209.100.91 b=AS:380 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7077 IN IP4 192.168.3.93 <-------------> [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK07cf60ab;rport=5060 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 59]: From: "Patient" ;tag=as3d2643ca [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 66]: To: ;tag=525491751 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 103 INVITE [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 43]: Contact: [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 29]: Content-Type: application/sdp [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 19]: Content-Length: 230 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 0]: [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000333 622 2565 IN IP4 89.209.100.91 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 3 [ 22]: c=IN IP4 89.209.100.91 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 6 [ 26]: m=audio 7076 RTP/AVP 9 101 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 7 [ 20]: a=rtpmap:9 G722/8000 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 9 [ 15]: a=fmtp:101 0-11 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9605 parse_request: Body 10 [ 31]: a=rtcp:7077 IN IP4 192.168.3.93 --- (10 headers 11 lines) --- [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (Checking To) --From tag as3d2643ca --To-tag 525491751 [May 27 17:35:41] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:4518 __sip_ack: Acked pending invite 103 [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' of Request 103: Match Found [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:22342 handle_response_invite: SIP response 200 to RE-invite on outgoing call 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=6000333 622 2565 IN IP4 89.209.100.91... OK. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [May 27 17:35:41] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '89.209.100.91' into... [May 27 17:35:41] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '89.209.100.91' and port ''. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 89.209.100.91... OK. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. Found RTP audio format 9 [May 27 17:35:41] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa8980c4ad0 Found RTP audio format 101 [May 27 17:35:41] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa8980c4ad0 Found audio description format G722 for ID 9 [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. Found audio description format telephone-event for ID 101 [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtcp:7077 IN IP4 192.168.3.93... UNSUPPORTED OR FAILED. Capabilities: us - (ulaw|g722|h264|vp8), peer - audio=(g722)/video=(nothing)/text=(nothing), combined - (g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 89.209.100.91:7076 [May 27 17:35:41] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa8980c4ad0 to 0x7fa8740266b0 [May 27 17:35:41] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa8980c4ad0 to 0x7fa8740266b0 [May 27 17:35:41] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa8740264e8' Peer doesn't provide video [May 27 17:35:41] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa874063918' Peer doesn't provide T.140 [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10668 process_sdp: We're settling with these formats: (g722) [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10675 process_sdp: We have an owner, now see if we need to change this call [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10681 process_sdp: Setting native formats after processing SDP. peer joint formats (g722), old nativeformats (ulaw) [May 27 17:35:41] DEBUG[28657][C-0000019a]: channel.c:5355 set_format: Set channel SIP/6000333-000000b0 to read format g722 [May 27 17:35:41] DEBUG[28657][C-0000019a]: channel.c:5355 set_format: Set channel SIP/6000333-000000b0 to write format g722 [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:6669 update_call_counter: Updating call counter for outgoing call [May 27 17:35:41] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000333 [May 27 17:35:41] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000333 [May 27 17:35:41] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000333 - state 2 (In use) [May 27 17:35:41] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000333' state '2' [May 27 17:35:41] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000333' changed to state '2' (In use) but we don't care because they're not a member of any queue. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 set_destination: Parsing for address/port to send to [May 27 17:35:41] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.93' into... [May 27 17:35:41] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.93' and port ''. set_destination: set destination to 192.168.3.93:5060 Transmitting (NAT) to 89.209.100.91:5060: ACK sip:linphone.iphone@192.168.3.93 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK039fec50;rport Max-Forwards: 70 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Contact: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 103 ACK User-Agent: Video Medicine PBX Content-Length: 0 --- [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'ACK sip:lin' onto UDP socket destined for 89.209.100.91:5060 [May 27 17:35:41] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. [May 27 17:35:41] DEBUG[20587][C-0000019a]: rtp_engine.c:1243 remote_bridge_loop: Oooh, 'SIP/6000333-000000b0' changed end address to 89.209.100.91:7076 (format (g722)) [May 27 17:35:41] DEBUG[20587][C-0000019a]: rtp_engine.c:1246 remote_bridge_loop: Oooh, 'SIP/6000333-000000b0' changed end vaddress to (null) (format (g722)) [May 27 17:35:41] DEBUG[20587][C-0000019a]: rtp_engine.c:1249 remote_bridge_loop: Oooh, 'SIP/6000333-000000b0' changed end taddress to (null) (format (g722)) [May 27 17:35:41] DEBUG[20587][C-0000019a]: rtp_engine.c:1252 remote_bridge_loop: Oooh, 'SIP/6000333-000000b0' was 89.209.100.91:7076/(format (ulaw|g722)) [May 27 17:35:41] DEBUG[20587][C-0000019a]: rtp_engine.c:1255 remote_bridge_loop: Oooh, 'SIP/6000333-000000b0' was (null)/(format (ulaw|g722)) [May 27 17:35:41] DEBUG[20587][C-0000019a]: rtp_engine.c:1258 remote_bridge_loop: Oooh, 'SIP/6000333-000000b0' was (null)/(format (ulaw|g722)) [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:32432 sip_set_rtp_peer: Sending reinvite on SIP '1500096782' - It's audio soon redirected to IP 89.209.100.91:7076 [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 1500096782 set_destination: Parsing for address/port to send to [May 27 17:35:41] DEBUG[20587][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107' into... [May 27 17:35:41] DEBUG[20587][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port ''. set_destination: set destination to 192.168.1.107:5060 [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:13054 add_sdp: ** Our capability: (g722) Video flag: True Text flag: True [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:13055 add_sdp: ** Our prefcodec: (nothing) [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:13059 add_sdp: ** Our native-bridge filtered capablity: (g722) Audio is at 11376 Adding codec 100012 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:13192 add_sdp: -- Done with adding codecs to SDP [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:13390 add_sdp: Done building SDP. Settling with this capability: (g722) [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:3505 initialize_initreq: Initializing already initialized SIP dialog 1500096782 (presumably reinvite) [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 0 [ 48]: INVITE sip:linphone.iphone@192.168.1.107 SIP/2.0 [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK095136b6;rport [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 2 [ 16]: Max-Forwards: 70 [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 3 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 4 [ 53]: To: ;tag=896355678 [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 5 [ 42]: Contact: [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 6 [ 19]: Call-ID: 1500096782 [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 7 [ 16]: CSeq: 103 INVITE [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 8 [ 30]: User-Agent: Video Medicine PBX [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 10 [ 26]: Supported: replaces, timer [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 12 [ 29]: Content-Type: application/sdp Reliably Transmitting (NAT) to 78.26.144.10:5060: INVITE sip:linphone.iphone@192.168.1.107 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK095136b6;rport Max-Forwards: 70 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Contact: Call-ID: 1500096782 CSeq: 103 INVITE User-Agent: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 261 v=0 o=root 426475338 426475340 IN IP4 89.209.100.91 s=Asterisk PBX 11.2.1 c=IN IP4 89.209.100.91 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432330 [May 27 17:35:41] DEBUG[20587][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:4098 retrans_pkt: SIP TIMER: Not rescheduling id #432325:OPTIONS (Method 3) (No timer T1) Retransmitting #2 (NAT) to 92.113.50.205:36857: OPTIONS sip:5000000248@92.113.50.205:36857;line=6666ca59cc8426b SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK13cfd4af;rport Max-Forwards: 70 From: "asterisk" ;tag=as503b4680 To: Contact: Call-ID: 61abb47f2f9188dc71e38d6d6c3e288f@209.239.114.51:5060 CSeq: 102 OPTIONS User-Agent: Video Medicine PBX Date: Mon, 27 May 2013 17:35:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [May 27 17:35:41] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 92.113.50.205:36857 <--- SIP read from UDP:78.26.144.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK095136b6;rport=5060 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Call-ID: 1500096782 CSeq: 103 INVITE User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK095136b6;rport=5060 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 53]: To: ;tag=896355678 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 103 INVITE [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 17]: Content-Length: 0 --- (8 headers 0 lines) --- [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking To) --From tag as025a484a --To-tag 896355678 [May 27 17:35:41] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:4590 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #432330 - INVITE (got response) [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:4597 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1500096782' Request 103: Found [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:22342 handle_response_invite: SIP response 100 to RE-invite on outgoing call 1500096782 [May 27 17:35:41] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:78.26.144.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK095136b6;rport=5060 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Call-ID: 1500096782 CSeq: 103 INVITE Contact: Content-Type: application/sdp User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 230 v=0 o=6000444 1786 3814 IN IP4 78.26.144.10 s=Talk c=IN IP4 78.26.144.10 b=AS:380 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7077 IN IP4 192.168.1.107 <-------------> [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK095136b6;rport=5060 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 53]: To: ;tag=896355678 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 103 INVITE [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 43]: Contact: [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 29]: Content-Type: application/sdp [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 19]: Content-Length: 230 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 0]: [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000444 1786 3814 IN IP4 78.26.144.10 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 3 [ 21]: c=IN IP4 78.26.144.10 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 6 [ 26]: m=audio 7076 RTP/AVP 9 101 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 7 [ 20]: a=rtpmap:9 G722/8000 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 9 [ 15]: a=fmtp:101 0-11 [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9605 parse_request: Body 10 [ 32]: a=rtcp:7077 IN IP4 192.168.1.107 --- (10 headers 11 lines) --- [May 27 17:35:41] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking To) --From tag as025a484a --To-tag 896355678 [May 27 17:35:41] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:4518 __sip_ack: Acked pending invite 103 [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '1500096782' of Request 103: Match Found [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:22342 handle_response_invite: SIP response 200 to RE-invite on outgoing call 1500096782 [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=6000444 1786 3814 IN IP4 78.26.144.10... OK. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [May 27 17:35:41] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '78.26.144.10' into... [May 27 17:35:41] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '78.26.144.10' and port ''. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 78.26.144.10... OK. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. Found RTP audio format 9 [May 27 17:35:41] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa8980c4ad0 Found RTP audio format 101 [May 27 17:35:41] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa8980c4ad0 Found audio description format G722 for ID 9 [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. Found audio description format telephone-event for ID 101 [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtcp:7077 IN IP4 192.168.1.107... UNSUPPORTED OR FAILED. Capabilities: us - (ulaw|g722|h264|vp8), peer - audio=(g722)/video=(nothing)/text=(nothing), combined - (g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 78.26.144.10:7076 [May 27 17:35:41] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa8980c4ad0 to 0x7fa8701bc660 [May 27 17:35:41] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa8980c4ad0 to 0x7fa8701bc660 [May 27 17:35:41] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa8701bc498' Peer doesn't provide video [May 27 17:35:41] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa870202818' Peer doesn't provide T.140 [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10668 process_sdp: We're settling with these formats: (g722) [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10675 process_sdp: We have an owner, now see if we need to change this call [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:10681 process_sdp: Setting native formats after processing SDP. peer joint formats (g722), old nativeformats (g722) [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:6669 update_call_counter: Updating call counter for incoming call [May 27 17:35:41] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000444 [May 27 17:35:41] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000444 [May 27 17:35:41] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000444 - state 2 (In use) [May 27 17:35:41] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000444' state '2' [May 27 17:35:41] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000444' changed to state '2' (In use) but we don't care because they're not a member of any queue. [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 1500096782 set_destination: Parsing for address/port to send to [May 27 17:35:41] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107' into... [May 27 17:35:41] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port ''. set_destination: set destination to 192.168.1.107:5060 Transmitting (NAT) to 78.26.144.10:5060: ACK sip:linphone.iphone@192.168.1.107 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK449aad07;rport Max-Forwards: 70 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Contact: Call-ID: 1500096782 CSeq: 103 ACK User-Agent: Video Medicine PBX Content-Length: 0 --- [May 27 17:35:41] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'ACK sip:lin' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:41] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. [May 27 17:35:42] DEBUG[28657]: chan_sip.c:4098 retrans_pkt: SIP TIMER: Not rescheduling id #432325:OPTIONS (Method 3) (No timer T1) Retransmitting #3 (NAT) to 92.113.50.205:36857: OPTIONS sip:5000000248@92.113.50.205:36857;line=6666ca59cc8426b SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK13cfd4af;rport Max-Forwards: 70 From: "asterisk" ;tag=as503b4680 To: Contact: Call-ID: 61abb47f2f9188dc71e38d6d6c3e288f@209.239.114.51:5060 CSeq: 102 OPTIONS User-Agent: Video Medicine PBX Date: Mon, 27 May 2013 17:35:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [May 27 17:35:42] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 92.113.50.205:36857 [May 27 17:35:43] DEBUG[28657]: chan_sip.c:4098 retrans_pkt: SIP TIMER: Not rescheduling id #432325:OPTIONS (Method 3) (No timer T1) Retransmitting #4 (NAT) to 92.113.50.205:36857: OPTIONS sip:5000000248@92.113.50.205:36857;line=6666ca59cc8426b SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK13cfd4af;rport Max-Forwards: 70 From: "asterisk" ;tag=as503b4680 To: Contact: Call-ID: 61abb47f2f9188dc71e38d6d6c3e288f@209.239.114.51:5060 CSeq: 102 OPTIONS User-Agent: Video Medicine PBX Date: Mon, 27 May 2013 17:35:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [May 27 17:35:43] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 92.113.50.205:36857 [May 27 17:35:43] DEBUG[28657]: chan_sip.c:6817 sip_destroy: Destroying SIP dialog 61abb47f2f9188dc71e38d6d6c3e288f@209.239.114.51:5060 Really destroying SIP dialog '61abb47f2f9188dc71e38d6d6c3e288f@209.239.114.51:5060' Method: OPTIONS <--- SIP read from UDP:78.26.144.10:5060 ---> INVITE sip:6000333@209.239.114.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1239891845 From: ;tag=896355678 To: "Doctor" ;tag=as025a484a Call-ID: 1500096782 CSeq: 22 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Subject: Media change Content-Length: 317 v=0 o=6000444 1786 3815 IN IP4 78.26.144.10 s=Talk c=IN IP4 78.26.144.10 b=AS:380 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7077 IN IP4 192.168.1.107 m=video 9078 RTP/AVP 102 a=rtpmap:102 H264/90000 a=fmtp:102 profile-level-id=428014 <-------------> [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 46]: INVITE sip:6000333@209.239.114.51:5060 SIP/2.0 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1239891845 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 55]: From: ;tag=896355678 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 56]: To: "Doctor" ;tag=as025a484a [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 15]: CSeq: 22 INVITE [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 43]: Contact: [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 29]: Content-Type: application/sdp [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 16]: Max-Forwards: 70 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 11 [ 21]: Subject: Media change [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 12 [ 19]: Content-Length: 317 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 13 [ 0]: [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000444 1786 3815 IN IP4 78.26.144.10 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 3 [ 21]: c=IN IP4 78.26.144.10 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 6 [ 26]: m=audio 7076 RTP/AVP 9 101 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 7 [ 20]: a=rtpmap:9 G722/8000 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 9 [ 15]: a=fmtp:101 0-11 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 10 [ 32]: a=rtcp:7077 IN IP4 192.168.1.107 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 11 [ 24]: m=video 9078 RTP/AVP 102 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 12 [ 23]: a=rtpmap:102 H264/90000 [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9605 parse_request: Body 13 [ 34]: a=fmtp:102 profile-level-id=428014 --- (13 headers 14 lines) --- [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking From) --From tag 896355678 --To-tag as025a484a [May 27 17:35:44] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:27866 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [May 27 17:35:44] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107:5060' into... [May 27 17:35:44] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port '5060'. [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:17845 check_via: NAT detected for 192.168.1.107:5060 / 78.26.144.10:5060 Sending to 78.26.144.10:5060 (NAT) [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:24999 handle_request_invite: Initializing initreq for method INVITE - callid 1500096782 [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=6000444 1786 3815 IN IP4 78.26.144.10... OK. [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [May 27 17:35:44] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '78.26.144.10' into... [May 27 17:35:44] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '78.26.144.10' and port ''. [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 78.26.144.10... OK. [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED. [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. Found RTP audio format 9 [May 27 17:35:44] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa8980c5630 Found RTP audio format 101 [May 27 17:35:44] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa8980c5630 Found audio description format G722 for ID 9 [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. Found audio description format telephone-event for ID 101 [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtcp:7077 IN IP4 192.168.1.107... UNSUPPORTED OR FAILED. Found RTP video format 102 [May 27 17:35:44] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 102 based on m type on 0x7fa8980c9980 Found video description format H264 for ID 102 [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=rtpmap:102 H264/90000... OK. [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=fmtp:102 profile-level-id=428014... OK. Capabilities: us - (ulaw|g722|h264|vp8), peer - audio=(g722)/video=(h264)/text=(nothing), combined - (g722|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 27 17:35:44] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa8701bc498' Peer audio RTP is at port 78.26.144.10:7076 [May 27 17:35:44] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa8980c5630 to 0x7fa8701bc660 [May 27 17:35:44] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa8980c5630 to 0x7fa8701bc660 [May 27 17:35:44] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3817 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7fa8701bc498' Peer video RTP is at port 78.26.144.10:9078 [May 27 17:35:44] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 102 from 0x7fa8980c9980 to 0x7fa8701cc190 [May 27 17:35:44] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa870202818' Peer doesn't provide T.140 [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10668 process_sdp: We're settling with these formats: (g722|h264) [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10675 process_sdp: We have an owner, now see if we need to change this call [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:10681 process_sdp: Setting native formats after processing SDP. peer joint formats (g722|h264), old nativeformats (g722) [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:25235 handle_request_invite: Got a SIP re-invite for call 1500096782 [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:25511 handle_request_invite: SIP/6000444-000000af: This call is UP.... <--- Transmitting (NAT) to 78.26.144.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK1239891845;received=78.26.144.10;rport=5060 From: ;tag=896355678 To: "Doctor" ;tag=as025a484a Call-ID: 1500096782 CSeq: 22 INVITE Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:13504 transmit_response_with_sdp: Setting framing from config on incoming call [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:13001 add_sdp: This call needs video offers, but caller probably did not offer it! [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:13054 add_sdp: ** Our capability: (g722|h264) Video flag: False Text flag: True [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:13055 add_sdp: ** Our prefcodec: (nothing) [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:13059 add_sdp: ** Our native-bridge filtered capablity: (g722) Audio is at 11376 Adding codec 100012 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:13192 add_sdp: -- Done with adding codecs to SDP [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:13390 add_sdp: Done building SDP. Settling with this capability: (g722) <--- Reliably Transmitting (NAT) to 78.26.144.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK1239891845;received=78.26.144.10;rport=5060 From: ;tag=896355678 To: "Doctor" ;tag=as025a484a Call-ID: 1500096782 CSeq: 22 INVITE Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 282 v=0 o=root 426475338 426475341 IN IP4 89.209.100.91 s=Asterisk PBX 11.2.1 c=IN IP4 89.209.100.91 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 0 RTP/AVP 102 <------------> [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432332 [May 27 17:35:44] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:44] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:78.26.144.10:5060 ---> jaK <-------------> [May 27 17:35:44] DEBUG[28657]: chan_sip.c:9605 parse_request: Header 0 [ 3]: jaK [May 27 17:35:44] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:2085 ast_rtp_update_source: Setting the marker bit due to a source update [May 27 17:35:44] DEBUG[20587][C-0000019a]: rtp_engine.c:1278 remote_bridge_loop: Oooh, 'SIP/6000444-000000af' changed end address to 78.26.144.10:7076 (format (g722|h264)) [May 27 17:35:44] DEBUG[20587][C-0000019a]: rtp_engine.c:1281 remote_bridge_loop: Oooh, 'SIP/6000444-000000af' was 78.26.144.10:7076/(format (g722|h264)) [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:32432 sip_set_rtp_peer: Sending reinvite on SIP '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' - It's audio soon redirected to IP 78.26.144.10:7076 [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 set_destination: Parsing for address/port to send to [May 27 17:35:44] DEBUG[20587][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.93' into... [May 27 17:35:44] DEBUG[20587][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.93' and port ''. set_destination: set destination to 192.168.3.93:5060 [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:13054 add_sdp: ** Our capability: (g722) Video flag: True Text flag: True [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:13055 add_sdp: ** Our prefcodec: (g722) [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:13059 add_sdp: ** Our native-bridge filtered capablity: (g722) Audio is at 13830 Adding codec 100012 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:13192 add_sdp: -- Done with adding codecs to SDP [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:13390 add_sdp: Done building SDP. Settling with this capability: (g722) [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:3505 initialize_initreq: Initializing already initialized SIP dialog 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (presumably reinvite) [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 0 [ 47]: INVITE sip:linphone.iphone@192.168.3.93 SIP/2.0 [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7b76db58;rport [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 2 [ 16]: Max-Forwards: 70 [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 3 [ 59]: From: "Patient" ;tag=as3d2643ca [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 4 [ 66]: To: ;tag=525491751 [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 5 [ 42]: Contact: [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 6 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 7 [ 16]: CSeq: 104 INVITE [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 8 [ 30]: User-Agent: Video Medicine PBX [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 10 [ 26]: Supported: replaces, timer [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 12 [ 29]: Content-Type: application/sdp Reliably Transmitting (NAT) to 89.209.100.91:5060: INVITE sip:linphone.iphone@192.168.3.93 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7b76db58;rport Max-Forwards: 70 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Contact: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 104 INVITE User-Agent: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 259 v=0 o=root 209909186 209909188 IN IP4 78.26.144.10 s=Asterisk PBX 11.2.1 c=IN IP4 78.26.144.10 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432333 [May 27 17:35:44] DEBUG[20587][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 89.209.100.91:5060 <--- SIP read from UDP:89.209.100.91:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7b76db58;rport=5060 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 104 INVITE User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7b76db58;rport=5060 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 59]: From: "Patient" ;tag=as3d2643ca [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 66]: To: ;tag=525491751 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 104 INVITE [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 17]: Content-Length: 0 --- (8 headers 0 lines) --- [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (Checking To) --From tag as3d2643ca --To-tag 525491751 [May 27 17:35:45] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:4590 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #432333 - INVITE (got response) [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:4597 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' Request 104: Found [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:22342 handle_response_invite: SIP response 100 to RE-invite on outgoing call 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:45] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:89.209.100.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7b76db58;rport=5060 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 104 INVITE Contact: Content-Type: application/sdp User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 230 v=0 o=6000333 622 2566 IN IP4 89.209.100.91 s=Talk c=IN IP4 89.209.100.91 b=AS:380 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7077 IN IP4 192.168.3.93 <-------------> [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7b76db58;rport=5060 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 59]: From: "Patient" ;tag=as3d2643ca [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 66]: To: ;tag=525491751 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 104 INVITE [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 43]: Contact: [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 29]: Content-Type: application/sdp [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 19]: Content-Length: 230 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 0]: [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000333 622 2566 IN IP4 89.209.100.91 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 3 [ 22]: c=IN IP4 89.209.100.91 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 6 [ 26]: m=audio 7076 RTP/AVP 9 101 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 7 [ 20]: a=rtpmap:9 G722/8000 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 9 [ 15]: a=fmtp:101 0-11 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9605 parse_request: Body 10 [ 31]: a=rtcp:7077 IN IP4 192.168.3.93 --- (10 headers 11 lines) --- [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (Checking To) --From tag as3d2643ca --To-tag 525491751 [May 27 17:35:45] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:4518 __sip_ack: Acked pending invite 104 [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' of Request 104: Match Found [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:22342 handle_response_invite: SIP response 200 to RE-invite on outgoing call 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=6000333 622 2566 IN IP4 89.209.100.91... OK. [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [May 27 17:35:45] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '89.209.100.91' into... [May 27 17:35:45] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '89.209.100.91' and port ''. [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 89.209.100.91... OK. [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED. [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. Found RTP audio format 9 [May 27 17:35:45] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa8980c4ad0 Found RTP audio format 101 [May 27 17:35:45] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa8980c4ad0 Found audio description format G722 for ID 9 [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. Found audio description format telephone-event for ID 101 [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtcp:7077 IN IP4 192.168.3.93... UNSUPPORTED OR FAILED. Capabilities: us - (ulaw|g722|h264|vp8), peer - audio=(g722)/video=(nothing)/text=(nothing), combined - (g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 89.209.100.91:7076 [May 27 17:35:45] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa8980c4ad0 to 0x7fa8740266b0 [May 27 17:35:45] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa8980c4ad0 to 0x7fa8740266b0 [May 27 17:35:45] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa8740264e8' Peer doesn't provide video [May 27 17:35:45] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa874063918' Peer doesn't provide T.140 [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10668 process_sdp: We're settling with these formats: (g722) [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10675 process_sdp: We have an owner, now see if we need to change this call [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:10681 process_sdp: Setting native formats after processing SDP. peer joint formats (g722), old nativeformats (g722) [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:6669 update_call_counter: Updating call counter for outgoing call [May 27 17:35:45] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000333 [May 27 17:35:45] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000333 [May 27 17:35:45] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000333 - state 2 (In use) [May 27 17:35:45] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000333' state '2' [May 27 17:35:45] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000333' changed to state '2' (In use) but we don't care because they're not a member of any queue. [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 set_destination: Parsing for address/port to send to [May 27 17:35:45] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.93' into... [May 27 17:35:45] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.93' and port ''. set_destination: set destination to 192.168.3.93:5060 Transmitting (NAT) to 89.209.100.91:5060: ACK sip:linphone.iphone@192.168.3.93 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK5431daf6;rport Max-Forwards: 70 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Contact: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 104 ACK User-Agent: Video Medicine PBX Content-Length: 0 --- [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'ACK sip:lin' onto UDP socket destined for 89.209.100.91:5060 [May 27 17:35:45] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:78.26.144.10:5060 ---> ACK sip:6000333@209.239.114.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1656624136 From: ;tag=896355678 To: "Doctor" ;tag=as025a484a Call-ID: 1500096782 CSeq: 22 ACK Contact: Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 43]: ACK sip:6000333@209.239.114.51:5060 SIP/2.0 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1656624136 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 55]: From: ;tag=896355678 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 56]: To: "Doctor" ;tag=as025a484a [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 12]: CSeq: 22 ACK [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 43]: Contact: [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 16]: Max-Forwards: 70 [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 17]: Content-Length: 0 --- (10 headers 0 lines) --- [May 27 17:35:45] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking From) --From tag 896355678 --To-tag as025a484a [May 27 17:35:45] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:27866 handle_incoming: **** Received ACK (6) - Command in SIP ACK [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:4523 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #432332 [May 27 17:35:45] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '1500096782' of Response 22: Match Found [May 27 17:35:45] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:89.209.100.91:5060 ---> INVITE sip:6000444@209.239.114.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK850286143 From: ;tag=525491751 To: "Patient" ;tag=as3d2643ca Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 2 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Subject: Media change Content-Length: 526 v=0 o=6000333 622 2567 IN IP4 89.209.100.91 s=Talk c=IN IP4 89.209.100.91 b=AS:380 t=0 0 m=audio 7076 RTP/AVP 120 111 110 0 8 121 100 9 3 101 a=rtpmap:120 SILK/16000 a=rtpmap:111 speex/16000 a=fmtp:111 vbr=on a=rtpmap:110 speex/8000 a=fmtp:110 vbr=on a=rtpmap:121 SILK/24000 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7077 IN IP4 192.168.3.93 m=video 9078 RTP/AVP 102 a=rtpmap:102 H264/90000 a=fmtp:102 profile-level-id=428014 <-------------> [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 46]: INVITE sip:6000444@209.239.114.51:5060 SIP/2.0 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK850286143 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 68]: From: ;tag=525491751 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 57]: To: "Patient" ;tag=as3d2643ca [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 14]: CSeq: 2 INVITE [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 43]: Contact: [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 29]: Content-Type: application/sdp [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 16]: Max-Forwards: 70 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 11 [ 21]: Subject: Media change [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 12 [ 19]: Content-Length: 526 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 13 [ 0]: [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000333 622 2567 IN IP4 89.209.100.91 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 3 [ 22]: c=IN IP4 89.209.100.91 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 6 [ 52]: m=audio 7076 RTP/AVP 120 111 110 0 8 121 100 9 3 101 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 7 [ 23]: a=rtpmap:120 SILK/16000 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 8 [ 24]: a=rtpmap:111 speex/16000 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 9 [ 17]: a=fmtp:111 vbr=on [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 10 [ 23]: a=rtpmap:110 speex/8000 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 11 [ 17]: a=fmtp:110 vbr=on [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 12 [ 23]: a=rtpmap:121 SILK/24000 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 13 [ 22]: a=rtpmap:100 iLBC/8000 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 14 [ 18]: a=fmtp:100 mode=30 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 15 [ 20]: a=rtpmap:9 G722/8000 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 16 [ 33]: a=rtpmap:101 telephone-event/8000 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 17 [ 15]: a=fmtp:101 0-11 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 18 [ 31]: a=rtcp:7077 IN IP4 192.168.3.93 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 19 [ 24]: m=video 9078 RTP/AVP 102 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 20 [ 23]: a=rtpmap:102 H264/90000 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9605 parse_request: Body 21 [ 34]: a=fmtp:102 profile-level-id=428014 --- (13 headers 22 lines) --- [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (Checking From) --From tag 525491751 --To-tag as3d2643ca [May 27 17:35:46] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:27866 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [May 27 17:35:46] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.93:5060' into... [May 27 17:35:46] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.93' and port '5060'. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:17845 check_via: NAT detected for 192.168.3.93:5060 / 89.209.100.91:5060 Sending to 89.209.100.91:5060 (NAT) [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:24999 handle_request_invite: Initializing initreq for method INVITE - callid 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=6000333 622 2567 IN IP4 89.209.100.91... OK. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [May 27 17:35:46] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '89.209.100.91' into... [May 27 17:35:46] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '89.209.100.91' and port ''. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 89.209.100.91... OK. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. Found RTP audio format 120 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 120 based on m type on 0x7fa8980c5630 Found RTP audio format 111 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 111 based on m type on 0x7fa8980c5630 Found RTP audio format 110 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 110 based on m type on 0x7fa8980c5630 Found RTP audio format 0 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7fa8980c5630 Found RTP audio format 8 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7fa8980c5630 Found RTP audio format 121 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 121 based on m type on 0x7fa8980c5630 Found RTP audio format 100 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 100 based on m type on 0x7fa8980c5630 Found RTP audio format 9 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa8980c5630 Found RTP audio format 3 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 3 based on m type on 0x7fa8980c5630 Found RTP audio format 101 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa8980c5630 Found audio description format SILK for ID 120 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:120 SILK/16000... OK. Found audio description format speex for ID 111 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:111 speex/16000... OK. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:111 vbr=on... OK. Found audio description format speex for ID 110 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:110 speex/8000... OK. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:110 vbr=on... OK. Found audio description format SILK for ID 121 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:121 SILK/24000... OK. Found audio description format iLBC for ID 100 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:100 iLBC/8000... OK. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:100 mode=30... OK. Found audio description format G722 for ID 9 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. Found audio description format telephone-event for ID 101 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtcp:7077 IN IP4 192.168.3.93... UNSUPPORTED OR FAILED. Found RTP video format 102 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 102 based on m type on 0x7fa8980c9980 Found video description format H264 for ID 102 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=rtpmap:102 H264/90000... OK. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=fmtp:102 profile-level-id=428014... OK. Capabilities: us - (ulaw|g722|h264|vp8), peer - audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|silk16|silk24)/video=(h264)/text=(nothing), combined - (ulaw|g722|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 27 17:35:46] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa8740264e8' Peer audio RTP is at port 89.209.100.91:7076 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0x7fa8980c5630 to 0x7fa8740266b0 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 3 from 0x7fa8980c5630 to 0x7fa8740266b0 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0x7fa8980c5630 to 0x7fa8740266b0 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa8980c5630 to 0x7fa8740266b0 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 100 from 0x7fa8980c5630 to 0x7fa8740266b0 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa8980c5630 to 0x7fa8740266b0 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 110 from 0x7fa8980c5630 to 0x7fa8740266b0 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 111 from 0x7fa8980c5630 to 0x7fa8740266b0 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 120 from 0x7fa8980c5630 to 0x7fa8740266b0 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 121 from 0x7fa8980c5630 to 0x7fa8740266b0 [May 27 17:35:46] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3817 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7fa8740264e8' Peer video RTP is at port 89.209.100.91:9078 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 102 from 0x7fa8980c9980 to 0x7fa87402ac50 [May 27 17:35:46] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa874063918' Peer doesn't provide T.140 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10668 process_sdp: We're settling with these formats: (ulaw|g722|h264) [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10675 process_sdp: We have an owner, now see if we need to change this call [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10681 process_sdp: Setting native formats after processing SDP. peer joint formats (ulaw|g722|h264), old nativeformats (g722) [May 27 17:35:46] DEBUG[28657][C-0000019a]: channel.c:5355 set_format: Set channel SIP/6000333-000000b0 to read format g722 [May 27 17:35:46] DEBUG[28657][C-0000019a]: channel.c:5355 set_format: Set channel SIP/6000333-000000b0 to write format g722 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:25235 handle_request_invite: Got a SIP re-invite for call 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:25511 handle_request_invite: SIP/6000333-000000b0: This call is UP.... <--- Transmitting (NAT) to 89.209.100.91:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.93:5060;branch=z9hG4bK850286143;received=89.209.100.91;rport=5060 From: ;tag=525491751 To: "Patient" ;tag=as3d2643ca Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 2 INVITE Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 89.209.100.91:5060 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:13504 transmit_response_with_sdp: Setting framing from config on incoming call [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:13004 add_sdp: This call needs video offers! [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:13054 add_sdp: ** Our capability: (ulaw|g722|h264) Video flag: False Text flag: True [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:13055 add_sdp: ** Our prefcodec: (g722) [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:13059 add_sdp: ** Our native-bridge filtered capablity: (g722|h264) Audio is at 13830 Video is at 78.26.144.10:9078 Adding codec 100012 (g722) to SDP Adding video codec 200004 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:13192 add_sdp: -- Done with adding codecs to SDP [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:13390 add_sdp: Done building SDP. Settling with this capability: (g722|h264) <--- Reliably Transmitting (NAT) to 89.209.100.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.93:5060;branch=z9hG4bK850286143;received=89.209.100.91;rport=5060 From: ;tag=525491751 To: "Patient" ;tag=as3d2643ca Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 2 INVITE Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 368 v=0 o=root 209909186 209909189 IN IP4 78.26.144.10 s=Asterisk PBX 11.2.1 c=IN IP4 78.26.144.10 b=CT:384 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 9078 RTP/AVP 102 a=rtpmap:102 H264/90000 a=fmtp:102 profile-level-id=428014 a=sendrecv <------------> [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432334 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 89.209.100.91:5060 [May 27 17:35:46] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:2085 ast_rtp_update_source: Setting the marker bit due to a source update [May 27 17:35:46] DEBUG[20587][C-0000019a]: rtp_engine.c:1243 remote_bridge_loop: Oooh, 'SIP/6000333-000000b0' changed end address to 89.209.100.91:7076 (format (gsm|ulaw|alaw|speex|speex16|ilbc|g722|h264|silk16|silk24)) [May 27 17:35:46] DEBUG[20587][C-0000019a]: rtp_engine.c:1246 remote_bridge_loop: Oooh, 'SIP/6000333-000000b0' changed end vaddress to 89.209.100.91:9078 (format (gsm|ulaw|alaw|speex|speex16|ilbc|g722|h264|silk16|silk24)) [May 27 17:35:46] DEBUG[20587][C-0000019a]: rtp_engine.c:1249 remote_bridge_loop: Oooh, 'SIP/6000333-000000b0' changed end taddress to (null) (format (gsm|ulaw|alaw|speex|speex16|ilbc|g722|h264|silk16|silk24)) [May 27 17:35:46] DEBUG[20587][C-0000019a]: rtp_engine.c:1252 remote_bridge_loop: Oooh, 'SIP/6000333-000000b0' was 89.209.100.91:7076/(format (gsm|ulaw|alaw|speex|speex16|ilbc|g722|h264|silk16|silk24)) [May 27 17:35:46] DEBUG[20587][C-0000019a]: rtp_engine.c:1255 remote_bridge_loop: Oooh, 'SIP/6000333-000000b0' was (null)/(format (gsm|ulaw|alaw|speex|speex16|ilbc|g722|h264|silk16|silk24)) [May 27 17:35:46] DEBUG[20587][C-0000019a]: rtp_engine.c:1258 remote_bridge_loop: Oooh, 'SIP/6000333-000000b0' was (null)/(format (gsm|ulaw|alaw|speex|speex16|ilbc|g722|h264|silk16|silk24)) [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:32432 sip_set_rtp_peer: Sending reinvite on SIP '1500096782' - It's audio soon redirected to IP 89.209.100.91:7076 [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 1500096782 set_destination: Parsing for address/port to send to [May 27 17:35:46] DEBUG[20587][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107' into... [May 27 17:35:46] DEBUG[20587][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port ''. set_destination: set destination to 192.168.1.107:5060 [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:13004 add_sdp: This call needs video offers! [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:13054 add_sdp: ** Our capability: (g722|h264) Video flag: False Text flag: True [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:13055 add_sdp: ** Our prefcodec: (nothing) [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:13059 add_sdp: ** Our native-bridge filtered capablity: (g722|h264) Audio is at 11376 Video is at 89.209.100.91:9078 Adding codec 100012 (g722) to SDP Adding video codec 200004 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:13192 add_sdp: -- Done with adding codecs to SDP [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:13390 add_sdp: Done building SDP. Settling with this capability: (g722|h264) [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:3505 initialize_initreq: Initializing already initialized SIP dialog 1500096782 (presumably reinvite) [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 0 [ 48]: INVITE sip:linphone.iphone@192.168.1.107 SIP/2.0 [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7dcce43e;rport [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 2 [ 16]: Max-Forwards: 70 [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 3 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 4 [ 53]: To: ;tag=896355678 [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 5 [ 42]: Contact: [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 6 [ 19]: Call-ID: 1500096782 [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 7 [ 16]: CSeq: 104 INVITE [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 8 [ 30]: User-Agent: Video Medicine PBX [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 10 [ 26]: Supported: replaces, timer [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 12 [ 29]: Content-Type: application/sdp Reliably Transmitting (NAT) to 78.26.144.10:5060: INVITE sip:linphone.iphone@192.168.1.107 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7dcce43e;rport Max-Forwards: 70 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Contact: Call-ID: 1500096782 CSeq: 104 INVITE User-Agent: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 370 v=0 o=root 426475338 426475342 IN IP4 89.209.100.91 s=Asterisk PBX 11.2.1 c=IN IP4 89.209.100.91 b=CT:384 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 9078 RTP/AVP 102 a=rtpmap:102 H264/90000 a=fmtp:102 profile-level-id=428014 a=sendrecv --- [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432335 [May 27 17:35:46] DEBUG[20587][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:46] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:89.209.100.91:5060 ---> ACK sip:6000444@209.239.114.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK235279206 From: ;tag=525491751 To: "Patient" ;tag=as3d2643ca Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 2 ACK Contact: Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 43]: ACK sip:6000444@209.239.114.51:5060 SIP/2.0 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK235279206 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 68]: From: ;tag=525491751 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 57]: To: "Patient" ;tag=as3d2643ca [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 11]: CSeq: 2 ACK [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 44]: Contact: [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 16]: Max-Forwards: 70 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 17]: Content-Length: 0 --- (10 headers 0 lines) --- [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (Checking From) --From tag 525491751 --To-tag as3d2643ca [May 27 17:35:46] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:27866 handle_incoming: **** Received ACK (6) - Command in SIP ACK [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:4523 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #432334 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' of Response 2: Match Found [May 27 17:35:46] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:78.26.144.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7dcce43e;rport=5060 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Call-ID: 1500096782 CSeq: 104 INVITE User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7dcce43e;rport=5060 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 53]: To: ;tag=896355678 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 104 INVITE [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 17]: Content-Length: 0 --- (8 headers 0 lines) --- [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking To) --From tag as025a484a --To-tag 896355678 [May 27 17:35:46] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:4590 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #432335 - INVITE (got response) [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:4597 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1500096782' Request 104: Found [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:22342 handle_response_invite: SIP response 100 to RE-invite on outgoing call 1500096782 [May 27 17:35:46] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:78.26.144.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7dcce43e;rport=5060 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Call-ID: 1500096782 CSeq: 104 INVITE Contact: Content-Type: application/sdp User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 317 v=0 o=6000444 1786 3816 IN IP4 78.26.144.10 s=Talk c=IN IP4 78.26.144.10 b=AS:380 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7077 IN IP4 192.168.1.107 m=video 9078 RTP/AVP 102 a=rtpmap:102 H264/90000 a=fmtp:102 profile-level-id=428014 <-------------> [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7dcce43e;rport=5060 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 53]: To: ;tag=896355678 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 104 INVITE [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 43]: Contact: [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 29]: Content-Type: application/sdp [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 19]: Content-Length: 317 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 0]: [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000444 1786 3816 IN IP4 78.26.144.10 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 3 [ 21]: c=IN IP4 78.26.144.10 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 6 [ 26]: m=audio 7076 RTP/AVP 9 101 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 7 [ 20]: a=rtpmap:9 G722/8000 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 9 [ 15]: a=fmtp:101 0-11 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 10 [ 32]: a=rtcp:7077 IN IP4 192.168.1.107 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 11 [ 24]: m=video 9078 RTP/AVP 102 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 12 [ 23]: a=rtpmap:102 H264/90000 [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9605 parse_request: Body 13 [ 34]: a=fmtp:102 profile-level-id=428014 --- (10 headers 14 lines) --- [May 27 17:35:46] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking To) --From tag as025a484a --To-tag 896355678 [May 27 17:35:46] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:4518 __sip_ack: Acked pending invite 104 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '1500096782' of Request 104: Match Found [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:22342 handle_response_invite: SIP response 200 to RE-invite on outgoing call 1500096782 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=6000444 1786 3816 IN IP4 78.26.144.10... OK. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [May 27 17:35:46] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '78.26.144.10' into... [May 27 17:35:46] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '78.26.144.10' and port ''. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 78.26.144.10... OK. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. Found RTP audio format 9 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa8980c4ad0 Found RTP audio format 101 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa8980c4ad0 Found audio description format G722 for ID 9 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. Found audio description format telephone-event for ID 101 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtcp:7077 IN IP4 192.168.1.107... UNSUPPORTED OR FAILED. Found RTP video format 102 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 102 based on m type on 0x7fa8980c8e20 Found video description format H264 for ID 102 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=rtpmap:102 H264/90000... OK. [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=fmtp:102 profile-level-id=428014... OK. Capabilities: us - (ulaw|g722|h264|vp8), peer - audio=(g722)/video=(h264)/text=(nothing), combined - (g722|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 78.26.144.10:7076 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa8980c4ad0 to 0x7fa8701bc660 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa8980c4ad0 to 0x7fa8701bc660 [May 27 17:35:46] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa8701bc498' Peer video RTP is at port 78.26.144.10:9078 [May 27 17:35:46] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 102 from 0x7fa8980c8e20 to 0x7fa8701cc190 [May 27 17:35:46] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa870202818' Peer doesn't provide T.140 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10668 process_sdp: We're settling with these formats: (g722|h264) [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10675 process_sdp: We have an owner, now see if we need to change this call [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:10681 process_sdp: Setting native formats after processing SDP. peer joint formats (g722|h264), old nativeformats (g722|h264) [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:6669 update_call_counter: Updating call counter for incoming call [May 27 17:35:46] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000444 [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 1500096782 set_destination: Parsing for address/port to send to [May 27 17:35:46] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000444 [May 27 17:35:46] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107' into... [May 27 17:35:46] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port ''. [May 27 17:35:46] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000444 - state 2 (In use) [May 27 17:35:46] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000444' state '2' set_destination: set destination to 192.168.1.107:5060 [May 27 17:35:46] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000444' changed to state '2' (In use) but we don't care because they're not a member of any queue. Transmitting (NAT) to 78.26.144.10:5060: ACK sip:linphone.iphone@192.168.1.107 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK7100159c;rport Max-Forwards: 70 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Contact: Call-ID: 1500096782 CSeq: 104 ACK User-Agent: Video Medicine PBX Content-Length: 0 --- [May 27 17:35:46] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'ACK sip:lin' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:46] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:78.26.144.10:5060 ---> REGISTER sip:vm.intersog.com:40412 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK246482778 From: ;tag=1180148849 To: Call-ID: 1620343365 CSeq: 13 REGISTER Contact: Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Expires: 60 Content-Length: 0 <-------------> [May 27 17:35:47] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 42]: REGISTER sip:vm.intersog.com:40412 SIP/2.0 [May 27 17:35:47] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK246482778 [May 27 17:35:47] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 56]: From: ;tag=1180148849 [May 27 17:35:47] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 39]: To: [May 27 17:35:47] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1620343365 [May 27 17:35:47] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 17]: CSeq: 13 REGISTER [May 27 17:35:47] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 61]: Contact: [May 27 17:35:47] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 16]: Max-Forwards: 70 [May 27 17:35:47] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:47] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 11]: Expires: 60 [May 27 17:35:47] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 17]: Content-Length: 0 --- (11 headers 0 lines) --- [May 27 17:35:47] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1620343365 (Checking From) --From tag 1180148849 --To-tag [May 27 17:35:47] DEBUG[28657]: acl.c:979 ast_ouraddrfor: For destination '78.26.144.10', our source address is '209.239.114.51'. [May 27 17:35:47] DEBUG[28657]: chan_sip.c:4021 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 209.239.114.51:5060 [May 27 17:35:47] DEBUG[28657]: chan_sip.c:8721 sip_alloc: Allocating new SIP dialog for 1620343365 - REGISTER (No RTP) [May 27 17:35:47] DEBUG[28657]: chan_sip.c:27866 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [May 27 17:35:47] DEBUG[28657]: chan_sip.c:27689 handle_request_register: Initializing initreq for method REGISTER - callid 1620343365 [May 27 17:35:47] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107:5060' into... [May 27 17:35:47] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port '5060'. [May 27 17:35:47] DEBUG[28657]: chan_sip.c:17845 check_via: NAT detected for 192.168.1.107:5060 / 78.26.144.10:5060 Sending to 78.26.144.10:5060 (NAT) [May 27 17:35:47] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:40412' into... [May 27 17:35:47] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port ''. <--- Transmitting (NAT) to 78.26.144.10:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK246482778;received=78.26.144.10;rport=5060 From: ;tag=1180148849 To: ;tag=as31c3413b Call-ID: 1620343365 CSeq: 13 REGISTER Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="vm.intersog.com", nonce="29617a4c" Content-Length: 0 <------------> [May 27 17:35:47] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 78.26.144.10:5060 Scheduling destruction of SIP dialog '1620343365' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:78.26.144.10:5060 ---> REGISTER sip:vm.intersog.com:40412 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK505217688 From: ;tag=1180148849 To: Call-ID: 1620343365 CSeq: 14 REGISTER Contact: Authorization: Digest username="6000444", realm="vm.intersog.com", nonce="29617a4c", uri="sip:vm.intersog.com:40412", response="44d0c3b7257edd450c1002976fd2bee8", algorithm=MD5 Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Expires: 60 Content-Length: 0 <-------------> [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 42]: REGISTER sip:vm.intersog.com:40412 SIP/2.0 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK505217688 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 56]: From: ;tag=1180148849 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 39]: To: [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1620343365 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 17]: CSeq: 14 REGISTER [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 61]: Contact: [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [176]: Authorization: Digest username="6000444", realm="vm.intersog.com", nonce="29617a4c", uri="sip:vm.intersog.com:40412", response="44d0c3b7257edd450c1002976fd2bee8", algorithm=MD5 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 16]: Max-Forwards: 70 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 11]: Expires: 60 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 11 [ 17]: Content-Length: 0 --- (12 headers 0 lines) --- [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1620343365 (Checking From) --From tag 1180148849 --To-tag [May 27 17:35:48] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:40412' into... [May 27 17:35:48] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port '40412'. [May 27 17:35:48] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:40412' into... [May 27 17:35:48] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port '40412'. [May 27 17:35:48] DEBUG[28657]: chan_sip.c:27866 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [May 27 17:35:48] DEBUG[28657]: chan_sip.c:27689 handle_request_register: Initializing initreq for method REGISTER - callid 1620343365 [May 27 17:35:48] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107:5060' into... [May 27 17:35:48] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port '5060'. [May 27 17:35:48] DEBUG[28657]: chan_sip.c:17845 check_via: NAT detected for 192.168.1.107:5060 / 78.26.144.10:5060 Sending to 78.26.144.10:5060 (NAT) [May 27 17:35:48] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:40412' into... [May 27 17:35:48] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port ''. [May 27 17:35:48] DEBUG[28657]: chan_sip.c:15927 parse_register_contact: Store REGISTER's src-IP:port for call routing. <--- Transmitting (NAT) to 78.26.144.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK505217688;received=78.26.144.10;rport=5060 From: ;tag=1180148849 To: ;tag=as31c3413b Call-ID: 1620343365 CSeq: 14 REGISTER Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Mon, 27 May 2013 17:35:48 GMT Content-Length: 0 <------------> [May 27 17:35:48] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 78.26.144.10:5060 Scheduling destruction of SIP dialog '1620343365' in 32000 ms (Method: REGISTER) [May 27 17:35:48] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000444 [May 27 17:35:48] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000444 [May 27 17:35:48] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000444 - state 2 (In use) [May 27 17:35:48] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000444' state '2' [May 27 17:35:48] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000444' changed to state '2' (In use) but we don't care because they're not a member of any queue. <--- SIP read from UDP:78.26.144.10:5060 ---> INFO sip:6000333@209.239.114.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1625566531 From: ;tag=896355678 To: "Doctor" ;tag=as025a484a Call-ID: 1500096782 CSeq: 23 INFO Contact: Content-Type: application/media_control+xml Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 185 <-------------> [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 44]: INFO sip:6000333@209.239.114.51:5060 SIP/2.0 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1625566531 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 55]: From: ;tag=896355678 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 56]: To: "Doctor" ;tag=as025a484a [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 13]: CSeq: 23 INFO [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 41]: Contact: [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 43]: Content-Type: application/media_control+xml [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 16]: Max-Forwards: 70 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 19]: Content-Length: 185 [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 11 [ 0]: [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9605 parse_request: Body 0 [172]: --- (11 headers 1 lines) --- [May 27 17:35:48] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking From) --From tag 896355678 --To-tag as025a484a [May 27 17:35:48] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:48] DEBUG[28657][C-0000019a]: chan_sip.c:27866 handle_incoming: **** Received INFO (13) - Command in SIP INFO Receiving INFO! <--- Transmitting (NAT) to 78.26.144.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK1625566531;received=78.26.144.10;rport=5060 From: ;tag=896355678 To: "Doctor" ;tag=as025a484a Call-ID: 1500096782 CSeq: 23 INFO Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [May 27 17:35:48] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:48] DEBUG[20587][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 set_destination: Parsing for address/port to send to [May 27 17:35:48] DEBUG[20587][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.93' into... [May 27 17:35:48] DEBUG[20587][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.93' and port ''. set_destination: set destination to 192.168.3.93:5060 Reliably Transmitting (NAT) to 89.209.100.91:5060: INFO sip:linphone.iphone@192.168.3.93 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK1cde901b;rport Max-Forwards: 70 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Contact: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 105 INFO User-Agent: Video Medicine PBX Content-Type: application/media_control+xml Content-Length: 205 --- [May 27 17:35:48] DEBUG[20587][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432339 [May 27 17:35:48] DEBUG[20587][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'INFO sip:li' onto UDP socket destined for 89.209.100.91:5060 [May 27 17:35:48] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:89.209.100.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK1cde901b;rport=5060 From: "Patient" ;tag=as3d2643ca To: ;tag=525491751 Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 105 INFO Contact: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:49] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [May 27 17:35:49] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK1cde901b;rport=5060 [May 27 17:35:49] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 59]: From: "Patient" ;tag=as3d2643ca [May 27 17:35:49] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 66]: To: ;tag=525491751 [May 27 17:35:49] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:49] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 14]: CSeq: 105 INFO [May 27 17:35:49] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 40]: Contact: [May 27 17:35:49] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:49] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 17]: Content-Length: 0 --- (9 headers 0 lines) --- [May 27 17:35:49] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (Checking To) --From tag as3d2643ca --To-tag 525491751 [May 27 17:35:49] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:49] DEBUG[28657][C-0000019a]: chan_sip.c:4523 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #432339 [May 27 17:35:49] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' of Request 105: Match Found [May 27 17:35:49] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:78.26.144.10:5060 ---> REGISTER sip:vm.intersog.com:40412 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK518071381 From: ;tag=1180148849 To: Call-ID: 1620343365 CSeq: 15 REGISTER Contact: Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Expires: 60 Content-Length: 0 <-------------> [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 42]: REGISTER sip:vm.intersog.com:40412 SIP/2.0 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK518071381 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 56]: From: ;tag=1180148849 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 39]: To: [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1620343365 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 17]: CSeq: 15 REGISTER [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 56]: Contact: [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 16]: Max-Forwards: 70 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 11]: Expires: 60 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 17]: Content-Length: 0 --- (11 headers 0 lines) --- [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1620343365 (Checking From) --From tag 1180148849 --To-tag [May 27 17:35:52] DEBUG[28657]: chan_sip.c:27866 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [May 27 17:35:52] DEBUG[28657]: chan_sip.c:27689 handle_request_register: Initializing initreq for method REGISTER - callid 1620343365 [May 27 17:35:52] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107:5060' into... [May 27 17:35:52] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port '5060'. [May 27 17:35:52] DEBUG[28657]: chan_sip.c:17845 check_via: NAT detected for 192.168.1.107:5060 / 78.26.144.10:5060 Sending to 78.26.144.10:5060 (NAT) [May 27 17:35:52] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:40412' into... [May 27 17:35:52] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port ''. <--- Transmitting (NAT) to 78.26.144.10:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK518071381;received=78.26.144.10;rport=5060 From: ;tag=1180148849 To: ;tag=as31c3413b Call-ID: 1620343365 CSeq: 15 REGISTER Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="vm.intersog.com", nonce="29969465" Content-Length: 0 <------------> [May 27 17:35:52] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 78.26.144.10:5060 Scheduling destruction of SIP dialog '1620343365' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:78.26.144.10:5060 ---> REGISTER sip:vm.intersog.com:40412 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1364102730 From: ;tag=1180148849 To: Call-ID: 1620343365 CSeq: 16 REGISTER Contact: Authorization: Digest username="6000444", realm="vm.intersog.com", nonce="29969465", uri="sip:vm.intersog.com:40412", response="c7ad41a80a11ee0026b6c6b2b7272712", algorithm=MD5 Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Expires: 60 Content-Length: 0 <-------------> [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 42]: REGISTER sip:vm.intersog.com:40412 SIP/2.0 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.1.107:5060;rport;branch=z9hG4bK1364102730 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 56]: From: ;tag=1180148849 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 39]: To: [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1620343365 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 17]: CSeq: 16 REGISTER [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 56]: Contact: [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [176]: Authorization: Digest username="6000444", realm="vm.intersog.com", nonce="29969465", uri="sip:vm.intersog.com:40412", response="c7ad41a80a11ee0026b6c6b2b7272712", algorithm=MD5 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 16]: Max-Forwards: 70 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 11]: Expires: 60 [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 11 [ 17]: Content-Length: 0 --- (12 headers 0 lines) --- [May 27 17:35:52] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1620343365 (Checking From) --From tag 1180148849 --To-tag [May 27 17:35:52] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:40412' into... [May 27 17:35:52] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port '40412'. [May 27 17:35:52] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:40412' into... [May 27 17:35:52] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port '40412'. [May 27 17:35:52] DEBUG[28657]: chan_sip.c:27866 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [May 27 17:35:52] DEBUG[28657]: chan_sip.c:27689 handle_request_register: Initializing initreq for method REGISTER - callid 1620343365 [May 27 17:35:52] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107:5060' into... [May 27 17:35:52] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port '5060'. [May 27 17:35:52] DEBUG[28657]: chan_sip.c:17845 check_via: NAT detected for 192.168.1.107:5060 / 78.26.144.10:5060 Sending to 78.26.144.10:5060 (NAT) [May 27 17:35:52] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:40412' into... [May 27 17:35:52] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port ''. [May 27 17:35:52] DEBUG[28657]: chan_sip.c:15927 parse_register_contact: Store REGISTER's src-IP:port for call routing. <--- Transmitting (NAT) to 78.26.144.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK1364102730;received=78.26.144.10;rport=5060 From: ;tag=1180148849 To: ;tag=as31c3413b Call-ID: 1620343365 CSeq: 16 REGISTER Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Mon, 27 May 2013 17:35:52 GMT Content-Length: 0 <------------> [May 27 17:35:52] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 78.26.144.10:5060 Scheduling destruction of SIP dialog '1620343365' in 32000 ms (Method: REGISTER) [May 27 17:35:52] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000444 [May 27 17:35:52] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000444 [May 27 17:35:52] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000444 - state 2 (In use) [May 27 17:35:52] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000444' state '2' [May 27 17:35:52] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000444' changed to state '2' (In use) but we don't care because they're not a member of any queue. [May 27 17:35:53] DEBUG[28657]: chan_sip.c:8721 sip_alloc: Allocating new SIP dialog for 2abf18e81a60f2db704bf95765862749@209.239.114.51:5060 - OPTIONS (No RTP) [May 27 17:35:53] DEBUG[28657]: acl.c:979 ast_ouraddrfor: For destination '92.113.50.205', our source address is '209.239.114.51'. [May 27 17:35:53] DEBUG[28657]: chan_sip.c:4021 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 209.239.114.51:5060 [May 27 17:35:53] DEBUG[28657]: chan_sip.c:3507 initialize_initreq: Initializing initreq for method OPTIONS - callid 6e45a9e569e1b903673cc72179e8d726@209.239.114.51:5060 [May 27 17:35:53] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 71]: OPTIONS sip:5000000248@92.113.50.205:36857;line=6666ca59cc8426b SIP/2.0 [May 27 17:35:53] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK45c83e3b;rport [May 27 17:35:53] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 16]: Max-Forwards: 70 [May 27 17:35:53] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 61]: From: "asterisk" ;tag=as14c1321f [May 27 17:35:53] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: To: [May 27 17:35:53] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 43]: Contact: [May 27 17:35:53] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 61]: Call-ID: 6e45a9e569e1b903673cc72179e8d726@209.239.114.51:5060 [May 27 17:35:53] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 17]: CSeq: 102 OPTIONS [May 27 17:35:53] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 30]: User-Agent: Video Medicine PBX [May 27 17:35:53] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 35]: Date: Mon, 27 May 2013 17:35:53 GMT [May 27 17:35:53] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 27 17:35:53] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 11 [ 26]: Supported: replaces, timer Reliably Transmitting (NAT) to 92.113.50.205:36857: OPTIONS sip:5000000248@92.113.50.205:36857;line=6666ca59cc8426b SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK45c83e3b;rport Max-Forwards: 70 From: "asterisk" ;tag=as14c1321f To: Contact: Call-ID: 6e45a9e569e1b903673cc72179e8d726@209.239.114.51:5060 CSeq: 102 OPTIONS User-Agent: Video Medicine PBX Date: Mon, 27 May 2013 17:35:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [May 27 17:35:53] DEBUG[28657]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432343 [May 27 17:35:53] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 92.113.50.205:36857 [May 27 17:35:54] DEBUG[28657]: chan_sip.c:4098 retrans_pkt: SIP TIMER: Not rescheduling id #432343:OPTIONS (Method 3) (No timer T1) Retransmitting #1 (NAT) to 92.113.50.205:36857: OPTIONS sip:5000000248@92.113.50.205:36857;line=6666ca59cc8426b SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK45c83e3b;rport Max-Forwards: 70 From: "asterisk" ;tag=as14c1321f To: Contact: Call-ID: 6e45a9e569e1b903673cc72179e8d726@209.239.114.51:5060 CSeq: 102 OPTIONS User-Agent: Video Medicine PBX Date: Mon, 27 May 2013 17:35:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [May 27 17:35:54] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 92.113.50.205:36857 <--- SIP read from UDP:89.209.100.91:5060 ---> BYE sip:6000444@209.239.114.51:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK68473946 From: ;tag=525491751 To: "Patient" ;tag=as3d2643ca Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 3 BYE Contact: Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:54] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 43]: BYE sip:6000444@209.239.114.51:5060 SIP/2.0 [May 27 17:35:54] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK68473946 [May 27 17:35:54] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 68]: From: ;tag=525491751 [May 27 17:35:54] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 57]: To: "Patient" ;tag=as3d2643ca [May 27 17:35:54] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:54] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 11]: CSeq: 3 BYE [May 27 17:35:54] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 40]: Contact: [May 27 17:35:54] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 16]: Max-Forwards: 70 [May 27 17:35:54] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:54] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 17]: Content-Length: 0 --- (10 headers 0 lines) --- [May 27 17:35:54] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 (Checking From) --From tag 525491751 --To-tag as3d2643ca [May 27 17:35:54] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:54] DEBUG[28657][C-0000019a]: chan_sip.c:27866 handle_incoming: **** Received BYE (8) - Command in SIP BYE [May 27 17:35:54] DEBUG[28657][C-0000019a]: chan_sip.c:26391 handle_request_bye: Initializing initreq for method BYE - callid 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:54] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.93:5060' into... [May 27 17:35:54] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.93' and port '5060'. [May 27 17:35:54] DEBUG[28657][C-0000019a]: chan_sip.c:17845 check_via: NAT detected for 192.168.3.93:5060 / 89.209.100.91:5060 Sending to 89.209.100.91:5060 (NAT) [May 27 17:35:54] DEBUG[28657][C-0000019a]: chan_sip.c:3520 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:54] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa874063918' [May 27 17:35:54] DEBUG[20587][C-0000019a]: rtp_engine.c:1315 remote_bridge_loop: Oooh, got a hangup [May 27 17:35:54] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3817 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7fa8701bc498' [May 27 17:35:54] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa8701cbfc8' [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:32432 sip_set_rtp_peer: Sending reinvite on SIP '1500096782' - It's audio soon redirected to IP 209.239.114.51:5060 [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 1500096782 set_destination: Parsing for address/port to send to [May 27 17:35:54] DEBUG[20587][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107' into... [May 27 17:35:54] DEBUG[20587][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port ''. set_destination: set destination to 192.168.1.107:5060 [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:13004 add_sdp: This call needs video offers! [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:13054 add_sdp: ** Our capability: (g722|h264) Video flag: False Text flag: True [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:13055 add_sdp: ** Our prefcodec: (nothing) Audio is at 11376 Video is at 209.239.114.51:13754 Adding codec 100012 (g722) to SDP Adding video codec 200004 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:13192 add_sdp: -- Done with adding codecs to SDP [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:13390 add_sdp: Done building SDP. Settling with this capability: (g722|h264) [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:3505 initialize_initreq: Initializing already initialized SIP dialog 1500096782 (presumably reinvite) [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 0 [ 48]: INVITE sip:linphone.iphone@192.168.1.107 SIP/2.0 [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK4448bb5e;rport [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 2 [ 16]: Max-Forwards: 70 [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 3 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 4 [ 53]: To: ;tag=896355678 [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 5 [ 42]: Contact: [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 6 [ 19]: Call-ID: 1500096782 [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 7 [ 16]: CSeq: 105 INVITE [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 8 [ 30]: User-Agent: Video Medicine PBX [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 10 [ 26]: Supported: replaces, timer [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:9568 parse_request: Header 12 [ 29]: Content-Type: application/sdp Reliably Transmitting (NAT) to 78.26.144.10:5060: INVITE sip:linphone.iphone@192.168.1.107 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK4448bb5e;rport Max-Forwards: 70 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Contact: Call-ID: 1500096782 CSeq: 105 INVITE User-Agent: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 374 v=0 o=root 426475338 426475343 IN IP4 209.239.114.51 s=Asterisk PBX 11.2.1 c=IN IP4 209.239.114.51 b=CT:384 t=0 0 m=audio 11376 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 13754 RTP/AVP 102 a=rtpmap:102 H264/90000 a=fmtp:102 profile-level-id=428014 a=sendrecv --- [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432345 [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:54] DEBUG[20587][C-0000019a]: channel.c:7960 ast_channel_bridge: Returning from native bridge, channels: SIP/6000444-000000af, SIP/6000333-000000b0 Scheduling destruction of SIP dialog '551a550202a0684a11641ae814d949d7@209.239.114.51:5060' in 32000 ms (Method: BYE) [May 27 17:35:54] DEBUG[28657][C-0000019a]: chan_sip.c:26493 handle_request_bye: Received bye, issuing owner hangup <--- Transmitting (NAT) to 89.209.100.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.93:5060;branch=z9hG4bK68473946;received=89.209.100.91;rport=5060 From: ;tag=525491751 To: "Patient" ;tag=as3d2643ca Call-ID: 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 CSeq: 3 BYE Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [May 27 17:35:54] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 89.209.100.91:5060 [May 27 17:35:54] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. [May 27 17:35:54] DEBUG[20587][C-0000019a]: cdr_mysql.c:336 mysql_log: Inserting a CDR record. [May 27 17:35:54] DEBUG[20587][C-0000019a]: cdr_mysql.c:339 mysql_log: SQL command as follows: INSERT INTO cdr (`calldate`,`clid`,`src`,`dst`,`dcontext`,`lastapp`,`lastdata`,`duration`,`billsec`,`disposition`,`channel`,`dstchannel`,`amaflags`,`uniqueid`,`answer`,`end`) VALUES ('2013-05-27 17:35:37','\"Patient\" <6000444>','6000444','6000333','webrtc','Dial','SIP/6000333,20','17.470993','14.924922','ANSWERED','SIP/6000444-000000af','SIP/6000333-000000b0','DOCUMENTATION','1369676137.176','2013-05-27 17:35:39','2013-05-27 17:35:54') [May 27 17:35:54] DEBUG[20587][C-0000019a]: channel.c:2836 ast_hangup: Hanging up channel 'SIP/6000333-000000b0' [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:7049 sip_hangup: Hangup call SIP/6000333-000000b0, SIP callid 551a550202a0684a11641ae814d949d7@209.239.114.51:5060 [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:7054 sip_hangup: update_call_counter(6000333) - decrement call limit counter on hangup [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:6669 update_call_counter: Updating call counter for outgoing call [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:6740 update_call_counter: Call to peer '6000333' removed from call limit 2147483647 [May 27 17:35:54] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa874063918' [May 27 17:35:54] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000333 [May 27 17:35:54] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000333 [May 27 17:35:54] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000333 - state 1 (Not in use) [May 27 17:35:54] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000333' state '1' [May 27 17:35:54] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000333 [May 27 17:35:54] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000333 [May 27 17:35:54] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000333 - state 1 (Not in use) [May 27 17:35:54] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000333' state '1' [May 27 17:35:54] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000333' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 27 17:35:54] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000333' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 27 17:35:54] DEBUG[20587][C-0000019a]: app_dial.c:3100 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [May 27 17:35:54] DEBUG[20587][C-0000019a]: pbx.c:6316 __ast_pbx_run: Spawn extension (webrtc,6000333,2) exited non-zero on 'SIP/6000444-000000af' == Spawn extension (webrtc, 6000333, 2) exited non-zero on 'SIP/6000444-000000af' [May 27 17:35:54] DEBUG[20587][C-0000019a]: channel.c:2657 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/6000444-000000af' [May 27 17:35:54] DEBUG[20587][C-0000019a]: channel.c:2836 ast_hangup: Hanging up channel 'SIP/6000444-000000af' [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:7049 sip_hangup: Hangup call SIP/6000444-000000af, SIP callid 1500096782 [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:7054 sip_hangup: update_call_counter(6000444) - decrement call limit counter on hangup [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:6669 update_call_counter: Updating call counter for incoming call [May 27 17:35:54] DEBUG[20587][C-0000019a]: chan_sip.c:6740 update_call_counter: Call from peer '6000444' removed from call limit 2147483647 [May 27 17:35:54] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa8701bc498' [May 27 17:35:54] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa8701cbfc8' [May 27 17:35:54] DEBUG[20587][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa870202818' Scheduling destruction of SIP dialog '1500096782' in 32000 ms (Method: INFO) [May 27 17:35:54] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000444 [May 27 17:35:54] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000444 [May 27 17:35:54] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000444 - state 1 (Not in use) [May 27 17:35:54] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000444' state '1' [May 27 17:35:54] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000444 [May 27 17:35:54] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000444 [May 27 17:35:54] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000444 - state 1 (Not in use) [May 27 17:35:54] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000444' state '1' [May 27 17:35:54] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000444' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 27 17:35:54] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000444' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from UDP:78.26.144.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK4448bb5e;rport=5060 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Call-ID: 1500096782 CSeq: 105 INVITE User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK4448bb5e;rport=5060 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 53]: To: ;tag=896355678 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 105 INVITE [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 17]: Content-Length: 0 --- (8 headers 0 lines) --- [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking To) --From tag as025a484a --To-tag 896355678 [May 27 17:35:55] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:4590 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #432345 - INVITE (got response) [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:4597 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1500096782' Request 105: Found [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:22342 handle_response_invite: SIP response 100 to RE-invite on outgoing call 1500096782 [May 27 17:35:55] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. <--- SIP read from UDP:78.26.144.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK4448bb5e;rport=5060 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Call-ID: 1500096782 CSeq: 105 INVITE Contact: Content-Type: application/sdp User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 317 v=0 o=6000444 1786 3817 IN IP4 78.26.144.10 s=Talk c=IN IP4 78.26.144.10 b=AS:380 t=0 0 m=audio 7076 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7077 IN IP4 192.168.1.107 m=video 9078 RTP/AVP 102 a=rtpmap:102 H264/90000 a=fmtp:102 profile-level-id=428014 <-------------> [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK4448bb5e;rport=5060 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 53]: To: ;tag=896355678 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 105 INVITE [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 43]: Contact: [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 29]: Content-Type: application/sdp [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 19]: Content-Length: 317 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 0]: [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000444 1786 3817 IN IP4 78.26.144.10 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 3 [ 21]: c=IN IP4 78.26.144.10 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 6 [ 26]: m=audio 7076 RTP/AVP 9 101 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 7 [ 20]: a=rtpmap:9 G722/8000 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 9 [ 15]: a=fmtp:101 0-11 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 10 [ 32]: a=rtcp:7077 IN IP4 192.168.1.107 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 11 [ 24]: m=video 9078 RTP/AVP 102 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Body 12 [ 23]: a=rtpmap:102 H264/90000 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9605 parse_request: Body 13 [ 34]: a=fmtp:102 profile-level-id=428014 --- (10 headers 14 lines) --- [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking To) --From tag as025a484a --To-tag 896355678 [May 27 17:35:55] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:4518 __sip_ack: Acked pending invite 105 [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '1500096782' of Request 105: Match Found [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:22342 handle_response_invite: SIP response 200 to RE-invite on outgoing call 1500096782 [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=6000444 1786 3817 IN IP4 78.26.144.10... OK. [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [May 27 17:35:55] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '78.26.144.10' into... [May 27 17:35:55] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '78.26.144.10' and port ''. [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 78.26.144.10... OK. [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED. [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. Found RTP audio format 9 [May 27 17:35:55] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa8980c4b50 Found RTP audio format 101 [May 27 17:35:55] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa8980c4b50 Found audio description format G722 for ID 9 [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. Found audio description format telephone-event for ID 101 [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtcp:7077 IN IP4 192.168.1.107... UNSUPPORTED OR FAILED. Found RTP video format 102 [May 27 17:35:55] DEBUG[28657][C-0000019a]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 102 based on m type on 0x7fa8980c8ea0 Found video description format H264 for ID 102 [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=rtpmap:102 H264/90000... OK. [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=fmtp:102 profile-level-id=428014... OK. Capabilities: us - (ulaw|g722|h264|vp8), peer - audio=(g722)/video=(h264)/text=(nothing), combined - (g722|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 27 17:35:55] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa8701bc498' Peer audio RTP is at port 78.26.144.10:7076 [May 27 17:35:55] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa8980c4b50 to 0x7fa8701bc660 [May 27 17:35:55] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa8980c4b50 to 0x7fa8701bc660 [May 27 17:35:55] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3817 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7fa8701bc498' [May 27 17:35:55] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa8701cbfc8' Peer video RTP is at port 78.26.144.10:9078 [May 27 17:35:55] DEBUG[28657][C-0000019a]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 102 from 0x7fa8980c8ea0 to 0x7fa8701cc190 [May 27 17:35:55] DEBUG[28657][C-0000019a]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa870202818' Peer doesn't provide T.140 [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:10668 process_sdp: We're settling with these formats: (g722|h264) [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:6669 update_call_counter: Updating call counter for incoming call [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 1500096782 set_destination: Parsing for address/port to send to [May 27 17:35:55] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107' into... [May 27 17:35:55] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port ''. set_destination: set destination to 192.168.1.107:5060 Transmitting (NAT) to 78.26.144.10:5060: ACK sip:linphone.iphone@192.168.1.107 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK1aea944f;rport Max-Forwards: 70 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Contact: Call-ID: 1500096782 CSeq: 105 ACK User-Agent: Video Medicine PBX Content-Length: 0 --- [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'ACK sip:lin' onto UDP socket destined for 78.26.144.10:5060 [May 27 17:35:55] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000444 [May 27 17:35:55] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000444 [May 27 17:35:55] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000444 - state 1 (Not in use) [May 27 17:35:55] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000444' state '1' [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:11802 reqprep: Strict routing enforced for session 1500096782 [May 27 17:35:55] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000444' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. set_destination: Parsing for address/port to send to [May 27 17:35:55] DEBUG[28657][C-0000019a]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.107' into... [May 27 17:35:55] DEBUG[28657][C-0000019a]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.107' and port ''. set_destination: set destination to 192.168.1.107:5060 Reliably Transmitting (NAT) to 78.26.144.10:5060: BYE sip:linphone.iphone@192.168.1.107 SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK5a694eda;rport Max-Forwards: 70 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Call-ID: 1500096782 CSeq: 106 BYE User-Agent: Video Medicine PBX Proxy-Authorization: Digest username="6000444", realm="vm.intersog.com", algorithm=MD5, uri="sip:209.239.114.51", nonce="7b87fadb", response="f325c87a9c1244fe1560014f55750810" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #432349 [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:3864 __sip_xmit: Trying to put 'BYE sip:lin' onto UDP socket destined for 78.26.144.10:5060 Scheduling destruction of SIP dialog '1500096782' in 32000 ms (Method: INFO) [May 27 17:35:55] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. [May 27 17:35:55] DEBUG[28657]: chan_sip.c:4412 __sip_autodestruct: Auto destroying SIP dialog '1783459290' [May 27 17:35:55] DEBUG[28657]: chan_sip.c:6817 sip_destroy: Destroying SIP dialog 1783459290 Really destroying SIP dialog '1783459290' Method: BYE [May 27 17:35:55] DEBUG[28657]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x7fa8701d3818' [May 27 17:35:55] DEBUG[28657]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x7fa8702ccaa8' [May 27 17:35:55] DEBUG[28657]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x7fa8702d1048' <--- SIP read from UDP:78.26.144.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK5a694eda;rport=5060 From: "Doctor" ;tag=as025a484a To: ;tag=896355678 Call-ID: 1500096782 CSeq: 106 BYE User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK5a694eda;rport=5060 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 58]: From: "Doctor" ;tag=as025a484a [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 53]: To: ;tag=896355678 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 19]: Call-ID: 1500096782 [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 13]: CSeq: 106 BYE [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 17]: Content-Length: 0 --- (8 headers 0 lines) --- [May 27 17:35:55] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 1500096782 (Checking To) --From tag as025a484a --To-tag 896355678 [May 27 17:35:55] DEBUG[28657][C-0000019a]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000019a] bound to thread. [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:4523 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #432349 [May 27 17:35:55] DEBUG[28657][C-0000019a]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '1500096782' of Request 106: Match Found [May 27 17:35:55] DEBUG[28657][C-0000019a]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000019a] being removed from thread. [May 27 17:35:55] DEBUG[28657]: chan_sip.c:6817 sip_destroy: Destroying SIP dialog 1500096782 Really destroying SIP dialog '1500096782' Method: INFO [May 27 17:35:55] DEBUG[28657]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x7fa8701bc498' [May 27 17:35:55] DEBUG[28657]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x7fa8701cbfc8' [May 27 17:35:55] DEBUG[28657]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x7fa870202818' [May 27 17:35:55] DEBUG[28657]: chan_sip.c:4412 __sip_autodestruct: Auto destroying SIP dialog '959752737' [May 27 17:35:55] DEBUG[28657]: chan_sip.c:6817 sip_destroy: Destroying SIP dialog 959752737 Really destroying SIP dialog '959752737' Method: REGISTER [May 27 17:35:55] DEBUG[28657]: chan_sip.c:4098 retrans_pkt: SIP TIMER: Not rescheduling id #432343:OPTIONS (Method 3) (No timer T1) Retransmitting #2 (NAT) to 92.113.50.205:36857: OPTIONS sip:5000000248@92.113.50.205:36857;line=6666ca59cc8426b SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK45c83e3b;rport Max-Forwards: 70 From: "asterisk" ;tag=as14c1321f To: Contact: Call-ID: 6e45a9e569e1b903673cc72179e8d726@209.239.114.51:5060 CSeq: 102 OPTIONS User-Agent: Video Medicine PBX Date: Mon, 27 May 2013 17:35:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [May 27 17:35:55] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 92.113.50.205:36857 [May 27 17:35:56] DEBUG[28657]: chan_sip.c:4098 retrans_pkt: SIP TIMER: Not rescheduling id #432343:OPTIONS (Method 3) (No timer T1) Retransmitting #3 (NAT) to 92.113.50.205:36857: OPTIONS sip:5000000248@92.113.50.205:36857;line=6666ca59cc8426b SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK45c83e3b;rport Max-Forwards: 70 From: "asterisk" ;tag=as14c1321f To: Contact: Call-ID: 6e45a9e569e1b903673cc72179e8d726@209.239.114.51:5060 CSeq: 102 OPTIONS User-Agent: Video Medicine PBX Date: Mon, 27 May 2013 17:35:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [May 27 17:35:56] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 92.113.50.205:36857 <--- SIP read from UDP:89.209.100.91:5060 ---> REGISTER sip:vm.intersog.com:56382 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK1483131460 From: ;tag=1581016424 To: Call-ID: 850242396 CSeq: 1 REGISTER Contact: Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Expires: 600 Content-Length: 0 <-------------> [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 42]: REGISTER sip:vm.intersog.com:56382 SIP/2.0 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK1483131460 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 56]: From: ;tag=1581016424 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 39]: To: [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 18]: Call-ID: 850242396 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 1 REGISTER [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 56]: Contact: [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 16]: Max-Forwards: 70 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 12]: Expires: 600 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 17]: Content-Length: 0 --- (11 headers 0 lines) --- [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 850242396 (Checking From) --From tag 1581016424 --To-tag [May 27 17:35:57] DEBUG[28657]: acl.c:979 ast_ouraddrfor: For destination '89.209.100.91', our source address is '209.239.114.51'. [May 27 17:35:57] DEBUG[28657]: chan_sip.c:4021 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 209.239.114.51:5060 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:8721 sip_alloc: Allocating new SIP dialog for 850242396 - REGISTER (No RTP) [May 27 17:35:57] DEBUG[28657]: chan_sip.c:27866 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [May 27 17:35:57] DEBUG[28657]: chan_sip.c:27689 handle_request_register: Initializing initreq for method REGISTER - callid 850242396 [May 27 17:35:57] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.93:5060' into... [May 27 17:35:57] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.93' and port '5060'. [May 27 17:35:57] DEBUG[28657]: chan_sip.c:17845 check_via: NAT detected for 192.168.3.93:5060 / 89.209.100.91:5060 Sending to 89.209.100.91:5060 (NAT) [May 27 17:35:57] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:56382' into... [May 27 17:35:57] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port ''. <--- Transmitting (NAT) to 89.209.100.91:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.93:5060;branch=z9hG4bK1483131460;received=89.209.100.91;rport=5060 From: ;tag=1581016424 To: ;tag=as4995aa90 Call-ID: 850242396 CSeq: 1 REGISTER Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="vm.intersog.com", nonce="5a674a70" Content-Length: 0 <------------> [May 27 17:35:57] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 89.209.100.91:5060 Scheduling destruction of SIP dialog '850242396' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:89.209.100.91:5060 ---> REGISTER sip:vm.intersog.com:56382 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK2115330110 From: ;tag=1581016424 To: Call-ID: 850242396 CSeq: 2 REGISTER Contact: Authorization: Digest username="6000333", realm="vm.intersog.com", nonce="5a674a70", uri="sip:vm.intersog.com:56382", response="f91b79c262ece744e5588c6e07ef5342", algorithm=MD5 Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Expires: 600 Content-Length: 0 <-------------> [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 42]: REGISTER sip:vm.intersog.com:56382 SIP/2.0 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK2115330110 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 56]: From: ;tag=1581016424 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 39]: To: [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 18]: Call-ID: 850242396 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 2 REGISTER [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 56]: Contact: [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [176]: Authorization: Digest username="6000333", realm="vm.intersog.com", nonce="5a674a70", uri="sip:vm.intersog.com:56382", response="f91b79c262ece744e5588c6e07ef5342", algorithm=MD5 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 16]: Max-Forwards: 70 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 12]: Expires: 600 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 11 [ 17]: Content-Length: 0 --- (12 headers 0 lines) --- [May 27 17:35:57] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 850242396 (Checking From) --From tag 1581016424 --To-tag [May 27 17:35:57] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:56382' into... [May 27 17:35:57] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port '56382'. [May 27 17:35:57] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:56382' into... [May 27 17:35:57] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port '56382'. [May 27 17:35:57] DEBUG[28657]: chan_sip.c:27866 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [May 27 17:35:57] DEBUG[28657]: chan_sip.c:27689 handle_request_register: Initializing initreq for method REGISTER - callid 850242396 [May 27 17:35:57] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.93:5060' into... [May 27 17:35:57] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.93' and port '5060'. [May 27 17:35:57] DEBUG[28657]: chan_sip.c:17845 check_via: NAT detected for 192.168.3.93:5060 / 89.209.100.91:5060 Sending to 89.209.100.91:5060 (NAT) [May 27 17:35:57] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:56382' into... [May 27 17:35:57] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port ''. [May 27 17:35:57] DEBUG[28657]: chan_sip.c:15927 parse_register_contact: Store REGISTER's src-IP:port for call routing. <--- Transmitting (NAT) to 89.209.100.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.93:5060;branch=z9hG4bK2115330110;received=89.209.100.91;rport=5060 From: ;tag=1581016424 To: ;tag=as4995aa90 Call-ID: 850242396 CSeq: 2 REGISTER Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 600 Contact: ;expires=600 Date: Mon, 27 May 2013 17:35:57 GMT Content-Length: 0 <------------> [May 27 17:35:57] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 89.209.100.91:5060 Scheduling destruction of SIP dialog '850242396' in 32000 ms (Method: REGISTER) [May 27 17:35:57] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000333 [May 27 17:35:57] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000333 [May 27 17:35:57] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000333 - state 1 (Not in use) [May 27 17:35:57] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000333' state '1' [May 27 17:35:57] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000333' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 27 17:35:57] DEBUG[28657]: chan_sip.c:4098 retrans_pkt: SIP TIMER: Not rescheduling id #432343:OPTIONS (Method 3) (No timer T1) Retransmitting #4 (NAT) to 92.113.50.205:36857: OPTIONS sip:5000000248@92.113.50.205:36857;line=6666ca59cc8426b SIP/2.0 Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK45c83e3b;rport Max-Forwards: 70 From: "asterisk" ;tag=as14c1321f To: Contact: Call-ID: 6e45a9e569e1b903673cc72179e8d726@209.239.114.51:5060 CSeq: 102 OPTIONS User-Agent: Video Medicine PBX Date: Mon, 27 May 2013 17:35:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [May 27 17:35:57] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 92.113.50.205:36857 [May 27 17:35:57] DEBUG[28657]: chan_sip.c:6817 sip_destroy: Destroying SIP dialog 6e45a9e569e1b903673cc72179e8d726@209.239.114.51:5060 Really destroying SIP dialog '6e45a9e569e1b903673cc72179e8d726@209.239.114.51:5060' Method: OPTIONS <--- SIP read from UDP:89.209.100.91:5060 ---> REGISTER sip:vm.intersog.com:56382 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK2128318610 From: ;tag=1581016424 To: Call-ID: 850242396 CSeq: 3 REGISTER Contact: Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Expires: 600 Content-Length: 0 <-------------> [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 42]: REGISTER sip:vm.intersog.com:56382 SIP/2.0 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK2128318610 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 56]: From: ;tag=1581016424 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 39]: To: [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 18]: Call-ID: 850242396 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 3 REGISTER [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 62]: Contact: [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [ 16]: Max-Forwards: 70 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 12]: Expires: 600 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 17]: Content-Length: 0 --- (11 headers 0 lines) --- [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 850242396 (Checking From) --From tag 1581016424 --To-tag [May 27 17:35:58] DEBUG[28657]: chan_sip.c:27866 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [May 27 17:35:58] DEBUG[28657]: chan_sip.c:27689 handle_request_register: Initializing initreq for method REGISTER - callid 850242396 [May 27 17:35:58] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.93:5060' into... [May 27 17:35:58] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.93' and port '5060'. [May 27 17:35:58] DEBUG[28657]: chan_sip.c:17845 check_via: NAT detected for 192.168.3.93:5060 / 89.209.100.91:5060 Sending to 89.209.100.91:5060 (NAT) [May 27 17:35:58] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:56382' into... [May 27 17:35:58] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port ''. <--- Transmitting (NAT) to 89.209.100.91:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.93:5060;branch=z9hG4bK2128318610;received=89.209.100.91;rport=5060 From: ;tag=1581016424 To: ;tag=as4995aa90 Call-ID: 850242396 CSeq: 3 REGISTER Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="vm.intersog.com", nonce="04925d73" Content-Length: 0 <------------> [May 27 17:35:58] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 89.209.100.91:5060 Scheduling destruction of SIP dialog '850242396' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:89.209.100.91:5060 ---> REGISTER sip:vm.intersog.com:56382 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK1785399379 From: ;tag=1581016424 To: Call-ID: 850242396 CSeq: 4 REGISTER Contact: Authorization: Digest username="6000333", realm="vm.intersog.com", nonce="04925d73", uri="sip:vm.intersog.com:56382", response="f33ae563f4a14cb27fb5dbe5ad20e9b7", algorithm=MD5 Max-Forwards: 70 User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Expires: 600 Content-Length: 0 <-------------> [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 0 [ 42]: REGISTER sip:vm.intersog.com:56382 SIP/2.0 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.3.93:5060;rport;branch=z9hG4bK1785399379 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 2 [ 56]: From: ;tag=1581016424 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 3 [ 39]: To: [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 4 [ 18]: Call-ID: 850242396 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 4 REGISTER [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 6 [ 62]: Contact: [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 7 [176]: Authorization: Digest username="6000333", realm="vm.intersog.com", nonce="04925d73", uri="sip:vm.intersog.com:56382", response="f33ae563f4a14cb27fb5dbe5ad20e9b7", algorithm=MD5 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 8 [ 16]: Max-Forwards: 70 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 9 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 10 [ 12]: Expires: 600 [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9568 parse_request: Header 11 [ 17]: Content-Length: 0 --- (12 headers 0 lines) --- [May 27 17:35:58] DEBUG[28657]: chan_sip.c:9118 find_call: = Looking for Call ID: 850242396 (Checking From) --From tag 1581016424 --To-tag [May 27 17:35:58] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:56382' into... [May 27 17:35:58] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port '56382'. [May 27 17:35:58] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:56382' into... [May 27 17:35:58] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port '56382'. [May 27 17:35:58] DEBUG[28657]: chan_sip.c:27866 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [May 27 17:35:58] DEBUG[28657]: chan_sip.c:27689 handle_request_register: Initializing initreq for method REGISTER - callid 850242396 [May 27 17:35:58] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.93:5060' into... [May 27 17:35:58] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.93' and port '5060'. [May 27 17:35:58] DEBUG[28657]: chan_sip.c:17845 check_via: NAT detected for 192.168.3.93:5060 / 89.209.100.91:5060 Sending to 89.209.100.91:5060 (NAT) [May 27 17:35:58] DEBUG[28657]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'vm.intersog.com:56382' into... [May 27 17:35:58] DEBUG[28657]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'vm.intersog.com' and port ''. [May 27 17:35:58] DEBUG[28657]: chan_sip.c:15927 parse_register_contact: Store REGISTER's src-IP:port for call routing. <--- Transmitting (NAT) to 89.209.100.91:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.93:5060;branch=z9hG4bK1785399379;received=89.209.100.91;rport=5060 From: ;tag=1581016424 To: ;tag=as4995aa90 Call-ID: 850242396 CSeq: 4 REGISTER Server: Video Medicine PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 600 Contact: ;expires=600 Date: Mon, 27 May 2013 17:35:58 GMT Content-Length: 0 <------------> [May 27 17:35:58] DEBUG[28657]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 89.209.100.91:5060 Scheduling destruction of SIP dialog '850242396' in 32000 ms (Method: REGISTER) [May 27 17:35:58] DEBUG[28644]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000333 [May 27 17:35:58] DEBUG[28644]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000333 [May 27 17:35:58] DEBUG[28644]: devicestate.c:467 do_state_change: Changing state for SIP/6000333 - state 1 (Not in use) [May 27 17:35:58] DEBUG[28644]: devicestate.c:442 devstate_event: device 'SIP/6000333' state '1' [May 27 17:35:58] DEBUG[28686]: app_queue.c:1804 handle_statechange: Device 'SIP/6000333' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. usloft1046*CLI>