[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 18:26:57] VERBOSE[31459] chan_sip.c: Really destroying SIP dialog 'h0htmQ92Vl23gn42ztp24XMJuaTqlS6P' Method: REGISTER [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.16:5060 ---> INVITE sip:123@10.24.17.195 SIP/2.0 Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjOe3s44qX-pUd83nE66aftfiMEP-VbR5x Max-Forwards: 70 From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod To: Contact: "333" Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f CSeq: 27930 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: Digium D40 1_4_2_0_63880 Content-Type: application/sdp Content-Length: 430 v=0 o=- 148793728 148793728 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4038 RTP/AVP 111 18 0 58 118 9 8 58 96 a=rtcp:4039 IN IP4 10.24.18.16 a=rtpmap:111 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:58 L16/16000 a=rtpmap:118 L16/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:58 L16-256/16000 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: --- (15 headers 19 lines) --- [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Using INVITE request as basis request - FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found peer '333' for '333' from 10.24.18.16:5060 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.24.18.16:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjOe3s44qX-pUd83nE66aftfiMEP-VbR5x;received=10.24.18.16;rport=5060 From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod To: ;tag=as428dd4e7 Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f CSeq: 27930 INVITE Server: Asterisk PBX 13.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="365d7cc9" Content-Length: 0 <------------> [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog 'FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f' in 32000 ms (Method: INVITE) [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.16:5060 ---> ACK sip:123@10.24.17.195 SIP/2.0 Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjOe3s44qX-pUd83nE66aftfiMEP-VbR5x Max-Forwards: 70 From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod To: ;tag=as428dd4e7 Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f CSeq: 27930 ACK Content-Length: 0 <-------------> [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: --- (8 headers 0 lines) --- [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.16:5060 ---> INVITE sip:123@10.24.17.195 SIP/2.0 Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjEFgx3zb40XYC8kUZHSSbZV1teC.bjvhN Max-Forwards: 70 From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod To: Contact: "333" Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f CSeq: 27931 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: Digium D40 1_4_2_0_63880 Authorization: Digest username="333", realm="asterisk", nonce="365d7cc9", uri="sip:123@10.24.17.195", response="574a419b0e669568763a7aa93e66f381", algorithm=MD5 Content-Type: application/sdp Content-Length: 430 v=0 o=- 148793728 148793728 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4038 RTP/AVP 111 18 0 58 118 9 8 58 96 a=rtcp:4039 IN IP4 10.24.18.16 a=rtpmap:111 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:58 L16/16000 a=rtpmap:118 L16/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:58 L16-256/16000 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: --- (16 headers 19 lines) --- [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Using INVITE request as basis request - FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found peer '333' for '333' from 10.24.18.16:5060 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 111 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 18 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 0 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 58 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 118 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 9 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 8 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 58 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 96 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format G726-32 for ID 111 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format G729 for ID 18 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format L16 for ID 58 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format L16 for ID 118 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format G722 for ID 9 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format L16-256 for ID 58 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Capabilities: us - (g722|alaw), peer - audio=(ulaw|alaw|g722|g729|slin16|g726|slin)/video=(nothing)/text=(nothing), combined - (g722|alaw) [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.16:4038 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Looking for 123 in users (domain 10.24.17.195) [Jan 21 18:27:17] VERBOSE[31459][C-00000000] sip/route.c: sip_route_dump: route/path hop: [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 10.24.18.16:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjEFgx3zb40XYC8kUZHSSbZV1teC.bjvhN;received=10.24.18.16;rport=5060 From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod To: Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f CSeq: 27931 INVITE Server: Asterisk PBX 13.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Audio is at 16704 [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Adding codec g722 to SDP [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Adding codec alaw to SDP [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.190:5060: INVITE sip:123@10.24.18.190 SIP/2.0 Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK68ab447c Max-Forwards: 70 From: "333" ;tag=as579facad To: Contact: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.7.0 Date: Fri, 22 Jan 2016 00:27:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 260 v=0 o=root 2107721231 2107721231 IN IP4 10.24.17.195 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.17.195 t=0 0 m=audio 16704 RTP/AVP 9 8 96 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv --- [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.190:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK68ab447c;received=10.24.17.195 From: "333" ;tag=as579facad To: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 102 INVITE Server: Asterisk PBX 13.7.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: --- (12 headers 0 lines) --- [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.190:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK68ab447c;received=10.24.17.195 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 102 INVITE Server: Asterisk PBX 13.7.0-rc3 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: 240 v=0 o=root 1696456408 1696456408 IN IP4 10.24.18.190 s=Asterisk PBX 13.7.0-rc3 c=IN IP4 10.24.18.190 t=0 0 m=audio 13926 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <-------------> [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: --- (14 headers 11 lines) --- [Jan 21 18:27:17] VERBOSE[31459][C-00000000] sip/route.c: sip_route_dump: route/path hop: [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 8 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 96 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.190:13926 [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK68ab447c;received=10.24.17.195 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 102 INVITE Server: Asterisk PBX 13.7.0-rc3 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: 240 v=0 o=root 1696456408 1696456408 IN IP4 10.24.18.190 s=Asterisk PBX 13.7.0-rc3 c=IN IP4 10.24.18.190 t=0 0 m=audio 13926 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <-------------> [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: --- (14 headers 11 lines) --- [Jan 21 18:27:17] VERBOSE[31459][C-00000000] sip/route.c: sip_route_dump: route/path hop: [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Transmitting (no NAT) to 10.24.18.190:5060: ACK sip:123@10.24.18.190:5060 SIP/2.0 Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK0859ac62 Max-Forwards: 70 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Contact: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Audio is at 19876 [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Adding codec g722 to SDP [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Adding codec alaw to SDP [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 10.24.18.16:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjEFgx3zb40XYC8kUZHSSbZV1teC.bjvhN;received=10.24.18.16;rport=5060 From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod To: ;tag=as1d33b1c9 Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f CSeq: 27931 INVITE Server: Asterisk PBX 13.7.0 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: 258 v=0 o=root 991912202 991912202 IN IP4 10.24.17.195 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.17.195 t=0 0 m=audio 19876 RTP/AVP 9 8 96 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <------------> [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Audio is at 19876 [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Adding codec g722 to SDP [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Adding codec alaw to SDP [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.24.18.16:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjEFgx3zb40XYC8kUZHSSbZV1teC.bjvhN;received=10.24.18.16;rport=5060 From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod To: ;tag=as1d33b1c9 Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f CSeq: 27931 INVITE Server: Asterisk PBX 13.7.0 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: 258 v=0 o=root 991912202 991912202 IN IP4 10.24.17.195 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.17.195 t=0 0 m=audio 19876 RTP/AVP 9 8 96 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <------------> [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Audio is at 16704 [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Adding codec alaw to SDP [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Adding codec g722 to SDP [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 18:27:17] VERBOSE[31480][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.190:5060: INVITE sip:123@10.24.18.190:5060 SIP/2.0 Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3e43084f Max-Forwards: 70 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Contact: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 13.7.0 Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 257 v=0 o=root 2107721231 2107721232 IN IP4 10.24.18.16 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.18.16 t=0 0 m=audio 4038 RTP/AVP 8 9 96 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv --- [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.190:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3e43084f;received=10.24.17.195 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 103 INVITE Server: Asterisk PBX 13.7.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: --- (12 headers 0 lines) --- [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3e43084f;received=10.24.17.195 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 103 INVITE Server: Asterisk PBX 13.7.0-rc3 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: 240 v=0 o=root 1696456408 1696456409 IN IP4 10.24.18.190 s=Asterisk PBX 13.7.0-rc3 c=IN IP4 10.24.18.190 t=0 0 m=audio 13926 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <-------------> [Jan 21 18:27:17] VERBOSE[31459] chan_sip.c: --- (14 headers 11 lines) --- [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 8 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 96 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.190:13926 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 18:27:17] VERBOSE[31459][C-00000000] chan_sip.c: Transmitting (no NAT) to 10.24.18.190:5060: ACK sip:123@10.24.18.190:5060 SIP/2.0 Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3773871a Max-Forwards: 70 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Contact: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 18:27:18] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.16:5060 ---> ACK sip:123@10.24.17.195:5060 SIP/2.0 Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPj8oprM1L1inGkvR3aWT7DvD2s9ZmUvvtZ Max-Forwards: 70 From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod To: ;tag=as1d33b1c9 Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f CSeq: 27931 ACK Content-Length: 0 <-------------> [Jan 21 18:27:18] VERBOSE[31459] chan_sip.c: --- (8 headers 0 lines) --- [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Audio is at 19876 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Adding codec alaw to SDP [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.16:5060: INVITE sip:333@10.24.18.16:5060;ob SIP/2.0 Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK7bd3ae06;rport Max-Forwards: 70 From: ;tag=as1d33b1c9 To: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod Contact: Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f CSeq: 102 INVITE User-Agent: Asterisk PBX 13.7.0 Session-Expires: 1800;refresher=uac Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 234 v=0 o=root 991912202 991912203 IN IP4 10.24.18.190 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.18.190 t=0 0 m=audio 13926 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv --- [Jan 21 18:27:18] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.16:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.24.17.195:5060;rport=5060;received=10.24.17.195;branch=z9hG4bK7bd3ae06 Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f From: ;tag=as1d33b1c9 To: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod CSeq: 102 INVITE Session-Expires: 1800;refresher=uac Contact: "333" Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 242 v=0 o=- 148793728 148793729 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4038 RTP/AVP 8 96 a=rtcp:4039 IN IP4 10.24.18.16 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 18:27:18] VERBOSE[31459] chan_sip.c: --- (12 headers 12 lines) --- [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 8 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 96 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.16:4038 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Transmitting (no NAT) to 10.24.18.16:5060: ACK sip:333@10.24.18.16:5060;ob SIP/2.0 Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK5227f473;rport Max-Forwards: 70 From: ;tag=as1d33b1c9 To: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod Contact: Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 18:27:18] VERBOSE[31481][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 18:27:18] VERBOSE[31481][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 18:27:18] VERBOSE[31481][C-00000000] chan_sip.c: Audio is at 16704 [Jan 21 18:27:18] VERBOSE[31481][C-00000000] chan_sip.c: Adding codec alaw to SDP [Jan 21 18:27:18] VERBOSE[31481][C-00000000] chan_sip.c: Adding codec g722 to SDP [Jan 21 18:27:18] VERBOSE[31481][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 18:27:18] VERBOSE[31481][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.190:5060: INVITE sip:123@10.24.18.190:5060 SIP/2.0 Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3d6f45c0 Max-Forwards: 70 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Contact: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 13.7.0 Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 257 v=0 o=root 2107721231 2107721233 IN IP4 10.24.18.16 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.18.16 t=0 0 m=audio 4038 RTP/AVP 8 9 96 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv --- [Jan 21 18:27:18] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.190:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3d6f45c0;received=10.24.17.195 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 104 INVITE Server: Asterisk PBX 13.7.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Jan 21 18:27:18] VERBOSE[31459] chan_sip.c: --- (12 headers 0 lines) --- [Jan 21 18:27:18] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK3d6f45c0;received=10.24.17.195 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 104 INVITE Server: Asterisk PBX 13.7.0-rc3 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: 240 v=0 o=root 1696456408 1696456410 IN IP4 10.24.18.190 s=Asterisk PBX 13.7.0-rc3 c=IN IP4 10.24.18.190 t=0 0 m=audio 13926 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <-------------> [Jan 21 18:27:18] VERBOSE[31459] chan_sip.c: --- (14 headers 11 lines) --- [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 8 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 96 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.190:13926 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 18:27:18] VERBOSE[31459][C-00000000] chan_sip.c: Transmitting (no NAT) to 10.24.18.190:5060: ACK sip:123@10.24.18.190:5060 SIP/2.0 Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK072e53d7 Max-Forwards: 70 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Contact: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 18:27:32] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.16:5060 ---> BYE sip:123@10.24.17.195:5060 SIP/2.0 Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjjxYkSqj0XqK8c32J7ykFLzMWe7rg5mDo Max-Forwards: 70 From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod To: ;tag=as1d33b1c9 Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f CSeq: 27932 BYE User-Agent: Digium D40 1_4_2_0_63880 Content-Length: 0 <-------------> [Jan 21 18:27:32] VERBOSE[31459] chan_sip.c: --- (9 headers 0 lines) --- [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog 'FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f' in 32000 ms (Method: BYE) [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 10.24.18.16:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjjxYkSqj0XqK8c32J7ykFLzMWe7rg5mDo;received=10.24.18.16;rport=5060 From: "333" ;tag=9TAc34PhBLOhwgPl7B8ICcsFQ7iJfSod To: ;tag=as1d33b1c9 Call-ID: FCKPz3prHiy.EYHQYVr3QZmITa6-hd1f CSeq: 27932 BYE Server: Asterisk PBX 13.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Jan 21 18:27:32] VERBOSE[31480][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 18:27:32] VERBOSE[31480][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 18:27:32] VERBOSE[31480][C-00000000] chan_sip.c: Audio is at 16704 [Jan 21 18:27:32] VERBOSE[31480][C-00000000] chan_sip.c: Adding codec alaw to SDP [Jan 21 18:27:32] VERBOSE[31480][C-00000000] chan_sip.c: Adding codec g722 to SDP [Jan 21 18:27:32] VERBOSE[31480][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 18:27:32] VERBOSE[31480][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.190:5060: INVITE sip:123@10.24.18.190:5060 SIP/2.0 Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK690650b6 Max-Forwards: 70 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Contact: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 105 INVITE User-Agent: Asterisk PBX 13.7.0 Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 260 v=0 o=root 2107721231 2107721234 IN IP4 10.24.17.195 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.17.195 t=0 0 m=audio 16704 RTP/AVP 8 9 96 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv --- [Jan 21 18:27:32] VERBOSE[31481][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' in 32000 ms (Method: INVITE) [Jan 21 18:27:32] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.190:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK690650b6;received=10.24.17.195 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 105 INVITE Server: Asterisk PBX 13.7.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [Jan 21 18:27:32] VERBOSE[31459] chan_sip.c: --- (12 headers 0 lines) --- [Jan 21 18:27:32] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK690650b6;received=10.24.17.195 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 105 INVITE Server: Asterisk PBX 13.7.0-rc3 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: 240 v=0 o=root 1696456408 1696456411 IN IP4 10.24.18.190 s=Asterisk PBX 13.7.0-rc3 c=IN IP4 10.24.18.190 t=0 0 m=audio 13926 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <-------------> [Jan 21 18:27:32] VERBOSE[31459] chan_sip.c: --- (14 headers 11 lines) --- [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 8 [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: Found RTP audio format 96 [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.190:13926 [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: Transmitting (no NAT) to 10.24.18.190:5060: ACK sip:123@10.24.18.190:5060 SIP/2.0 Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK246a9484 Max-Forwards: 70 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Contact: Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 105 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.190:5060: BYE sip:123@10.24.18.190:5060 SIP/2.0 Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK1a071eca Max-Forwards: 70 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 106 BYE User-Agent: Asterisk PBX 13.7.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jan 21 18:27:32] VERBOSE[31459][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' in 32000 ms (Method: INVITE) [Jan 21 18:27:32] VERBOSE[31459] chan_sip.c: <--- SIP read from UDP:10.24.18.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.24.17.195:5060;branch=z9hG4bK1a071eca;received=10.24.17.195 From: "333" ;tag=as579facad To: ;tag=as1e3a359e Call-ID: 2acabcf37565694c4d407489038d76a9@10.24.17.195:5060 CSeq: 106 BYE Server: Asterisk PBX 13.7.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jan 21 18:27:32] VERBOSE[31459] chan_sip.c: --- (10 headers 0 lines) --- [Jan 21 18:27:32] VERBOSE[31459] chan_sip.c: Really destroying SIP dialog '2acabcf37565694c4d407489038d76a9@10.24.17.195:5060' Method: INVITE [Jan 21 18:27:36] VERBOSE[31478] asterisk.c: Asterisk cleanly ending (0). [Jan 21 18:27:36] VERBOSE[31478] asterisk.c: Executing last minute cleanups