[Jan 21 17:16:44] Asterisk 13.7.0 built by root @ debian1 on a i686 running Linux on 2016-01-21 14:51:32 UTC [Jan 21 17:16:53] VERBOSE[31319] 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=z9hG4bKPjVR2HDliMH6imPphN1YFyb-UIP8XF34a3 Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: Contact: "333" Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 29297 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=- 148789504 148789504 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 111 18 0 58 118 9 8 58 96 a=rtcp:4037 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 17:16:53] VERBOSE[31319] chan_sip.c: --- (15 headers 19 lines) --- [Jan 21 17:16:53] VERBOSE[31319] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Using INVITE request as basis request - yjo08nkBGlkgv1gORGcfATVEbAUmGzZb [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found peer '333' for '333' from 10.24.18.16:5060 [Jan 21 17:16:53] VERBOSE[31319][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=z9hG4bKPjVR2HDliMH6imPphN1YFyb-UIP8XF34a3;received=10.24.18.16;rport=5060 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as486618b1 Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 29297 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="7ea9c3a4" Content-Length: 0 <------------> [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog 'yjo08nkBGlkgv1gORGcfATVEbAUmGzZb' in 32000 ms (Method: INVITE) [Jan 21 17:16:53] VERBOSE[31319] 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=z9hG4bKPjVR2HDliMH6imPphN1YFyb-UIP8XF34a3 Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as486618b1 Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 29297 ACK Content-Length: 0 <-------------> [Jan 21 17:16:53] VERBOSE[31319] chan_sip.c: --- (8 headers 0 lines) --- [Jan 21 17:16:53] VERBOSE[31319] 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=z9hG4bKPj0lC6A2VvqVmm7B-w2LlyJwk8uGRHr1c7 Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: Contact: "333" Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 29298 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="7ea9c3a4", uri="sip:123@10.24.17.195", response="c40f82530bc5bfd19281d80bdbe81d22", algorithm=MD5 Content-Type: application/sdp Content-Length: 430 v=0 o=- 148789504 148789504 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 111 18 0 58 118 9 8 58 96 a=rtcp:4037 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 17:16:53] VERBOSE[31319] chan_sip.c: --- (16 headers 19 lines) --- [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Using INVITE request as basis request - yjo08nkBGlkgv1gORGcfATVEbAUmGzZb [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found peer '333' for '333' from 10.24.18.16:5060 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 111 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 18 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 0 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 58 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 118 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 9 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 8 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 58 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 96 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format G726-32 for ID 111 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format G729 for ID 18 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format L16 for ID 58 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format L16 for ID 118 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format G722 for ID 9 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format L16-256 for ID 58 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Capabilities: us - (g722), peer - audio=(ulaw|alaw|g722|g729|slin16|g726|slin)/video=(nothing)/text=(nothing), combined - (g722) [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.16:4036 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Looking for 123 in users (domain 10.24.17.195) [Jan 21 17:16:53] VERBOSE[31319][C-00000000] sip/route.c: sip_route_dump: route/path hop: [Jan 21 17:16:53] VERBOSE[31319][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=z9hG4bKPj0lC6A2VvqVmm7B-w2LlyJwk8uGRHr1c7;received=10.24.18.16;rport=5060 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 29298 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 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Audio is at 17216 [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Adding codec g722 to SDP [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Adding codec alaw to SDP [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 17:16:53] VERBOSE[31337][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=z9hG4bK53401a28 Max-Forwards: 70 From: "333" ;tag=as59d0b62d To: Contact: Call-ID: 5e03f5042f326c3f4be367537982d134@10.24.17.195:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.7.0 Date: Thu, 21 Jan 2016 23:16:53 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 2112936065 2112936065 IN IP4 10.24.17.195 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.17.195 t=0 0 m=audio 17216 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 17:16:53] VERBOSE[31319] 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=z9hG4bK53401a28;received=10.24.17.195 From: "333" ;tag=as59d0b62d To: Call-ID: 5e03f5042f326c3f4be367537982d134@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 17:16:53] VERBOSE[31319] chan_sip.c: --- (12 headers 0 lines) --- [Jan 21 17:16:53] VERBOSE[31319] 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=z9hG4bK53401a28;received=10.24.17.195 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Call-ID: 5e03f5042f326c3f4be367537982d134@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: 238 v=0 o=root 105845436 105845436 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 16064 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 17:16:53] VERBOSE[31319] chan_sip.c: --- (14 headers 11 lines) --- [Jan 21 17:16:53] VERBOSE[31319][C-00000000] sip/route.c: sip_route_dump: route/path hop: [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 8 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 96 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.190:16064 [Jan 21 17:16:53] VERBOSE[31319] 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=z9hG4bK53401a28;received=10.24.17.195 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Call-ID: 5e03f5042f326c3f4be367537982d134@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: 238 v=0 o=root 105845436 105845436 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 16064 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 17:16:53] VERBOSE[31319] chan_sip.c: --- (14 headers 11 lines) --- [Jan 21 17:16:53] VERBOSE[31319][C-00000000] sip/route.c: sip_route_dump: route/path hop: [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 17:16:53] VERBOSE[31319][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=z9hG4bK4d6564ed Max-Forwards: 70 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Contact: Call-ID: 5e03f5042f326c3f4be367537982d134@10.24.17.195:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Audio is at 13314 [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Adding codec g722 to SDP [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 17:16:53] VERBOSE[31337][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=z9hG4bKPj0lC6A2VvqVmm7B-w2LlyJwk8uGRHr1c7;received=10.24.18.16;rport=5060 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 29298 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: 234 v=0 o=root 649984430 649984430 IN IP4 10.24.17.195 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.17.195 t=0 0 m=audio 13314 RTP/AVP 9 96 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <------------> [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Audio is at 13314 [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Adding codec g722 to SDP [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 17:16:53] VERBOSE[31337][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=z9hG4bKPj0lC6A2VvqVmm7B-w2LlyJwk8uGRHr1c7;received=10.24.18.16;rport=5060 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 29298 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: 234 v=0 o=root 649984430 649984430 IN IP4 10.24.17.195 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.17.195 t=0 0 m=audio 13314 RTP/AVP 9 96 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=maxptime:150 a=sendrecv <------------> [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Audio is at 17216 [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Adding codec alaw to SDP [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Adding codec g722 to SDP [Jan 21 17:16:53] VERBOSE[31337][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 17:16:53] VERBOSE[31337][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=z9hG4bK0fcd628f Max-Forwards: 70 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Contact: Call-ID: 5e03f5042f326c3f4be367537982d134@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 2112936065 2112936066 IN IP4 10.24.18.16 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.18.16 t=0 0 m=audio 4036 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 17:16:53] VERBOSE[31319] 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=z9hG4bK0fcd628f;received=10.24.17.195 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Call-ID: 5e03f5042f326c3f4be367537982d134@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 17:16:53] VERBOSE[31319] chan_sip.c: --- (12 headers 0 lines) --- [Jan 21 17:16:53] VERBOSE[31319] 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=z9hG4bK0fcd628f;received=10.24.17.195 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Call-ID: 5e03f5042f326c3f4be367537982d134@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: 238 v=0 o=root 105845436 105845437 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 16064 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 17:16:53] VERBOSE[31319] chan_sip.c: --- (14 headers 11 lines) --- [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 8 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 96 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.190:16064 [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:16:53] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 17:16:53] VERBOSE[31319][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=z9hG4bK7d5b771e Max-Forwards: 70 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Contact: Call-ID: 5e03f5042f326c3f4be367537982d134@10.24.17.195:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:16:54] VERBOSE[31319] 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=z9hG4bKPjpiC5uq9DFUnXtqyl2VG47o21q7LlAiwD Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 29298 ACK Content-Length: 0 <-------------> [Jan 21 17:16:54] VERBOSE[31319] chan_sip.c: --- (8 headers 0 lines) --- [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: Audio is at 13314 [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 17:16:54] VERBOSE[31319][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=z9hG4bK646a96bf;rport Max-Forwards: 70 From: ;tag=as698e1ba6 To: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU Contact: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb 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: 194 v=0 o=root 649984430 649984431 IN IP4 10.24.18.190 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.18.190 t=0 0 m=audio 16064 RTP/AVP 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=sendrecv --- [Jan 21 17:16:54] VERBOSE[31319] 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=z9hG4bKPjk7LhK8W7xu4NPxgQQfQqdS1uKoH5ZV5B Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 29299 BYE Content-Length: 0 <-------------> [Jan 21 17:16:54] VERBOSE[31319] chan_sip.c: --- (8 headers 0 lines) --- [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog 'yjo08nkBGlkgv1gORGcfATVEbAUmGzZb' in 32000 ms (Method: BYE) [Jan 21 17:16:54] VERBOSE[31319][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=z9hG4bKPjk7LhK8W7xu4NPxgQQfQqdS1uKoH5ZV5B;received=10.24.18.16;rport=5060 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 29299 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 17:16:54] VERBOSE[31337][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:16:54] VERBOSE[31337][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 17:16:54] VERBOSE[31337][C-00000000] chan_sip.c: Audio is at 17216 [Jan 21 17:16:54] VERBOSE[31337][C-00000000] chan_sip.c: Adding codec alaw to SDP [Jan 21 17:16:54] VERBOSE[31337][C-00000000] chan_sip.c: Adding codec g722 to SDP [Jan 21 17:16:54] VERBOSE[31337][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 21 17:16:54] VERBOSE[31337][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=z9hG4bK0168ada6 Max-Forwards: 70 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Contact: Call-ID: 5e03f5042f326c3f4be367537982d134@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: 260 v=0 o=root 2112936065 2112936067 IN IP4 10.24.17.195 s=Asterisk PBX 13.7.0 c=IN IP4 10.24.17.195 t=0 0 m=audio 17216 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 17:16:54] VERBOSE[31319] 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=z9hG4bK0168ada6;received=10.24.17.195 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Call-ID: 5e03f5042f326c3f4be367537982d134@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 17:16:54] VERBOSE[31319] chan_sip.c: --- (12 headers 0 lines) --- [Jan 21 17:16:54] VERBOSE[31319] 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=z9hG4bK0168ada6;received=10.24.17.195 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Call-ID: 5e03f5042f326c3f4be367537982d134@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: 238 v=0 o=root 105845436 105845438 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 16064 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 17:16:54] VERBOSE[31319] chan_sip.c: --- (14 headers 11 lines) --- [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 8 [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: Found RTP audio format 96 [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.190:16064 [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 17:16:54] VERBOSE[31319][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=z9hG4bK626ca0fe Max-Forwards: 70 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Contact: Call-ID: 5e03f5042f326c3f4be367537982d134@10.24.17.195:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:16:54] VERBOSE[31319] 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=z9hG4bK646a96bf Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb From: ;tag=as698e1ba6 To: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU CSeq: 102 INVITE 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: 218 v=0 o=- 148789504 148789505 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 96 a=rtcp:4037 IN IP4 10.24.18.16 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 17:16:54] VERBOSE[31338][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '5e03f5042f326c3f4be367537982d134@10.24.17.195:5060' in 32000 ms (Method: INVITE) [Jan 21 17:16:54] VERBOSE[31319] chan_sip.c: --- (11 headers 11 lines) --- [Jan 21 17:16:54] VERBOSE[31338][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:16:54] VERBOSE[31338][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.190:5060 [Jan 21 17:16:54] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 17:16:54] VERBOSE[31319][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=z9hG4bK515fbec1;rport Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Contact: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:16:54] VERBOSE[31338][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=z9hG4bK67aa6bf9 Max-Forwards: 70 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Call-ID: 5e03f5042f326c3f4be367537982d134@10.24.17.195:5060 CSeq: 105 BYE User-Agent: Asterisk PBX 13.7.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jan 21 17:16:54] VERBOSE[31319] 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=z9hG4bK67aa6bf9;received=10.24.17.195 From: "333" ;tag=as59d0b62d To: ;tag=as45fc47e8 Call-ID: 5e03f5042f326c3f4be367537982d134@10.24.17.195:5060 CSeq: 105 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 17:16:54] VERBOSE[31319] chan_sip.c: --- (10 headers 0 lines) --- [Jan 21 17:16:54] VERBOSE[31319] chan_sip.c: Really destroying SIP dialog '5e03f5042f326c3f4be367537982d134@10.24.17.195:5060' Method: INVITE [Jan 21 17:16:55] VERBOSE[31319] 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=z9hG4bK646a96bf Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb From: ;tag=as698e1ba6 To: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU CSeq: 102 INVITE 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: 218 v=0 o=- 148789504 148789505 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 96 a=rtcp:4037 IN IP4 10.24.18.16 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 17:16:55] VERBOSE[31319] chan_sip.c: --- (11 headers 11 lines) --- [Jan 21 17:16:55] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:16:55] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 17:16:55] VERBOSE[31319][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=z9hG4bK237de6c8;rport Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Contact: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:16:56] VERBOSE[31319] 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=z9hG4bK646a96bf Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb From: ;tag=as698e1ba6 To: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU CSeq: 102 INVITE 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: 218 v=0 o=- 148789504 148789505 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 96 a=rtcp:4037 IN IP4 10.24.18.16 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 17:16:56] VERBOSE[31319] chan_sip.c: --- (11 headers 11 lines) --- [Jan 21 17:16:56] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:16:56] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 17:16:56] VERBOSE[31319][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=z9hG4bK5384c4b7;rport Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Contact: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:16:58] VERBOSE[31319] 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=z9hG4bK646a96bf Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb From: ;tag=as698e1ba6 To: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU CSeq: 102 INVITE 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: 218 v=0 o=- 148789504 148789505 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 96 a=rtcp:4037 IN IP4 10.24.18.16 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 17:16:58] VERBOSE[31319] chan_sip.c: --- (11 headers 11 lines) --- [Jan 21 17:16:58] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:16:58] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 17:16:58] VERBOSE[31319][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=z9hG4bK2298ca0a;rport Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Contact: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:17:02] VERBOSE[31319] 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=z9hG4bK646a96bf Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb From: ;tag=as698e1ba6 To: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU CSeq: 102 INVITE 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: 218 v=0 o=- 148789504 148789505 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 96 a=rtcp:4037 IN IP4 10.24.18.16 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 17:17:02] VERBOSE[31319] chan_sip.c: --- (11 headers 11 lines) --- [Jan 21 17:17:02] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:17:02] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 17:17:02] VERBOSE[31319][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=z9hG4bK026936f9;rport Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Contact: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:17:06] VERBOSE[31319] 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=z9hG4bK646a96bf Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb From: ;tag=as698e1ba6 To: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU CSeq: 102 INVITE 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: 218 v=0 o=- 148789504 148789505 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 96 a=rtcp:4037 IN IP4 10.24.18.16 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 17:17:06] VERBOSE[31319] chan_sip.c: --- (11 headers 11 lines) --- [Jan 21 17:17:06] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:17:06] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 17:17:06] VERBOSE[31319][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=z9hG4bK4419dec9;rport Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Contact: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:17:06] VERBOSE[31319] chan_sip.c: Really destroying SIP dialog 'fvCAfxQoDLDfUaFxoo-SDdrQJjEkcsKA' Method: REGISTER [Jan 21 17:17:10] VERBOSE[31319] 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=z9hG4bK646a96bf Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb From: ;tag=as698e1ba6 To: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU CSeq: 102 INVITE 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: 218 v=0 o=- 148789504 148789505 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 96 a=rtcp:4037 IN IP4 10.24.18.16 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 17:17:10] VERBOSE[31319] chan_sip.c: --- (11 headers 11 lines) --- [Jan 21 17:17:10] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:17:10] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 17:17:10] VERBOSE[31319][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=z9hG4bK53615a46;rport Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Contact: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:17:14] VERBOSE[31319] 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=z9hG4bK646a96bf Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb From: ;tag=as698e1ba6 To: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU CSeq: 102 INVITE 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: 218 v=0 o=- 148789504 148789505 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 96 a=rtcp:4037 IN IP4 10.24.18.16 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 17:17:14] VERBOSE[31319] chan_sip.c: --- (11 headers 11 lines) --- [Jan 21 17:17:14] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:17:14] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 17:17:14] VERBOSE[31319][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=z9hG4bK09857fb5;rport Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Contact: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:17:18] VERBOSE[31319] 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=z9hG4bK646a96bf Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb From: ;tag=as698e1ba6 To: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU CSeq: 102 INVITE 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: 218 v=0 o=- 148789504 148789505 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 96 a=rtcp:4037 IN IP4 10.24.18.16 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 17:17:18] VERBOSE[31319] chan_sip.c: --- (11 headers 11 lines) --- [Jan 21 17:17:18] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:17:18] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 17:17:18] VERBOSE[31319][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=z9hG4bK183384f8;rport Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Contact: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:17:22] VERBOSE[31319] 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=z9hG4bK646a96bf Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb From: ;tag=as698e1ba6 To: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU CSeq: 102 INVITE 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: 218 v=0 o=- 148789504 148789505 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 96 a=rtcp:4037 IN IP4 10.24.18.16 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 17:17:22] VERBOSE[31319] chan_sip.c: --- (11 headers 11 lines) --- [Jan 21 17:17:22] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:17:22] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 17:17:22] VERBOSE[31319][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=z9hG4bK63b08fdb;rport Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Contact: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:17:26] VERBOSE[31319] 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=z9hG4bK646a96bf Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb From: ;tag=as698e1ba6 To: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU CSeq: 102 INVITE 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: 218 v=0 o=- 148789504 148789505 IN IP4 10.24.18.16 s=digphn c=IN IP4 10.24.18.16 t=0 0 a=X-nat:0 m=audio 4036 RTP/AVP 96 a=rtcp:4037 IN IP4 10.24.18.16 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> [Jan 21 17:17:26] VERBOSE[31319] chan_sip.c: --- (11 headers 11 lines) --- [Jan 21 17:17:26] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 21 17:17:26] VERBOSE[31319][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Jan 21 17:17:26] VERBOSE[31319][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=z9hG4bK43b3a2aa;rport Max-Forwards: 70 From: "333" ;tag=Q9smPlsbBZ-xV3mr6py9Q.7x8CQidAcU To: ;tag=as698e1ba6 Contact: Call-ID: yjo08nkBGlkgv1gORGcfATVEbAUmGzZb CSeq: 102 ACK User-Agent: Asterisk PBX 13.7.0 Content-Length: 0 --- [Jan 21 17:17:26] VERBOSE[31319] chan_sip.c: Really destroying SIP dialog 'yjo08nkBGlkgv1gORGcfATVEbAUmGzZb' Method: BYE [Jan 21 17:17:27] VERBOSE[31336] asterisk.c: Asterisk cleanly ending (0). [Jan 21 17:17:27] VERBOSE[31336] asterisk.c: Executing last minute cleanups