[Mar 31 01:17:14] VERBOSE[31027] chan_sip.c: <--- SIP read from UDP:10.0.0.8:5060 ---> <-------------> [Mar 31 01:17:20] VERBOSE[31027] chan_sip.c: <--- SIP read from UDP:212.15.176.39:5060 ---> INVITE sip:1@valentin-vidic.from.hr SIP/2.0 v: SIP/2.0/UDP 212.15.176.39:5060;rport;branch=z9hG4bKPjHyDNjvIw1aK2IZ9Sd.rp0Fs8Gg0eisOx Max-Forwards: 70 f: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs t: m: "Galaksija" i: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6072 INVITE Route: k: replaces, 100rel, timer, norefersub x: 1800 Min-SE: 90 User-Agent: CSipSimple_GT-I9100-16/r2457 c: application/sdp l: 301 v=0 o=- 3636746239 3636746239 IN IP4 212.15.176.39 s=pjmedia c=IN IP4 212.15.176.39 t=0 0 m=audio 4000 RTP/AVP 96 3 0 8 101 c=IN IP4 212.15.176.39 a=rtcp:4001 IN IP4 212.15.176.39 a=sendrecv a=rtpmap:96 SILK/8000 a=fmtp:96 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 0 [ 43]: INVITE sip:1@valentin-vidic.from.hr SIP/2.0 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 1 [ 88]: v: SIP/2.0/UDP 212.15.176.39:5060;rport;branch=z9hG4bKPjHyDNjvIw1aK2IZ9Sd.rp0Fs8Gg0eisOx [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 3 [ 90]: f: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 4 [ 33]: t: [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 5 [ 52]: m: "Galaksija" [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 6 [ 35]: i: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 7 [ 17]: CSeq: 6072 INVITE [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 8 [ 52]: Route: [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 9 [ 38]: k: replaces, 100rel, timer, norefersub [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 10 [ 7]: x: 1800 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 12 [ 40]: User-Agent: CSipSimple_GT-I9100-16/r2457 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 13 [ 18]: c: application/sdp [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 14 [ 6]: l: 301 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Header 15 [ 0]: [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 0 [ 3]: v=0 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 1 [ 46]: o=- 3636746239 3636746239 IN IP4 212.15.176.39 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 2 [ 9]: s=pjmedia [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 3 [ 22]: c=IN IP4 212.15.176.39 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 5 [ 33]: m=audio 4000 RTP/AVP 96 3 0 8 101 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 6 [ 22]: c=IN IP4 212.15.176.39 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 7 [ 32]: a=rtcp:4001 IN IP4 212.15.176.39 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 8 [ 10]: a=sendrecv [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 9 [ 21]: a=rtpmap:96 SILK/8000 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 10 [ 24]: a=fmtp:96 useinbandfec=0 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-16 [Mar 31 01:17:20] VERBOSE[31027] chan_sip.c: --- (15 headers 13 lines) --- [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: = Looking for Call ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW (Checking From) --From tag cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs --To-tag [Mar 31 01:17:20] DEBUG[31027] acl.c: For destination '212.15.176.39', our source address is '192.168.0.7'. [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.0.7:5060 [Mar 31 01:17:20] DEBUG[31027] netsock2.c: Splitting '212.15.176.39:5060' into... [Mar 31 01:17:20] DEBUG[31027] netsock2.c: ...host '212.15.176.39' and port '5060'. [Mar 31 01:17:20] VERBOSE[31027] chan_sip.c: Sending to 212.15.176.39:5060 (NAT) [Mar 31 01:17:20] DEBUG[31027] chan_sip.c: Allocating new SIP dialog for H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW - INVITE (No RTP) [Mar 31 01:17:20] DEBUG[31027][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 31 01:17:20] DEBUG[31027][C-00000000] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, timer, norefersub" [Mar 31 01:17:20] DEBUG[31027][C-00000000] sip/reqresp_parser.c: Found SIP option: -replaces- [Mar 31 01:17:20] DEBUG[31027][C-00000000] sip/reqresp_parser.c: Matched SIP option: replaces [Mar 31 01:17:20] DEBUG[31027][C-00000000] sip/reqresp_parser.c: Found SIP option: -100rel- [Mar 31 01:17:20] DEBUG[31027][C-00000000] sip/reqresp_parser.c: Matched SIP option: 100rel [Mar 31 01:17:20] DEBUG[31027][C-00000000] sip/reqresp_parser.c: Found SIP option: -timer- [Mar 31 01:17:20] DEBUG[31027][C-00000000] sip/reqresp_parser.c: Matched SIP option: timer [Mar 31 01:17:20] DEBUG[31027][C-00000000] sip/reqresp_parser.c: Found SIP option: -norefersub- [Mar 31 01:17:20] DEBUG[31027][C-00000000] sip/reqresp_parser.c: Matched SIP option: norefersub [Mar 31 01:17:20] DEBUG[31027][C-00000000] netsock2.c: Splitting '212.15.176.39:5060' into... [Mar 31 01:17:20] DEBUG[31027][C-00000000] netsock2.c: ...host '212.15.176.39' and port '5060'. [Mar 31 01:17:20] VERBOSE[31027][C-00000000] chan_sip.c: Sending to 212.15.176.39:5060 (NAT) [Mar 31 01:17:20] DEBUG[31027][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:20] VERBOSE[31027][C-00000000] chan_sip.c: Using INVITE request as basis request - H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:20] DEBUG[31027][C-00000000] netsock2.c: Splitting 'valentin-vidic.from.hr' into... [Mar 31 01:17:20] DEBUG[31027][C-00000000] netsock2.c: ...host 'valentin-vidic.from.hr' and port ''. [Mar 31 01:17:20] VERBOSE[31027][C-00000000] chan_sip.c: Found peer 'galaksija' for 'galaksija' from 212.15.176.39:5060 [Mar 31 01:17:20] VERBOSE[31027][C-00000000] chan_sip.c: <--- Reliably Transmitting (NAT) to 212.15.176.39:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 212.15.176.39:5060;branch=z9hG4bKPjHyDNjvIw1aK2IZ9Sd.rp0Fs8Gg0eisOx;received=212.15.176.39;rport=5060 From: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs To: ;tag=as7256e212 Call-ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6072 INVITE Server: Asterisk PBX 11.13.1~dfsg-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="67f4ce91" Content-Length: 0 <------------> [Mar 31 01:17:20] DEBUG[31027][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #314 [Mar 31 01:17:20] DEBUG[31027][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 212.15.176.39:5060 [Mar 31 01:17:20] VERBOSE[31027][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog 'H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW' in 12992 ms (Method: INVITE) [Mar 31 01:17:21] VERBOSE[31027] chan_sip.c: <--- SIP read from UDP:212.15.176.39:5060 ---> INVITE sip:1@valentin-vidic.from.hr SIP/2.0 v: SIP/2.0/UDP 212.15.176.39:5060;rport;branch=z9hG4bKPjHyDNjvIw1aK2IZ9Sd.rp0Fs8Gg0eisOx Max-Forwards: 70 f: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs t: m: "Galaksija" i: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6072 INVITE Route: k: replaces, 100rel, timer, norefersub x: 1800 Min-SE: 90 User-Agent: CSipSimple_GT-I9100-16/r2457 c: application/sdp l: 301 v=0 o=- 3636746239 3636746239 IN IP4 212.15.176.39 s=pjmedia c=IN IP4 212.15.176.39 t=0 0 m=audio 4000 RTP/AVP 96 3 0 8 101 c=IN IP4 212.15.176.39 a=rtcp:4001 IN IP4 212.15.176.39 a=sendrecv a=rtpmap:96 SILK/8000 a=fmtp:96 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 0 [ 43]: INVITE sip:1@valentin-vidic.from.hr SIP/2.0 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 1 [ 88]: v: SIP/2.0/UDP 212.15.176.39:5060;rport;branch=z9hG4bKPjHyDNjvIw1aK2IZ9Sd.rp0Fs8Gg0eisOx [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 3 [ 90]: f: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 4 [ 33]: t: [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 5 [ 52]: m: "Galaksija" [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 6 [ 35]: i: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 7 [ 17]: CSeq: 6072 INVITE [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 8 [ 52]: Route: [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 9 [ 38]: k: replaces, 100rel, timer, norefersub [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 10 [ 7]: x: 1800 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 12 [ 40]: User-Agent: CSipSimple_GT-I9100-16/r2457 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 13 [ 18]: c: application/sdp [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 14 [ 6]: l: 301 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 15 [ 0]: [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 0 [ 3]: v=0 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 1 [ 46]: o=- 3636746239 3636746239 IN IP4 212.15.176.39 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 2 [ 9]: s=pjmedia [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 3 [ 22]: c=IN IP4 212.15.176.39 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 5 [ 33]: m=audio 4000 RTP/AVP 96 3 0 8 101 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 6 [ 22]: c=IN IP4 212.15.176.39 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 7 [ 32]: a=rtcp:4001 IN IP4 212.15.176.39 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 8 [ 10]: a=sendrecv [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 9 [ 21]: a=rtpmap:96 SILK/8000 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 10 [ 24]: a=fmtp:96 useinbandfec=0 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-16 [Mar 31 01:17:21] VERBOSE[31027] chan_sip.c: --- (15 headers 13 lines) --- [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: = Looking for Call ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW (Checking From) --From tag cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs --To-tag [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Ignoring SIP message because of retransmit (INVITE Seqno 6072, ours 6072) [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Ignoring this INVITE request [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Got a SIP re-transmit of INVITE for call H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Incoming INVITE with 'timer' option supported [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: INVITE also has "Session-Expires" header. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Session-Expires: 1800 [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: INVITE also has "Min-SE" header. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Received Min-SE: 90 [Mar 31 01:17:21] VERBOSE[31027] chan_sip.c: <--- SIP read from UDP:212.15.176.39:5060 ---> ACK sip:1@valentin-vidic.from.hr SIP/2.0 v: SIP/2.0/UDP 212.15.176.39:5060;rport;branch=z9hG4bKPjHyDNjvIw1aK2IZ9Sd.rp0Fs8Gg0eisOx Max-Forwards: 70 f: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs t: ;tag=as7256e212 i: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6072 ACK Route: l: 0 <-------------> [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 0 [ 40]: ACK sip:1@valentin-vidic.from.hr SIP/2.0 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 1 [ 88]: v: SIP/2.0/UDP 212.15.176.39:5060;rport;branch=z9hG4bKPjHyDNjvIw1aK2IZ9Sd.rp0Fs8Gg0eisOx [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 3 [ 90]: f: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 4 [ 48]: t: ;tag=as7256e212 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 5 [ 35]: i: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 6 [ 14]: CSeq: 6072 ACK [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 7 [ 52]: Route: [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 8 [ 4]: l: 0 [Mar 31 01:17:21] VERBOSE[31027] chan_sip.c: --- (9 headers 0 lines) --- [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: = Looking for Call ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW (Checking From) --From tag cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs --To-tag as7256e212 [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #314 [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Stopping retransmission on 'H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW' of Response 6072: Match Found [Mar 31 01:17:21] VERBOSE[31027] chan_sip.c: <--- SIP read from UDP:212.15.176.39:5060 ---> INVITE sip:1@valentin-vidic.from.hr SIP/2.0 v: SIP/2.0/UDP 212.15.176.39:5060;rport;branch=z9hG4bKPjjrMxi-YFQoqdTN0AsXWbm6z-DBtWrf-d Max-Forwards: 70 f: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs t: m: "Galaksija" i: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6073 INVITE Route: k: replaces, 100rel, timer, norefersub x: 1800 Min-SE: 90 User-Agent: CSipSimple_GT-I9100-16/r2457 Authorization: Digest username="galaksija", realm="asterisk", nonce="67f4ce91", uri="sip:1@valentin-vidic.from.hr", response="84b9207601538b3b0ddbf02c6ea41916", algorithm=MD5 c: application/sdp l: 301 v=0 o=- 3636746239 3636746239 IN IP4 212.15.176.39 s=pjmedia c=IN IP4 212.15.176.39 t=0 0 m=audio 4000 RTP/AVP 96 3 0 8 101 c=IN IP4 212.15.176.39 a=rtcp:4001 IN IP4 212.15.176.39 a=sendrecv a=rtpmap:96 SILK/8000 a=fmtp:96 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 0 [ 43]: INVITE sip:1@valentin-vidic.from.hr SIP/2.0 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 1 [ 88]: v: SIP/2.0/UDP 212.15.176.39:5060;rport;branch=z9hG4bKPjjrMxi-YFQoqdTN0AsXWbm6z-DBtWrf-d [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 3 [ 90]: f: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 4 [ 33]: t: [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 5 [ 52]: m: "Galaksija" [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 6 [ 35]: i: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 7 [ 17]: CSeq: 6073 INVITE [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 8 [ 52]: Route: [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 9 [ 38]: k: replaces, 100rel, timer, norefersub [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 10 [ 7]: x: 1800 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 12 [ 40]: User-Agent: CSipSimple_GT-I9100-16/r2457 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 13 [174]: Authorization: Digest username="galaksija", realm="asterisk", nonce="67f4ce91", uri="sip:1@valentin-vidic.from.hr", response="84b9207601538b3b0ddbf02c6ea41916", algorithm=MD5 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 14 [ 18]: c: application/sdp [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 15 [ 6]: l: 301 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Header 16 [ 0]: [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 0 [ 3]: v=0 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 1 [ 46]: o=- 3636746239 3636746239 IN IP4 212.15.176.39 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 2 [ 9]: s=pjmedia [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 3 [ 22]: c=IN IP4 212.15.176.39 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 5 [ 33]: m=audio 4000 RTP/AVP 96 3 0 8 101 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 6 [ 22]: c=IN IP4 212.15.176.39 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 7 [ 32]: a=rtcp:4001 IN IP4 212.15.176.39 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 8 [ 10]: a=sendrecv [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 9 [ 21]: a=rtpmap:96 SILK/8000 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 10 [ 24]: a=fmtp:96 useinbandfec=0 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-16 [Mar 31 01:17:21] VERBOSE[31027] chan_sip.c: --- (16 headers 13 lines) --- [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: = Looking for Call ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW (Checking From) --From tag cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs --To-tag [Mar 31 01:17:21] DEBUG[31027] netsock2.c: Splitting 'valentin-vidic.from.hr' into... [Mar 31 01:17:21] DEBUG[31027] netsock2.c: ...host 'valentin-vidic.from.hr' and port ''. [Mar 31 01:17:21] DEBUG[31027] netsock2.c: Splitting 'valentin-vidic.from.hr' into... [Mar 31 01:17:21] DEBUG[31027] netsock2.c: ...host 'valentin-vidic.from.hr' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting '212.15.176.39:5060' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host '212.15.176.39' and port '5060'. [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Sending to 212.15.176.39:5060 (NAT) [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Using INVITE request as basis request - H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting 'valentin-vidic.from.hr' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host 'valentin-vidic.from.hr' and port ''. [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Found peer 'galaksija' for 'galaksija' from 212.15.176.39:5060 [Mar 31 01:17:21] DEBUG[31027][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb8f41ebc' [Mar 31 01:17:21] DEBUG[31027][C-00000000] res_rtp_asterisk.c: Allocated port 10274 for RTP instance '0xb8f41ebc' [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting 'fe80::250:bfff:fed0:db53' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host 'fe80::250:bfff:fed0:db53' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting 'fe80::225:22ff:fe82:2154' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host 'fe80::225:22ff:fe82:2154' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting 'fe80::20f:54ff:fe12:362f' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host 'fe80::20f:54ff:fe12:362f' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting '2001:470:1f0b:3b7::1' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host '2001:470:1f0b:3b7::1' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting '2001:15c0:6614::1' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host '2001:15c0:6614::1' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting 'fe80::225:22ff:fe82:2154' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host 'fe80::225:22ff:fe82:2154' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting '2001:15c0:65ff:250::2' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host '2001:15c0:65ff:250::2' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting 'fe80::14c0:65ff:250:2' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host 'fe80::14c0:65ff:250:2' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting '2001:470:1f0a:3b7::2' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host '2001:470:1f0a:3b7::2' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting 'fe80::6dc8:1715' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host 'fe80::6dc8:1715' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting 'fe80::e8ee:5fff:fee8:e246' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host 'fe80::e8ee:5fff:fee8:e246' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting '192.168.0.7' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host '192.168.0.7' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting '10.0.0.7' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host '10.0.0.7' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting '109.200.23.21' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host '109.200.23.21' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] rtp_engine.c: RTP instance '0xb8f41ebc' is setup and ready to go [Mar 31 01:17:21] DEBUG[31027][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb8f41ebc' [Mar 31 01:17:21] VERBOSE[31027][C-00000000] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Setting NAT on RTP to On [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Processing session-level SDP o=- 3636746239 3636746239 IN IP4 212.15.176.39... OK. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Processing session-level SDP s=pjmedia... UNSUPPORTED OR FAILED. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting '212.15.176.39' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host '212.15.176.39' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 212.15.176.39... OK. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 31 01:17:21] DEBUG[31027][C-00000000] rtp_engine.c: Setting payload 96 based on m type on 0xb39ef310 [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Found RTP audio format 3 [Mar 31 01:17:21] DEBUG[31027][C-00000000] rtp_engine.c: Setting payload 3 based on m type on 0xb39ef310 [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 31 01:17:21] DEBUG[31027][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0xb39ef310 [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Found RTP audio format 8 [Mar 31 01:17:21] DEBUG[31027][C-00000000] rtp_engine.c: Setting payload 8 based on m type on 0xb39ef310 [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Found RTP audio format 101 [Mar 31 01:17:21] DEBUG[31027][C-00000000] rtp_engine.c: Setting payload 101 based on m type on 0xb39ef310 [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting '212.15.176.39' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host '212.15.176.39' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 212.15.176.39... OK. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4001 IN IP4 212.15.176.39... UNSUPPORTED OR FAILED. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Found audio description format SILK for ID 96 [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 SILK/8000... OK. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 useinbandfec=0... OK. [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101 [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw|alaw|silk8)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw) [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 31 01:17:21] DEBUG[31027][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb8f41ebc' [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Peer audio RTP is at port 212.15.176.39:4000 [Mar 31 01:17:21] DEBUG[31027][C-00000000] rtp_engine.c: Copying payload 0 from 0xb39ef310 to 0xb8f42068 [Mar 31 01:17:21] DEBUG[31027][C-00000000] rtp_engine.c: Copying payload 3 from 0xb39ef310 to 0xb8f42068 [Mar 31 01:17:21] DEBUG[31027][C-00000000] rtp_engine.c: Copying payload 8 from 0xb39ef310 to 0xb8f42068 [Mar 31 01:17:21] DEBUG[31027][C-00000000] rtp_engine.c: Copying payload 96 from 0xb39ef310 to 0xb8f42068 [Mar 31 01:17:21] DEBUG[31027][C-00000000] rtp_engine.c: Copying payload 101 from 0xb39ef310 to 0xb8f42068 [Mar 31 01:17:21] DEBUG[31027][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb8f41ebc' [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: We're settling with these formats: (gsm|ulaw|alaw) [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Checking SIP call limits for device galaksija [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Updating call counter for incoming call [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Call from peer 'galaksija' is 1 out of 5 [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting 'valentin-vidic.from.hr' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host 'valentin-vidic.from.hr' and port ''. [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: Splitting 'valentin-vidic.from.hr' into... [Mar 31 01:17:21] DEBUG[31027][C-00000000] netsock2.c: ...host 'valentin-vidic.from.hr' and port ''. [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: Looking for 1 in from-home (domain valentin-vidic.from.hr) [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Incoming INVITE with 'timer' option supported [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: INVITE also has "Session-Expires" header. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Session-Expires: 1800 [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: INVITE also has "Min-SE" header. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Received Min-SE: 90 [Mar 31 01:17:21] DEBUG[31013] devicestate.c: No provider found, checking channel drivers for SIP - galaksija [Mar 31 01:17:21] DEBUG[31013] chan_sip.c: Checking device state for peer galaksija [Mar 31 01:17:21] DEBUG[31013] devicestate.c: Changing state for SIP/galaksija - state 2 (In use) [Mar 31 01:17:21] DEBUG[31013] devicestate.c: device 'SIP/galaksija' state '2' [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: *** Our native formats are (gsm) [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: *** Joint capabilities are (gsm|ulaw|alaw) [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: *** Our capabilities are (gsm|ulaw|alaw) [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: *** AST_CODEC_CHOOSE formats are gsm [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: This channel will not be able to handle video. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: build_route: Contact hop: "Galaksija" [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: list_route: hop: [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: SIP/galaksija-00000000: New call is still down.... Trying... [Mar 31 01:17:21] VERBOSE[31027][C-00000000] chan_sip.c: <--- Transmitting (NAT) to 212.15.176.39:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.15.176.39:5060;branch=z9hG4bKPjjrMxi-YFQoqdTN0AsXWbm6z-DBtWrf-d;received=212.15.176.39;rport=5060 From: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs To: Call-ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6073 INVITE Server: Asterisk PBX 11.13.1~dfsg-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Mar 31 01:17:21] DEBUG[31055] app_queue.c: Device 'SIP/galaksija' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Mar 31 01:17:21] DEBUG[31027][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 212.15.176.39:5060 [Mar 31 01:17:21] DEBUG[31013] devicestate.c: No provider found, checking channel drivers for SIP - galaksija [Mar 31 01:17:21] DEBUG[31013] chan_sip.c: Checking device state for peer galaksija [Mar 31 01:17:21] DEBUG[31013] devicestate.c: Changing state for SIP/galaksija - state 2 (In use) [Mar 31 01:17:21] DEBUG[31013] devicestate.c: device 'SIP/galaksija' state '2' [Mar 31 01:17:21] DEBUG[422][C-00000000] pbx.c: Launching 'VoiceMailMain' [Mar 31 01:17:21] VERBOSE[422][C-00000000] pbx.c: -- Executing [1@from-home:1] VoiceMailMain("SIP/galaksija-00000000", "1") in new stack [Mar 31 01:17:21] DEBUG[422][C-00000000] app_voicemail.c: Before ast_answer [Mar 31 01:17:21] DEBUG[422][C-00000000] chan_sip.c: SIP answering channel: SIP/galaksija-00000000 [Mar 31 01:17:21] DEBUG[422][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Mar 31 01:17:21] DEBUG[422][C-00000000] chan_sip.c: Setting framing from config on incoming call [Mar 31 01:17:21] DEBUG[31013] devicestate.c: No provider found, checking channel drivers for SIP - galaksija [Mar 31 01:17:21] DEBUG[31013] chan_sip.c: Checking device state for peer galaksija [Mar 31 01:17:21] DEBUG[422][C-00000000] chan_sip.c: ** Our capability: (gsm|ulaw|alaw) Video flag: True Text flag: True [Mar 31 01:17:21] DEBUG[31013] devicestate.c: Changing state for SIP/galaksija - state 2 (In use) [Mar 31 01:17:21] DEBUG[31013] devicestate.c: device 'SIP/galaksija' state '2' [Mar 31 01:17:21] DEBUG[422][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [Mar 31 01:17:21] VERBOSE[422][C-00000000] chan_sip.c: Audio is at 10274 [Mar 31 01:17:21] VERBOSE[422][C-00000000] chan_sip.c: Adding codec 100002 (gsm) to SDP [Mar 31 01:17:21] VERBOSE[422][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 31 01:17:21] VERBOSE[422][C-00000000] chan_sip.c: Adding codec 100004 (alaw) to SDP [Mar 31 01:17:21] VERBOSE[422][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 31 01:17:21] DEBUG[422][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Mar 31 01:17:21] DEBUG[422][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw|alaw) [Mar 31 01:17:21] VERBOSE[422][C-00000000] chan_sip.c: <--- Reliably Transmitting (NAT) to 212.15.176.39:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.15.176.39:5060;branch=z9hG4bKPjjrMxi-YFQoqdTN0AsXWbm6z-DBtWrf-d;received=212.15.176.39;rport=5060 From: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs To: ;tag=as4f35302a Call-ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6073 INVITE Server: Asterisk PBX 11.13.1~dfsg-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 288 v=0 o=root 1599805638 1599805638 IN IP4 192.168.0.7 s=Asterisk PBX 11.13.1~dfsg-2 c=IN IP4 192.168.0.7 t=0 0 m=audio 10274 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Mar 31 01:17:21] DEBUG[422][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #317 [Mar 31 01:17:21] DEBUG[422][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 212.15.176.39:5060 [Mar 31 01:17:21] DEBUG[422][C-00000000] chan_sip.c: Session timer started: 318 - H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW 900000ms [Mar 31 01:17:21] DEBUG[31055] app_queue.c: Device 'SIP/galaksija' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Mar 31 01:17:21] DEBUG[31055] app_queue.c: Device 'SIP/galaksija' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: SIP TIMER: Rescheduling retransmission #317 (1) SIP/2.0 - 1 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 406 ms (t1 203 ms (Retrans id #317)) [Mar 31 01:17:21] VERBOSE[31027] chan_sip.c: Retransmitting #1 (NAT) to 212.15.176.39:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.15.176.39:5060;branch=z9hG4bKPjjrMxi-YFQoqdTN0AsXWbm6z-DBtWrf-d;received=212.15.176.39;rport=5060 From: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs To: ;tag=as4f35302a Call-ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6073 INVITE Server: Asterisk PBX 11.13.1~dfsg-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 288 v=0 o=root 1599805638 1599805638 IN IP4 192.168.0.7 s=Asterisk PBX 11.13.1~dfsg-2 c=IN IP4 192.168.0.7 t=0 0 m=audio 10274 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 212.15.176.39:5060 [Mar 31 01:17:21] DEBUG[422][C-00000000] channel.c: Didn't receive a media frame from SIP/galaksija-00000000 within 500 ms of answering. Continuing anyway [Mar 31 01:17:21] DEBUG[422][C-00000000] app_voicemail.c: Before find user for mailbox 1 [Mar 31 01:17:21] DEBUG[422][C-00000000] res_rtp_asterisk.c: Ooh, format changed from unknown to gsm [Mar 31 01:17:21] DEBUG[422][C-00000000] res_rtp_asterisk.c: Created smoother: format: gsm ms: 20 len: 33 [Mar 31 01:17:21] DEBUG[422][C-00000000] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xb8f41ebc' [Mar 31 01:17:21] DEBUG[422][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Mar 31 01:17:21] VERBOSE[422][C-00000000] file.c: -- Playing 'vm-password.gsm' (language 'en') [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: SIP TIMER: Rescheduling retransmission #317 (2) SIP/2.0 - 1 [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 812 ms (t1 203 ms (Retrans id #317)) [Mar 31 01:17:21] VERBOSE[31027] chan_sip.c: Retransmitting #2 (NAT) to 212.15.176.39:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.15.176.39:5060;branch=z9hG4bKPjjrMxi-YFQoqdTN0AsXWbm6z-DBtWrf-d;received=212.15.176.39;rport=5060 From: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs To: ;tag=as4f35302a Call-ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6073 INVITE Server: Asterisk PBX 11.13.1~dfsg-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 288 v=0 o=root 1599805638 1599805638 IN IP4 192.168.0.7 s=Asterisk PBX 11.13.1~dfsg-2 c=IN IP4 192.168.0.7 t=0 0 m=audio 10274 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Mar 31 01:17:21] DEBUG[31027] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 212.15.176.39:5060 [Mar 31 01:17:21] DEBUG[422][C-00000000] res_rtp_asterisk.c: 0xb8f8adb8 -- Probation learning mode pass with source address 212.15.176.39:4000 [Mar 31 01:17:22] DEBUG[422][C-00000000] res_rtp_asterisk.c: Got RTCP report of 84 bytes [Mar 31 01:17:22] VERBOSE[31027] chan_sip.c: <--- SIP read from UDP:10.0.0.152:32912 ---> <-------------> [Mar 31 01:17:22] DEBUG[422][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Mar 31 01:17:22] DEBUG[422][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Mar 31 01:17:22] DEBUG[422][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Mar 31 01:17:22] DEBUG[31027] chan_sip.c: SIP TIMER: Rescheduling retransmission #317 (3) SIP/2.0 - 1 [Mar 31 01:17:22] DEBUG[31027] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 1624 ms (t1 203 ms (Retrans id #317)) [Mar 31 01:17:22] VERBOSE[31027] chan_sip.c: Retransmitting #3 (NAT) to 212.15.176.39:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.15.176.39:5060;branch=z9hG4bKPjjrMxi-YFQoqdTN0AsXWbm6z-DBtWrf-d;received=212.15.176.39;rport=5060 From: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs To: ;tag=as4f35302a Call-ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6073 INVITE Server: Asterisk PBX 11.13.1~dfsg-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 288 v=0 o=root 1599805638 1599805638 IN IP4 192.168.0.7 s=Asterisk PBX 11.13.1~dfsg-2 c=IN IP4 192.168.0.7 t=0 0 m=audio 10274 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Mar 31 01:17:22] DEBUG[31027] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 212.15.176.39:5060 [Mar 31 01:17:24] DEBUG[31027] chan_sip.c: SIP TIMER: Rescheduling retransmission #317 (4) SIP/2.0 - 1 [Mar 31 01:17:24] DEBUG[31027] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 3248 ms (t1 203 ms (Retrans id #317)) [Mar 31 01:17:24] VERBOSE[31027] chan_sip.c: Retransmitting #4 (NAT) to 212.15.176.39:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.15.176.39:5060;branch=z9hG4bKPjjrMxi-YFQoqdTN0AsXWbm6z-DBtWrf-d;received=212.15.176.39;rport=5060 From: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs To: ;tag=as4f35302a Call-ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6073 INVITE Server: Asterisk PBX 11.13.1~dfsg-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 288 v=0 o=root 1599805638 1599805638 IN IP4 192.168.0.7 s=Asterisk PBX 11.13.1~dfsg-2 c=IN IP4 192.168.0.7 t=0 0 m=audio 10274 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Mar 31 01:17:24] DEBUG[31027] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 212.15.176.39:5060 [Mar 31 01:17:25] DEBUG[422][C-00000000] res_rtp_asterisk.c: Got RTCP report of 84 bytes [Mar 31 01:17:27] DEBUG[31027] chan_sip.c: SIP TIMER: Rescheduling retransmission #317 (5) SIP/2.0 - 1 [Mar 31 01:17:27] DEBUG[31027] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 203 ms (Retrans id #317)) [Mar 31 01:17:27] VERBOSE[31027] chan_sip.c: Retransmitting #5 (NAT) to 212.15.176.39:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.15.176.39:5060;branch=z9hG4bKPjjrMxi-YFQoqdTN0AsXWbm6z-DBtWrf-d;received=212.15.176.39;rport=5060 From: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs To: ;tag=as4f35302a Call-ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6073 INVITE Server: Asterisk PBX 11.13.1~dfsg-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 288 v=0 o=root 1599805638 1599805638 IN IP4 192.168.0.7 s=Asterisk PBX 11.13.1~dfsg-2 c=IN IP4 192.168.0.7 t=0 0 m=audio 10274 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Mar 31 01:17:27] DEBUG[31027] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 212.15.176.39:5060 [Mar 31 01:17:30] DEBUG[422][C-00000000] res_rtp_asterisk.c: Got RTCP report of 84 bytes [Mar 31 01:17:31] DEBUG[31027] chan_sip.c: SIP TIMER: Rescheduling retransmission #317 (6) SIP/2.0 - 1 [Mar 31 01:17:31] DEBUG[31027] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 203 ms (Retrans id #317)) [Mar 31 01:17:31] VERBOSE[31027] chan_sip.c: Retransmitting #6 (NAT) to 212.15.176.39:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.15.176.39:5060;branch=z9hG4bKPjjrMxi-YFQoqdTN0AsXWbm6z-DBtWrf-d;received=212.15.176.39;rport=5060 From: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs To: ;tag=as4f35302a Call-ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 6073 INVITE Server: Asterisk PBX 11.13.1~dfsg-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 288 v=0 o=root 1599805638 1599805638 IN IP4 192.168.0.7 s=Asterisk PBX 11.13.1~dfsg-2 c=IN IP4 192.168.0.7 t=0 0 m=audio 10274 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Mar 31 01:17:31] DEBUG[31027] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 212.15.176.39:5060 [Mar 31 01:17:32] VERBOSE[422][C-00000000] app_voicemail.c: -- Incorrect password '' for user '1' (context = default) [Mar 31 01:17:32] DEBUG[422][C-00000000] res_rtp_asterisk.c: Difference is 80024, ms is 10023 [Mar 31 01:17:32] DEBUG[422][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Mar 31 01:17:32] VERBOSE[422][C-00000000] file.c: -- Playing 'vm-incorrect.gsm' (language 'en') [Mar 31 01:17:32] DEBUG[31039] chan_iax2.c: Allocate call number [Mar 31 01:17:32] DEBUG[31039] chan_iax2.c: ip callno count incremented to 2 for 161.53.30.231 [Mar 31 01:17:32] DEBUG[31039] chan_iax2.c: Registration created on call 6432 [Mar 31 01:17:32] DEBUG[31033] chan_iax2.c: schedule decrement of callno used for 161.53.30.231 in 60 seconds [Mar 31 01:17:34] DEBUG[422][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Mar 31 01:17:34] DEBUG[422][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Mar 31 01:17:34] DEBUG[422][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Mar 31 01:17:34] DEBUG[422][C-00000000] app_voicemail.c: Before find user for mailbox 1 [Mar 31 01:17:34] DEBUG[422][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Mar 31 01:17:34] VERBOSE[422][C-00000000] file.c: -- Playing 'vm-password.gsm' (language 'en') [Mar 31 01:17:34] WARNING[31027] chan_sip.c: Retransmission timeout reached on transmission H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW for seqno 6073 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 12992ms with no response [Mar 31 01:17:34] WARNING[31027] chan_sip.c: Hanging up call H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Mar 31 01:17:34] DEBUG[422][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Mar 31 01:17:34] DEBUG[422][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Mar 31 01:17:34] WARNING[422][C-00000000] app_voicemail.c: Unable to read password [Mar 31 01:17:34] DEBUG[422][C-00000000] app_voicemail.c: After vm_authenticate [Mar 31 01:17:34] DEBUG[422][C-00000000] pbx.c: Extension 1, priority 1 returned normally even though call was hung up [Mar 31 01:17:34] DEBUG[422][C-00000000] channel.c: Soft-Hanging up channel 'SIP/galaksija-00000000' [Mar 31 01:17:34] DEBUG[422][C-00000000] channel.c: Hanging up channel 'SIP/galaksija-00000000' [Mar 31 01:17:34] DEBUG[422][C-00000000] chan_sip.c: Hangup call SIP/galaksija-00000000, SIP callid H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:34] DEBUG[422][C-00000000] chan_sip.c: update_call_counter(galaksija) - decrement call limit counter on hangup [Mar 31 01:17:34] DEBUG[422][C-00000000] chan_sip.c: Updating call counter for incoming call [Mar 31 01:17:34] DEBUG[422][C-00000000] chan_sip.c: Call from peer 'galaksija' removed from call limit 5 [Mar 31 01:17:34] DEBUG[422][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb8f41ebc' [Mar 31 01:17:34] VERBOSE[422][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog 'H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW' in 12992 ms (Method: INVITE) [Mar 31 01:17:34] DEBUG[422][C-00000000] chan_sip.c: Session timer stopped: 318 - H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:34] DEBUG[31013] devicestate.c: No provider found, checking channel drivers for SIP - galaksija [Mar 31 01:17:34] DEBUG[31013] chan_sip.c: Checking device state for peer galaksija [Mar 31 01:17:34] DEBUG[422][C-00000000] chan_sip.c: Strict routing enforced for session H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:34] DEBUG[31013] devicestate.c: Changing state for SIP/galaksija - state 1 (Not in use) [Mar 31 01:17:34] DEBUG[31013] devicestate.c: device 'SIP/galaksija' state '1' [Mar 31 01:17:34] VERBOSE[422][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 31 01:17:34] DEBUG[422][C-00000000] netsock2.c: Splitting '212.15.176.39:5060' into... [Mar 31 01:17:34] DEBUG[422][C-00000000] netsock2.c: ...host '212.15.176.39' and port '5060'. [Mar 31 01:17:34] VERBOSE[422][C-00000000] chan_sip.c: set_destination: set destination to 212.15.176.39:5060 [Mar 31 01:17:34] VERBOSE[422][C-00000000] chan_sip.c: Reliably Transmitting (NAT) to 212.15.176.39:5060: BYE sip:galaksija@212.15.176.39:5060;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.0.7:5060;branch=z9hG4bK27623294;rport Max-Forwards: 70 From: ;tag=as4f35302a To: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs Call-ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW CSeq: 102 BYE User-Agent: Asterisk PBX 11.13.1~dfsg-2 Proxy-Authorization: Digest username="galaksija", realm="asterisk", algorithm=MD5, uri="sip:valentin-vidic.from.hr", nonce="67f4ce91", response="641561dfafcec8588bfaa66d200aeb6b" X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0 --- [Mar 31 01:17:34] DEBUG[31055] app_queue.c: Device 'SIP/galaksija' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 31 01:17:34] DEBUG[422][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #321 [Mar 31 01:17:34] DEBUG[422][C-00000000] chan_sip.c: Trying to put 'BYE sip:gal' onto UDP socket destined for 212.15.176.39:5060 [Mar 31 01:17:34] DEBUG[422][C-00000000] cdr_mysql.c: Inserting a CDR record. [Mar 31 01:17:34] DEBUG[422][C-00000000] cdr_mysql.c: SQL command as follows: INSERT INTO cdr (`calldate`,`clid`,`src`,`dst`,`dcontext`,`channel`,`lastapp`,`lastdata`,`duration`,`billsec`,`disposition`,`amaflags`,`uniqueid`) VALUES ('2015-03-31 01:17:21','\"Galaksija\" ','galaksija','1','from-home','SIP/galaksija-00000000','VoiceMailMain','1','13','13','ANSWERED','3','1427757441.0') [Mar 31 01:17:34] VERBOSE[31027] chan_sip.c: <--- SIP read from UDP:212.15.176.39:5060 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.0.7:5060;rport=5060;received=193.198.125.23;branch=z9hG4bK27623294 i: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW f: ;tag=as4f35302a t: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs CSeq: 102 BYE l: 0 <-------------> [Mar 31 01:17:34] DEBUG[31027] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 31 01:17:34] DEBUG[31027] chan_sip.c: Header 1 [ 89]: v: SIP/2.0/UDP 192.168.0.7:5060;rport=5060;received=193.198.125.23;branch=z9hG4bK27623294 [Mar 31 01:17:34] DEBUG[31027] chan_sip.c: Header 2 [ 35]: i: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:34] DEBUG[31027] chan_sip.c: Header 3 [ 48]: f: ;tag=as4f35302a [Mar 31 01:17:34] DEBUG[31027] chan_sip.c: Header 4 [ 90]: t: "Galaksija" ;tag=cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs [Mar 31 01:17:34] DEBUG[31027] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE [Mar 31 01:17:34] DEBUG[31027] chan_sip.c: Header 6 [ 4]: l: 0 [Mar 31 01:17:34] VERBOSE[31027] chan_sip.c: --- (7 headers 0 lines) --- [Mar 31 01:17:34] DEBUG[31027] chan_sip.c: = Looking for Call ID: H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW (Checking To) --From tag as4f35302a --To-tag cXOF-eHs3Q4WKcyuh.Crg0SqIdVtEzUs [Mar 31 01:17:34] DEBUG[31027][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #321 [Mar 31 01:17:34] DEBUG[31027][C-00000000] chan_sip.c: Stopping retransmission on 'H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW' of Request 102: Match Found [Mar 31 01:17:34] VERBOSE[31027][C-00000000] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Mar 31 01:17:34] DEBUG[31027] chan_sip.c: Destroying SIP dialog H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW [Mar 31 01:17:34] VERBOSE[31027] chan_sip.c: Really destroying SIP dialog 'H91V0Q7qmbYyUeQKNcM6CiEkEnR9oFbW' Method: INVITE [Mar 31 01:17:34] DEBUG[31027] rtp_engine.c: Destroyed RTP instance '0xb8f41ebc'