[Jan 21 18:26:55] Asterisk 13.7.0 built by root @ debian1 on a i686 running Linux on 2016-01-21 14:51:32 UTC [Jan 21 18:26:55] DEBUG[31478] config.c: Parsing /etc/asterisk/logger.conf [Jan 21 18:26:57] DEBUG[31459] chan_sip.c: Auto destroying SIP dialog 'h0htmQ92Vl23gn42ztp24XMJuaTqlS6P' [Jan 21 18:26:57] DEBUG[31459] chan_sip.c: Destroying SIP dialog h0htmQ92Vl23gn42ztp24XMJuaTqlS6P [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 0 [ 35]: INVITE sip:123@10.24.17.195 SIP/2.0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjOe3s44qX-pUd83nE66aftfiMEP-VbR5x [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 3 [ 71]: From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 4 [ 26]: To: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 5 [ 44]: Contact: "333" [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 6 [ 41]: Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 7 [ 18]: CSeq: 27930 INVITE [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 12 [ 36]: User-Agent: Digium D40 1_4_2_0_63880 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 14 [ 19]: Content-Length: 430 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 15 [ 0]: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 0 [ 3]: v=0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 1 [ 42]: o=- 148793728 148793728 IN IP4 10.24.18.16 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 2 [ 8]: s=digphn [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 6 [ 46]: m=audio 4038 RTP/AVP 111 18 0 58 118 9 8 58 96 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 7 [ 30]: a=rtcp:4039 IN IP4 10.24.18.16 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 8 [ 25]: a=rtpmap:111 G726-32/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 9 [ 21]: a=rtpmap:18 G729/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 10 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 11 [ 21]: a=rtpmap:58 L16/16000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 12 [ 21]: a=rtpmap:118 L16/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 13 [ 20]: a=rtpmap:9 G722/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 14 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 15 [ 25]: a=rtpmap:58 L16-256/16000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 16 [ 10]: a=sendrecv [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 17 [ 32]: a=rtpmap:96 telephone-event/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 18 [ 14]: a=fmtp:96 0-15 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: = Looking for Call ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f (Checking From) --From tag 9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod --To-tag [Jan 21 18:27:17] DEBUG[31459] acl.c: For destination '10.24.18.16', our source address is '10.24.17.195'. [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31459] netsock2.c: Splitting '10.24.18.16:5060' into... [Jan 21 18:27:17] DEBUG[31459] netsock2.c: ...host '10.24.18.16' and port '5060'. [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Allocating new SIP dialog for FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f - INVITE (No RTP) [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 21 18:27:17] DEBUG[31459][C-00000000] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, timer, norefersub" [Jan 21 18:27:17] DEBUG[31459][C-00000000] sip/reqresp_parser.c: Found SIP option: -replaces- [Jan 21 18:27:17] DEBUG[31459][C-00000000] sip/reqresp_parser.c: Matched SIP option: replaces [Jan 21 18:27:17] DEBUG[31459][C-00000000] sip/reqresp_parser.c: Found SIP option: -100rel- [Jan 21 18:27:17] DEBUG[31459][C-00000000] sip/reqresp_parser.c: Matched SIP option: 100rel [Jan 21 18:27:17] DEBUG[31459][C-00000000] sip/reqresp_parser.c: Found SIP option: -timer- [Jan 21 18:27:17] DEBUG[31459][C-00000000] sip/reqresp_parser.c: Matched SIP option: timer [Jan 21 18:27:17] DEBUG[31459][C-00000000] sip/reqresp_parser.c: Found SIP option: -norefersub- [Jan 21 18:27:17] DEBUG[31459][C-00000000] sip/reqresp_parser.c: Matched SIP option: norefersub [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.17.195' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.17.195' and port ''. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.16:5060 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 0 [ 32]: ACK sip:123@10.24.17.195 SIP/2.0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjOe3s44qX-pUd83nE66aftfiMEP-VbR5x [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 3 [ 71]: From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 4 [ 41]: To: ;tag=as428dd4e7 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 5 [ 41]: Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 6 [ 15]: CSeq: 27930 ACK [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: = Looking for Call ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f (Checking From) --From tag 9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod --To-tag as428dd4e7 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Stopping retransmission on 'FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f' of Response 27930: Match Found [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 0 [ 35]: INVITE sip:123@10.24.17.195 SIP/2.0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjEFgx3zb40XYC8kUZHSSbZV1teC.bjvhN [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 3 [ 71]: From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 4 [ 26]: To: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 5 [ 44]: Contact: "333" [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 6 [ 41]: Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 7 [ 18]: CSeq: 27931 INVITE [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 12 [ 36]: User-Agent: Digium D40 1_4_2_0_63880 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 13 [160]: Authorization: Digest username="333", realm="asterisk", nonce="365d7cc9", uri="sip:123@10.24.17.195", response="574a419b0e669568763a7aa93e66f381", algorithm=MD5 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 15 [ 19]: Content-Length: 430 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 16 [ 0]: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 0 [ 3]: v=0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 1 [ 42]: o=- 148793728 148793728 IN IP4 10.24.18.16 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 2 [ 8]: s=digphn [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 6 [ 46]: m=audio 4038 RTP/AVP 111 18 0 58 118 9 8 58 96 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 7 [ 30]: a=rtcp:4039 IN IP4 10.24.18.16 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 8 [ 25]: a=rtpmap:111 G726-32/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 9 [ 21]: a=rtpmap:18 G729/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 10 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 11 [ 21]: a=rtpmap:58 L16/16000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 12 [ 21]: a=rtpmap:118 L16/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 13 [ 20]: a=rtpmap:9 G722/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 14 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 15 [ 25]: a=rtpmap:58 L16-256/16000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 16 [ 10]: a=sendrecv [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 17 [ 32]: a=rtpmap:96 telephone-event/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 18 [ 14]: a=fmtp:96 0-15 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: = Looking for Call ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f (Checking From) --From tag 9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod --To-tag [Jan 21 18:27:17] DEBUG[31459] netsock2.c: Splitting '10.24.17.195' into... [Jan 21 18:27:17] DEBUG[31459] netsock2.c: ...host '10.24.17.195' and port ''. [Jan 21 18:27:17] DEBUG[31459] netsock2.c: Splitting '10.24.17.195' into... [Jan 21 18:27:17] DEBUG[31459] netsock2.c: ...host '10.24.17.195' and port ''. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.17.195' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.17.195' and port ''. [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9ad357c' [Jan 21 18:27:17] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Allocated port 19876 for RTP instance '0x9ad357c' [Jan 21 18:27:17] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Creating ICE session 0.0.0.0:19876 (19876) for RTP instance '0x9ad357c' [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.17.195' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.17.195' and port ''. [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: RTP instance '0x9ad357c' is setup and ready to go [Jan 21 18:27:17] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9ad357c' [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP o=- 148793728 148793728 IN IP4 10.24.18.16... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.16' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.16' and port ''. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.16... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Setting tx payload type 111 based on m type on 0xb3680ac0 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Setting tx payload type 18 based on m type on 0xb3680ac0 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Setting tx payload type 0 based on m type on 0xb3680ac0 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Don't have a default tx payload type 58 format for m type on 0xb3680ac0 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Setting tx payload type 118 based on m type on 0xb3680ac0 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Setting tx payload type 9 based on m type on 0xb3680ac0 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb3680ac0 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Don't have a default tx payload type 58 format for m type on 0xb3680ac0 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Don't have a default tx payload type 96 format for m type on 0xb3680ac0 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4039 IN IP4 10.24.18.16... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 G726-32/8000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:58 L16/16000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:118 L16/8000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:58 L16-256/16000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9ad357c' [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 0 (0x984bf84) from 0xb3680ac0 to 0x9ad3728 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 8 (0x9acc1dc) from 0xb3680ac0 to 0x9ad3728 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 9 (0x9d539a4) from 0xb3680ac0 to 0x9ad3728 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 18 (0x9ac6e6c) from 0xb3680ac0 to 0x9ad3728 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 58 (0x9acc204) from 0xb3680ac0 to 0x9ad3728 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 96 (0x9acc3ac) from 0xb3680ac0 to 0x9ad3728 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 111 (0x9c723c4) from 0xb3680ac0 to 0x9ad3728 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 118 (0x9d5397c) from 0xb3680ac0 to 0x9ad3728 [Jan 21 18:27:17] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x9ad357c' [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: We're settling with these formats: (g722|alaw) [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Checking SIP call limits for device 333 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Updating call counter for incoming call [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.17.195' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.17.195' and port ''. [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.17.195' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.17.195' and port ''. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Incoming INVITE with 'timer' option supported [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: INVITE also has "Session-Expires" header. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Session-Expires: 1800 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: INVITE also has "Min-SE" header. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Received Min-SE: 90 [Jan 21 18:27:17] DEBUG[31415] threadpool.c: Increasing threadpool stasis-core's size by 1 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: *** Our native formats are (g722) [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: *** Joint capabilities are (g722|alaw) [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: *** Our capabilities are (g722|alaw) [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: *** AST_CODEC_CHOOSE formats are g722 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: This channel will not be able to handle video. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: SIP/333-00000000: New call is still down.... Trying... [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.24.18.16:5060 [Jan 21 18:27:17] DEBUG[31422] devicestate.c: No provider found, checking channel drivers for SIP - 333 [Jan 21 18:27:17] DEBUG[31422] chan_sip.c: Checking device state for peer 333 [Jan 21 18:27:17] DEBUG[31422] devicestate.c: Changing state for SIP/333 - state 1 (Not in use) [Jan 21 18:27:17] DEBUG[31480][C-00000000] pbx.c: Launching 'NoOp' [Jan 21 18:27:17] DEBUG[31480][C-00000000] pbx.c: Launching 'Set' [Jan 21 18:27:17] DEBUG[31480][C-00000000] pbx.c: Function CALLERID(number) result is '00333' [Jan 21 18:27:17] DEBUG[31480][C-00000000] pbx.c: Launching 'Set' [Jan 21 18:27:17] DEBUG[31480][C-00000000] pbx.c: Function CALLERID(number) result is '00333' [Jan 21 18:27:17] DEBUG[31480][C-00000000] pbx.c: Launching 'Set' [Jan 21 18:27:17] DEBUG[31480][C-00000000] pbx.c: Result of 'EXTEN' is '123' [Jan 21 18:27:17] DEBUG[31480][C-00000000] pbx.c: Launching 'Dial' [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Asked to create a SIP channel with formats: (g722) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Allocating new SIP dialog for 6a863735370ef17b00d77c0d0ad3dae7@127.0.1.1:5060 - INVITE (No RTP) [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9ace6e4' [Jan 21 18:27:17] DEBUG[31480][C-00000000] res_rtp_asterisk.c: Allocated port 16704 for RTP instance '0x9ace6e4' [Jan 21 18:27:17] DEBUG[31480][C-00000000] res_rtp_asterisk.c: Creating ICE session 0.0.0.0:16704 (16704) for RTP instance '0x9ace6e4' [Jan 21 18:27:17] DEBUG[31480][C-00000000] netsock2.c: Splitting '10.24.17.195' into... [Jan 21 18:27:17] DEBUG[31480][C-00000000] netsock2.c: ...host '10.24.17.195' and port ''. [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: RTP instance '0x9ace6e4' is setup and ready to go [Jan 21 18:27:17] DEBUG[31480][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9ace6e4' [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jan 21 18:27:17] DEBUG[31480][C-00000000] acl.c: For destination '10.24.18.190', our source address is '10.24.17.195'. [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: SIP call-id changed from '6a863735370ef17b00d77c0d0ad3dae7@127.0.1.1:5060' to '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: *** Our native formats are (g722) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: *** Joint capabilities are (g722) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: *** Our capabilities are (g722|alaw) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: *** AST_CODEC_CHOOSE formats are g722 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: *** Our preferred formats from the incoming channel are (g722) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: This channel will not be able to handle video. [Jan 21 18:27:17] DEBUG[31480][C-00000000] channel_internal_api.c: Channel Call ID changing from [C-00000000] to [C-00000000] [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: Copying payload 0 (0x984bf84) from 0x9ad3728 to 0x9ace890 [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: Copying payload 8 (0x9acc1dc) from 0x9ad3728 to 0x9ace890 [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: Copying payload 9 (0x9d539a4) from 0x9ad3728 to 0x9ace890 [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: Copying payload 18 (0x9ac6e6c) from 0x9ad3728 to 0x9ace890 [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: Copying payload 58 (0x9acc204) from 0x9ad3728 to 0x9ace890 [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: Copying payload 96 (0x9acc3ac) from 0x9ad3728 to 0x9ace890 [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: Copying payload 111 (0x9c723c4) from 0x9ad3728 to 0x9ace890 [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: Copying payload 118 (0x9d5397c) from 0x9ad3728 to 0x9ace890 [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: Seeded SDP of 'SIP/OutTrunk-00000001' with that of 'SIP/333-00000000' [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Outgoing Call for 123 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Updating call counter for outgoing call [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: ** Our capability: (g722|alaw) Video flag: False Text flag: False [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: ** Our prefcodec: (g722) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (g722|alaw) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 0 [ 35]: INVITE sip:123@10.24.18.190 SIP/2.0 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK68ab447c [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 3 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 4 [ 26]: To: [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 5 [ 36]: Contact: [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 6 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.7.0 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 9 [ 35]: Date: Fri, 22 Jan 2016 00:27:17 GMT [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.190:5060 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK68ab447c;received=10.24.17.195 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 2 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 3 [ 26]: To: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 4 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 6 [ 31]: Server: Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 10 [ 36]: Contact: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: = Looking for Call ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (Checking To) --From tag as579facad --To-tag [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' Request 102: Found [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: SIP response 100 to standard invite [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 0 [ 28]: SIP/2.0 183 Session Progress [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK68ab447c;received=10.24.17.195 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 2 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 3 [ 41]: To: ;tag=as1e3a359e [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 4 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 6 [ 31]: Server: Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 10 [ 36]: Contact: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 12 [ 14]: Require: timer [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 13 [ 19]: Content-Length: 240 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 14 [ 0]: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 0 [ 3]: v=0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 1 [ 48]: o=root 1696456408 1696456408 IN IP4 10.24.18.190 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 2 [ 25]: s=Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.190 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 5 [ 26]: m=audio 13926 RTP/AVP 8 96 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 7 [ 32]: a=rtpmap:96 telephone-event/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 8 [ 14]: a=fmtp:96 0-16 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 9 [ 14]: a=maxptime:150 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: = Looking for Call ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (Checking To) --From tag as579facad --To-tag as1e3a359e [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' Request 102: Found [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: SIP response 183 to standard invite [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP o=root 1696456408 1696456408 IN IP4 10.24.18.190... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP s=Asterisk PBX 13.7.0-rc3... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port ''. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.190... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb367fc50 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Don't have a default tx payload type 96 format for m type on 0xb367fc50 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-16... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Set role to CONTROLLING (0x9ace6e4) [Jan 21 18:27:17] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Set role failed; no ice instance (0x9ace6e4) [Jan 21 18:27:17] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9ace6e4' [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 8 (0x9adb48c) from 0xb367fc50 to 0x9ace890 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 96 (0x9adb4b4) from 0xb367fc50 to 0x9ace890 [Jan 21 18:27:17] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x9ace6e4' [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: We're settling with these formats: (alaw) [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (g722) [Jan 21 18:27:17] DEBUG[31459][C-00000000] channel.c: Channel SIP/OutTrunk-00000001 setting read format path: alaw -> g722 [Jan 21 18:27:17] DEBUG[31459][C-00000000] channel.c: Channel SIP/OutTrunk-00000001 setting write format path: g722 -> alaw [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK68ab447c;received=10.24.17.195 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 2 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 3 [ 41]: To: ;tag=as1e3a359e [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 4 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 6 [ 31]: Server: Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 10 [ 36]: Contact: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 12 [ 14]: Require: timer [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 13 [ 19]: Content-Length: 240 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 14 [ 0]: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 0 [ 3]: v=0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 1 [ 48]: o=root 1696456408 1696456408 IN IP4 10.24.18.190 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 2 [ 25]: s=Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.190 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 5 [ 26]: m=audio 13926 RTP/AVP 8 96 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 7 [ 32]: a=rtpmap:96 telephone-event/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 8 [ 14]: a=fmtp:96 0-16 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 9 [ 14]: a=maxptime:150 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: = Looking for Call ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (Checking To) --From tag as579facad --To-tag as1e3a359e [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Acked pending invite 102 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Stopping retransmission on '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' of Request 102: Match Found [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: SIP response 200 to standard invite [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Call 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 responded to our reinvite without changing SDP version; ignoring SDP. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Updating call counter for outgoing call [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190:5060' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port '5060'. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Session-Expires: 1800 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Refresher: UAS [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Session timer started: 2 - 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 1768000ms [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Strict routing enforced for session 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190:5060' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port '5060'. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Trying to put 'ACK sip:123' onto UDP socket destined for 10.24.18.190:5060 [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: Setting early bridge SDP of 'SIP/333-00000000' with that of 'SIP/OutTrunk-00000001' [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: ** Our capability: (g722|alaw) Video flag: True Text flag: True [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Setting framing on incoming call: 0 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (g722|alaw) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 10.24.18.16:5060 [Jan 21 18:27:17] DEBUG[31422] devicestate.c: No provider found, checking channel drivers for SIP - OutTrunk [Jan 21 18:27:17] DEBUG[31422] chan_sip.c: Checking device state for peer OutTrunk [Jan 21 18:27:17] DEBUG[31422] devicestate.c: Changing state for SIP/OutTrunk - state 1 (Not in use) [Jan 21 18:27:17] DEBUG[31473] app_queue.c: Device 'SIP/OutTrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 21 18:27:17] DEBUG[31480][C-00000000] rtp_engine.c: Setting early bridge SDP of 'SIP/333-00000000' with that of 'SIP/OutTrunk-00000001' [Jan 21 18:27:17] DEBUG[31480][C-00000000] channel.c: Channel SIP/OutTrunk-00000001 setting read format path: alaw -> slin16 [Jan 21 18:27:17] DEBUG[31480][C-00000000] channel.c: Channel SIP/333-00000000 setting write format path: slin16 -> g722 [Jan 21 18:27:17] DEBUG[31480][C-00000000] channel.c: Channel SIP/333-00000000 setting read format path: g722 -> slin16 [Jan 21 18:27:17] DEBUG[31480][C-00000000] channel.c: Channel SIP/OutTrunk-00000001 setting write format path: slin16 -> alaw [Jan 21 18:27:17] DEBUG[31422] devicestate.c: No provider found, checking channel drivers for SIP - 333 [Jan 21 18:27:17] DEBUG[31422] chan_sip.c: Checking device state for peer 333 [Jan 21 18:27:17] DEBUG[31422] devicestate.c: Changing state for SIP/333 - state 1 (Not in use) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: SIP answering channel: SIP/333-00000000 [Jan 21 18:27:17] DEBUG[31480][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: ** Our capability: (g722|alaw) Video flag: True Text flag: True [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Setting framing on incoming call: 0 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (g722|alaw) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.16:5060 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Session timer started: 9 - FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f 900000ms [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge_native_rtp.c: Bridge '61ad6afe-27f5-44a8-aedf-846e5ba29d0f' can not use native RTP bridge as two channels are required [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Chose bridge technology simple_bridge [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: calling simple_bridge technology constructor [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: calling simple_bridge technology start [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge_channel.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: 0x9ae2c2c(SIP/OutTrunk-00000001) is joining [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge_channel.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: pushing 0x9ae2c2c(SIP/OutTrunk-00000001) [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge_native_rtp.c: Bridge '61ad6afe-27f5-44a8-aedf-846e5ba29d0f' can not use native RTP bridge as two channels are required [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge.c: Chose bridge technology simple_bridge [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f is already using the new technology. [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: 0x9ae2c2c(SIP/OutTrunk-00000001) is joining simple_bridge technology [Jan 21 18:27:17] DEBUG[31481][C-00000000] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge_channel.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: 0x9ae37c4(SIP/333-00000000) is joining [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge_channel.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: pushing 0x9ae37c4(SIP/333-00000000) [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Chose bridge technology native_rtp [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: calling native_rtp technology constructor [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: moving 0x9ae2c2c(SIP/OutTrunk-00000001) to dummy bridge temporarily [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: 0x9ae2c2c(SIP/OutTrunk-00000001) is leaving simple_bridge technology (dummy) [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: calling simple_bridge technology stop [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: 0x9ae37c4(SIP/333-00000000) is joining native_rtp technology [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Deferring reinvite on SIP 'FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f' - It's audio will be redirected to IP 10.24.18.190:13926 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Sending reinvite on SIP '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' - It's audio soon redirected to IP 10.24.18.16:4038 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Strict routing enforced for session 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31480][C-00000000] netsock2.c: Splitting '10.24.18.190:5060' into... [Jan 21 18:27:17] DEBUG[31480][C-00000000] netsock2.c: ...host '10.24.18.190' and port '5060'. [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: ** Our native-bridge filtered capablity: (alaw) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: ** Our prefcodec: (g722) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Initializing already initialized SIP dialog 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (presumably reinvite) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 0 [ 40]: INVITE sip:123@10.24.18.190:5060 SIP/2.0 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3e43084f [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 3 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 4 [ 41]: To: ;tag=as1e3a359e [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 5 [ 36]: Contact: [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 6 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.7.0 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 11 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8 [Jan 21 18:27:17] DEBUG[31480][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.190:5060 [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: 0x9ae2c2c(SIP/OutTrunk-00000001) is joining native_rtp technology [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: calling native_rtp technology start [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: calling simple_bridge technology destructor [Jan 21 18:27:17] DEBUG[31480][C-00000000] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Chose bridge technology native_rtp [Jan 21 18:27:17] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f is already using the new technology. [Jan 21 18:27:17] DEBUG[31428] cdr.c: Finalized CDR for SIP/OutTrunk-00000001 - start 1453422437.945158 answer 1453422437.959182 end 1453422437.963872 dispo ANSWERED [Jan 21 18:27:17] DEBUG[31415] threadpool.c: Increasing threadpool stasis-core's size by 1 [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge.c: Chose bridge technology native_rtp [Jan 21 18:27:17] DEBUG[31481][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f is already using the new technology. [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3e43084f;received=10.24.17.195 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 2 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 3 [ 41]: To: ;tag=as1e3a359e [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 4 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 6 [ 31]: Server: Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 10 [ 36]: Contact: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: = Looking for Call ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (Checking To) --From tag as579facad --To-tag as1e3a359e [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: *** SIP TIMER: Cancelling retransmission #8 - INVITE (got response) [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' Request 103: Found [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: SIP response 100 to RE-invite on outgoing call 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3e43084f;received=10.24.17.195 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 2 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 3 [ 41]: To: ;tag=as1e3a359e [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 4 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 6 [ 31]: Server: Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 10 [ 36]: Contact: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 12 [ 14]: Require: timer [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 13 [ 19]: Content-Length: 240 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Header 14 [ 0]: [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 0 [ 3]: v=0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 1 [ 48]: o=root 1696456408 1696456409 IN IP4 10.24.18.190 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 2 [ 25]: s=Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.190 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 5 [ 26]: m=audio 13926 RTP/AVP 8 96 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 7 [ 32]: a=rtpmap:96 telephone-event/8000 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 8 [ 14]: a=fmtp:96 0-16 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 9 [ 14]: a=maxptime:150 [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jan 21 18:27:17] DEBUG[31459] chan_sip.c: = Looking for Call ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (Checking To) --From tag as579facad --To-tag as1e3a359e [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Acked pending invite 103 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Stopping retransmission on '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' of Request 103: Match Found [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: SIP response 200 to RE-invite on outgoing call 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP o=root 1696456408 1696456409 IN IP4 10.24.18.190... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP s=Asterisk PBX 13.7.0-rc3... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port ''. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.190... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb367fc60 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Don't have a default tx payload type 96 format for m type on 0xb367fc60 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-16... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 21 18:27:17] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Set role to CONTROLLING (0x9ace6e4) [Jan 21 18:27:17] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Set role failed; no ice instance (0x9ace6e4) [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 8 (0x9ae2efc) from 0xb367fc60 to 0x9ace890 [Jan 21 18:27:17] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 96 (0x9ae129c) from 0xb367fc60 to 0x9ace890 [Jan 21 18:27:17] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9ace6e4' [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: We're settling with these formats: (alaw) [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Updating call counter for outgoing call [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190:5060' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port '5060'. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Session-Expires: 1800 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Refresher: UAS [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Session timer stopped: 2 - 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Session timer started: 11 - 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 1768000ms [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Strict routing enforced for session 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190:5060' into... [Jan 21 18:27:17] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port '5060'. [Jan 21 18:27:17] DEBUG[31459][C-00000000] chan_sip.c: Trying to put 'ACK sip:123' onto UDP socket destined for 10.24.18.190:5060 [Jan 21 18:27:18] DEBUG[31480][C-00000000] res_rtp_asterisk.c: 0x9adcb50 -- Probation learning mode pass with source address 10.24.18.16:4038 [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31481][C-00000000] res_rtp_asterisk.c: Ooh, format changed from none to alaw [Jan 21 18:27:18] DEBUG[31481][C-00000000] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x9ace6e4' [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 0 [ 37]: ACK sip:123@10.24.17.195:5060 SIP/2.0 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPj8oprM1L1inGkvR3aWT7DvD2s9ZmUvvtZ [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 3 [ 71]: From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 4 [ 41]: To: ;tag=as1d33b1c9 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 5 [ 41]: Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 6 [ 15]: CSeq: 27931 ACK [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: = Looking for Call ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f (Checking From) --From tag 9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod --To-tag as1d33b1c9 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Stopping retransmission on 'FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f' of Response 27931: Match Found [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Sending pending reinvite on 'FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f' [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Strict routing enforced for session FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: ** Our native-bridge filtered capablity: (alaw) [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Setting framing on incoming call: 0 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Initializing already initialized SIP dialog FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f (presumably reinvite) [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 0 [ 42]: INVITE sip:333@10.24.18.16:5060;ob SIP/2.0 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK7bd3ae06;rport [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 3 [ 43]: From: ;tag=as1d33b1c9 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 4 [ 69]: To: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 5 [ 36]: Contact: [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 6 [ 41]: Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.7.0 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uac [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 11 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.16:5060 [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31481][C-00000000] translate.c: Sample size different 320 vs 160 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.17.195:5060;rport=5060;received=10.24.17.195;branch=z9hG4bK7bd3ae06 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 2 [ 41]: Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 3 [ 43]: From: ;tag=as1d33b1c9 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 4 [ 69]: To: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 6 [ 35]: Session-Expires: 1800;refresher=uac [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 7 [ 44]: Contact: "333" [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 11 [ 19]: Content-Length: 242 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 12 [ 0]: [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 0 [ 3]: v=0 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 1 [ 42]: o=- 148793728 148793729 IN IP4 10.24.18.16 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 2 [ 8]: s=digphn [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 6 [ 25]: m=audio 4038 RTP/AVP 8 96 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 7 [ 30]: a=rtcp:4039 IN IP4 10.24.18.16 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: = Looking for Call ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f (Checking To) --From tag as1d33b1c9 --To-tag 9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Acked pending invite 102 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Stopping retransmission on 'FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f' of Request 102: Match Found [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: SIP response 200 to RE-invite on outgoing call FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP o=- 148793728 148793729 IN IP4 10.24.18.16... OK. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.16' into... [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.16' and port ''. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.16... OK. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Jan 21 18:27:18] DEBUG[31459][C-00000000] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb367fc60 [Jan 21 18:27:18] DEBUG[31459][C-00000000] rtp_engine.c: Don't have a default tx payload type 96 format for m type on 0xb367fc60 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4039 IN IP4 10.24.18.16... UNSUPPORTED OR FAILED. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Jan 21 18:27:18] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Set role to CONTROLLING (0x9ad357c) [Jan 21 18:27:18] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Set role failed; no ice instance (0x9ad357c) [Jan 21 18:27:18] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 8 (0x9acc1b4) from 0xb367fc60 to 0x9ad3728 [Jan 21 18:27:18] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 96 (0xa31e114) from 0xb367fc60 to 0x9ad3728 [Jan 21 18:27:18] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9ad357c' [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: We're settling with these formats: (alaw) [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (g722) [Jan 21 18:27:18] DEBUG[31459][C-00000000] channel.c: Channel SIP/333-00000000 setting read format path: alaw -> slin16 [Jan 21 18:27:18] DEBUG[31459][C-00000000] channel.c: Channel SIP/333-00000000 setting write format path: slin16 -> alaw [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Updating call counter for incoming call [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Session-Expires: 1800 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Refresher: UAC [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Session timer stopped: 9 - FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Session timer started: 10 - FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f 900000ms [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Strict routing enforced for session FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Trying to put 'ACK sip:333' onto UDP socket destined for 10.24.18.16:5060 [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Sending reinvite on SIP '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' - It's audio soon redirected to IP 10.24.18.16:4038 [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Strict routing enforced for session 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:18] DEBUG[31481][C-00000000] netsock2.c: Splitting '10.24.18.190:5060' into... [Jan 21 18:27:18] DEBUG[31481][C-00000000] netsock2.c: ...host '10.24.18.190' and port '5060'. [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: ** Our native-bridge filtered capablity: (alaw) [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: ** Our prefcodec: (g722) [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Initializing already initialized SIP dialog 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (presumably reinvite) [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 0 [ 40]: INVITE sip:123@10.24.18.190:5060 SIP/2.0 [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3d6f45c0 [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 3 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 4 [ 41]: To: ;tag=as1e3a359e [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 5 [ 36]: Contact: [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 6 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.7.0 [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 11 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Jan 21 18:27:18] DEBUG[31481][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.190:5060 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3d6f45c0;received=10.24.17.195 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 2 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 3 [ 41]: To: ;tag=as1e3a359e [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 4 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 6 [ 31]: Server: Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 10 [ 36]: Contact: [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: = Looking for Call ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (Checking To) --From tag as579facad --To-tag as1e3a359e [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: *** SIP TIMER: Cancelling retransmission #15 - INVITE (got response) [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' Request 104: Found [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: SIP response 100 to RE-invite on outgoing call 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3d6f45c0;received=10.24.17.195 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 2 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 3 [ 41]: To: ;tag=as1e3a359e [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 4 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 6 [ 31]: Server: Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 10 [ 36]: Contact: [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 12 [ 14]: Require: timer [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 13 [ 19]: Content-Length: 240 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Header 14 [ 0]: [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 0 [ 3]: v=0 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 1 [ 48]: o=root 1696456408 1696456410 IN IP4 10.24.18.190 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 2 [ 25]: s=Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.190 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 5 [ 26]: m=audio 13926 RTP/AVP 8 96 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 7 [ 32]: a=rtpmap:96 telephone-event/8000 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 8 [ 14]: a=fmtp:96 0-16 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 9 [ 14]: a=maxptime:150 [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jan 21 18:27:18] DEBUG[31459] chan_sip.c: = Looking for Call ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (Checking To) --From tag as579facad --To-tag as1e3a359e [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Acked pending invite 104 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Stopping retransmission on '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' of Request 104: Match Found [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: SIP response 200 to RE-invite on outgoing call 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP o=root 1696456408 1696456410 IN IP4 10.24.18.190... OK. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP s=Asterisk PBX 13.7.0-rc3... UNSUPPORTED OR FAILED. [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190' into... [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port ''. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.190... OK. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 21 18:27:18] DEBUG[31459][C-00000000] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb367fc60 [Jan 21 18:27:18] DEBUG[31459][C-00000000] rtp_engine.c: Don't have a default tx payload type 96 format for m type on 0xb367fc60 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-16... UNSUPPORTED OR FAILED. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 21 18:27:18] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Set role to CONTROLLING (0x9ace6e4) [Jan 21 18:27:18] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Set role failed; no ice instance (0x9ace6e4) [Jan 21 18:27:18] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 8 (0x9acc054) from 0xb367fc60 to 0x9ace890 [Jan 21 18:27:18] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 96 (0x9acbff4) from 0xb367fc60 to 0x9ace890 [Jan 21 18:27:18] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9ace6e4' [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: We're settling with these formats: (alaw) [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Updating call counter for outgoing call [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190:5060' into... [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port '5060'. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Session-Expires: 1800 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Refresher: UAS [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Session timer stopped: 11 - 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Session timer started: 16 - 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 1768000ms [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Strict routing enforced for session 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190:5060' into... [Jan 21 18:27:18] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port '5060'. [Jan 21 18:27:18] DEBUG[31459][C-00000000] chan_sip.c: Trying to put 'ACK sip:123' onto UDP socket destined for 10.24.18.190:5060 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 0 [ 37]: BYE sip:123@10.24.17.195:5060 SIP/2.0 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjjxYkSqj0XqK8c32J7ykFLzMWe7rg5mDo [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 3 [ 71]: From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 4 [ 41]: To: ;tag=as1d33b1c9 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 5 [ 41]: Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 6 [ 15]: CSeq: 27932 BYE [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 7 [ 36]: User-Agent: Digium D40 1_4_2_0_63880 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: = Looking for Call ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f (Checking From) --From tag 9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod --To-tag as1d33b1c9 [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Initializing initreq for method BYE - callid FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:32] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Jan 21 18:27:32] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Setting SIP_ALREADYGONE on dialog FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Session timer stopped: 10 - FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Received bye, issuing owner hangup [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.16:5060 [Jan 21 18:27:32] DEBUG[31480][C-00000000] bridge_channel.c: Setting 0x9ae37c4(SIP/333-00000000) state from:0 to:1 [Jan 21 18:27:32] DEBUG[31480][C-00000000] bridge_channel.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: pulling 0x9ae37c4(SIP/333-00000000) [Jan 21 18:27:32] DEBUG[31480][C-00000000] bridge_channel.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: 0x9ae37c4(SIP/333-00000000) is leaving native_rtp technology [Jan 21 18:27:32] DEBUG[31480][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x9ace6e4' [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Sending reinvite on SIP '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' - It's audio soon redirected to IP 10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Strict routing enforced for session 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31480][C-00000000] netsock2.c: Splitting '10.24.18.190:5060' into... [Jan 21 18:27:32] DEBUG[31480][C-00000000] netsock2.c: ...host '10.24.18.190' and port '5060'. [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: ** Our prefcodec: (g722) [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Initializing already initialized SIP dialog 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (presumably reinvite) [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 0 [ 40]: INVITE sip:123@10.24.18.190:5060 SIP/2.0 [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK690650b6 [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 3 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 4 [ 41]: To: ;tag=as1e3a359e [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 5 [ 36]: Contact: [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 6 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 105 INVITE [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 13.7.0 [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 11 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #5 [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.190:5060 [Jan 21 18:27:32] DEBUG[31480][C-00000000] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/333-00000000' and 'SIP/OutTrunk-00000001' - media will flow through Asterisk core [Jan 21 18:27:32] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: dissolving bridge with cause 16(Normal Clearing) [Jan 21 18:27:32] DEBUG[31480][C-00000000] bridge_channel.c: Setting 0x9ae2c2c(SIP/OutTrunk-00000001) state from:0 to:2 [Jan 21 18:27:32] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: queueing action type:13 sub:1001 [Jan 21 18:27:32] DEBUG[31480][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f is dissolved, not performing smart bridge operation. [Jan 21 18:27:32] DEBUG[31480][C-00000000] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jan 21 18:27:32] DEBUG[31480][C-00000000] pbx.c: Spawn extension (users,123,5) exited non-zero on 'SIP/333-00000000' [Jan 21 18:27:32] DEBUG[31480][C-00000000] channel.c: Soft-Hanging (0x10) up channel 'SIP/333-00000000' [Jan 21 18:27:32] DEBUG[31480][C-00000000] channel.c: Hanging up channel 'SIP/333-00000000' [Jan 21 18:27:32] DEBUG[31480][C-00000000] chan_sip.c: Hangup call SIP/333-00000000, SIP callid FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:32] DEBUG[31481][C-00000000] bridge_channel.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: pulling 0x9ae2c2c(SIP/OutTrunk-00000001) [Jan 21 18:27:32] DEBUG[31481][C-00000000] bridge_channel.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: 0x9ae2c2c(SIP/OutTrunk-00000001) is leaving native_rtp technology [Jan 21 18:27:32] DEBUG[31481][C-00000000] bridge_channel.c: Channel SIP/OutTrunk-00000001 will survive this bridge; clearing outgoing (dialed) flag [Jan 21 18:27:32] DEBUG[31481][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f is dissolved, not performing smart bridge operation. [Jan 21 18:27:32] DEBUG[31481][C-00000000] res_rtp_asterisk.c: Changing ssrc from 2124939824 to 532130985 due to a source change [Jan 21 18:27:32] DEBUG[31481][C-00000000] channel.c: Hanging up channel 'SIP/OutTrunk-00000001' [Jan 21 18:27:32] DEBUG[31481][C-00000000] chan_sip.c: Hangup call SIP/OutTrunk-00000001, SIP callid 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31481][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9ace6e4' [Jan 21 18:27:32] DEBUG[31481][C-00000000] chan_sip.c: Session timer stopped: 16 - 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31421][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: actually destroying basic bridge, nobody wants it anymore [Jan 21 18:27:32] DEBUG[31421][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: calling basic bridge destructor [Jan 21 18:27:32] DEBUG[31421][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: calling native_rtp technology stop [Jan 21 18:27:32] DEBUG[31421][C-00000000] bridge.c: Bridge 61ad6afe-27f5-44a8-aedf-846e5ba29d0f: calling native_rtp technology destructor [Jan 21 18:27:32] DEBUG[31422] devicestate.c: No provider found, checking channel drivers for SIP - 333 [Jan 21 18:27:32] DEBUG[31422] chan_sip.c: Checking device state for peer 333 [Jan 21 18:27:32] DEBUG[31422] devicestate.c: Changing state for SIP/333 - state 1 (Not in use) [Jan 21 18:27:32] DEBUG[31422] devicestate.c: No provider found, checking channel drivers for SIP - OutTrunk [Jan 21 18:27:32] DEBUG[31422] chan_sip.c: Checking device state for peer OutTrunk [Jan 21 18:27:32] DEBUG[31422] devicestate.c: Changing state for SIP/OutTrunk - state 1 (Not in use) [Jan 21 18:27:32] DEBUG[31428] cdr.c: Finalized CDR for SIP/333-00000000 - start 1453422437.941386 answer 1453422437.960950 end 1453422452.437912 dispo ANSWERED [Jan 21 18:27:32] DEBUG[31428] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: About to query table structure: SELECT sql FROM sqlite_master WHERE type='table' AND tbl_name='ast_cdr' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: id INTEGER [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: clid VARCHAR(80) NOT NULL DEFAULT '' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: src VARCHAR(80) NOT NULL DEFAULT '' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: dst VARCHAR(80) NOT NULL DEFAULT '' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: dcontext VARCHAR(80) NOT NULL DEFAULT '' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: channel VARCHAR(80) NOT NULL DEFAULT '' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: dstchannel VARCHAR(80) NOT NULL DEFAULT '' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: lastapp VARCHAR(80) NOT NULL DEFAULT '' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: lastdata VARCHAR(80) NOT NULL DEFAULT '' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: start DATETIME NOT NULL DEFAULT '0000-00-00 00:00:00' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: answer DATETIME NOT NULL DEFAULT '0000-00-00 00:00:00' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: end DATETIME NOT NULL DEFAULT '0000-00-00 00:00:00' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: duration INT(11) NOT NULL DEFAULT 0 [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: billsec INT(11) NOT NULL DEFAULT 0 [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: disposition VARCHAR(45) NOT NULL DEFAULT '' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: amaflags INT(11) NOT NULL DEFAULT 0 [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: accountcode VARCHAR(20) NOT NULL DEFAULT '' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: uniqueid VARCHAR(32) NOT NULL DEFAULT '' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: userfield VARCHAR(255) NOT NULL DEFAULT '' [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: Found field: PRIMARY KEY (id) [Jan 21 18:27:32] DEBUG[31428] res_config_sqlite.c: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('"333" <333>','333','123','users','SIP/333-00000000','SIP/OutTrunk-00000001','Dial','SIP/123@OutTrunk,60,','2016-01-21 18:27:17','2016-01-21 18:27:17','2016-01-21 18:27:32','14','14','ANSWERED','DOCUMENTATION','1453422437.0') [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK690650b6;received=10.24.17.195 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 2 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 3 [ 41]: To: ;tag=as1e3a359e [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 4 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 5 [ 16]: CSeq: 105 INVITE [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 6 [ 31]: Server: Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 10 [ 36]: Contact: [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: = Looking for Call ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (Checking To) --From tag as579facad --To-tag as1e3a359e [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: *** SIP TIMER: Cancelling retransmission #5 - INVITE (got response) [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' Request 105: Found [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: SIP response 100 to RE-invite on outgoing call 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK690650b6;received=10.24.17.195 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 2 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 3 [ 41]: To: ;tag=as1e3a359e [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 4 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 5 [ 16]: CSeq: 105 INVITE [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 6 [ 31]: Server: Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 10 [ 36]: Contact: [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 12 [ 14]: Require: timer [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 13 [ 19]: Content-Length: 240 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 14 [ 0]: [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Body 0 [ 3]: v=0 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Body 1 [ 48]: o=root 1696456408 1696456411 IN IP4 10.24.18.190 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Body 2 [ 25]: s=Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.190 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Body 5 [ 26]: m=audio 13926 RTP/AVP 8 96 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Body 7 [ 32]: a=rtpmap:96 telephone-event/8000 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Body 8 [ 14]: a=fmtp:96 0-16 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Body 9 [ 14]: a=maxptime:150 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: = Looking for Call ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (Checking To) --From tag as579facad --To-tag as1e3a359e [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Acked pending invite 105 [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Stopping retransmission on '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' of Request 105: Match Found [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: SIP response 200 to RE-invite on outgoing call 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP o=root 1696456408 1696456411 IN IP4 10.24.18.190... OK. [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP s=Asterisk PBX 13.7.0-rc3... UNSUPPORTED OR FAILED. [Jan 21 18:27:32] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190' into... [Jan 21 18:27:32] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port ''. [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.190... OK. [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 21 18:27:32] DEBUG[31459][C-00000000] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb367fce0 [Jan 21 18:27:32] DEBUG[31459][C-00000000] rtp_engine.c: Don't have a default tx payload type 96 format for m type on 0xb367fce0 [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-16... UNSUPPORTED OR FAILED. [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED. [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 21 18:27:32] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Set role to CONTROLLING (0x9ace6e4) [Jan 21 18:27:32] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Set role failed; no ice instance (0x9ace6e4) [Jan 21 18:27:32] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9ace6e4' [Jan 21 18:27:32] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 8 (0xa319ba4) from 0xb367fce0 to 0x9ace890 [Jan 21 18:27:32] DEBUG[31459][C-00000000] rtp_engine.c: Copying payload 96 (0xa32de24) from 0xb367fce0 to 0x9ace890 [Jan 21 18:27:32] DEBUG[31459][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x9ace6e4' [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: We're settling with these formats: (alaw) [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Updating call counter for outgoing call [Jan 21 18:27:32] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190:5060' into... [Jan 21 18:27:32] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port '5060'. [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Session-Expires: 1800 [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Refresher: UAS [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Session timer started: 8 - 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 1768000ms [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Strict routing enforced for session 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190:5060' into... [Jan 21 18:27:32] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port '5060'. [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Trying to put 'ACK sip:123' onto UDP socket destined for 10.24.18.190:5060 [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Strict routing enforced for session 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31459][C-00000000] netsock2.c: Splitting '10.24.18.190:5060' into... [Jan 21 18:27:32] DEBUG[31459][C-00000000] netsock2.c: ...host '10.24.18.190' and port '5060'. [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #2 [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Trying to put 'BYE sip:123' onto UDP socket destined for 10.24.18.190:5060 [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Session timer stopped: 8 - 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK1a071eca;received=10.24.17.195 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 2 [ 49]: From: "333" ;tag=as579facad [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 3 [ 41]: To: ;tag=as1e3a359e [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 4 [ 59]: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 5 [ 13]: CSeq: 106 BYE [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 6 [ 31]: Server: Asterisk PBX 13.7.0-rc3 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: = Looking for Call ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 (Checking To) --From tag as579facad --To-tag as1e3a359e [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #2 [Jan 21 18:27:32] DEBUG[31459][C-00000000] chan_sip.c: Stopping retransmission on '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' of Request 106: Match Found [Jan 21 18:27:32] DEBUG[31459] chan_sip.c: Destroying SIP dialog 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 [Jan 21 18:27:32] DEBUG[31459] rtp_engine.c: Destroyed RTP instance '0x9ace6e4' [Jan 21 18:27:32] DEBUG[31428] cdr.c: CDR for SIP/OutTrunk-00000001 is dialed and has no Party B; discarding [Jan 21 18:27:36] DEBUG[31478] cdr.c: CDR Engine termination request received; waiting on messages... [Jan 21 18:27:36] DEBUG[31478] taskprocessor.c: destroying taskprocessor 'ast_msg_queue' [Jan 21 18:27:36] DEBUG[31478] res_musiconhold.c: Destroying MOH class 'default' [Jan 21 18:27:36] DEBUG[31478] taskprocessor.c: destroying taskprocessor '0d1118e9-d5a7-4278-a997-851178a1e30e' [Jan 21 18:27:36] DEBUG[31478] asterisk.c: Asterisk ending (0).