<--- Received SIP request (1118 bytes) from UDP:108.41.213.129:63450 ---> INVITE sip:205@155.138.217.63:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.173.52;branch=z9hG4bKe07c679eD4D4AC1D From: "201" ;tag=F03F1D54-E191A52B To: CSeq: 1 INVITE Call-ID: c903e3be18c6e663172c749c6bbdb6bc Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomVVX-VVX_411-UA/6.3.0.14929 Accept-Language: en Supported: replaces,100rel Allow-Events: conference,talk,hold Max-Forwards: 70 Content-Type: application/sdp Content-Length: 517 v=0 o=- 1597268538 1597268538 IN IP4 10.10.173.52 s=Polycom IP Phone c=IN IP4 10.10.173.52 t=0 0 a=sendrecv m=audio 2264 RTP/AVP 107 0 9 102 8 127 126 a=rtpmap:107 opus/48000/2 a=fmtp:107 maxplaybackrate=16000; sprop-maxcapturerate=16000; maxptime=20; ptime=20; maxaveragebitrate=24000; cbr=0; useinbandfec=0; usedtx=0 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=rtpmap:126 telephone-event/48000 <--- Transmitting SIP response (514 bytes) to UDP:108.41.213.129:63450 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.173.52;rport=63450;received=108.41.213.129;branch=z9hG4bKe07c679eD4D4AC1D Call-ID: c903e3be18c6e663172c749c6bbdb6bc From: "201" ;tag=F03F1D54-E191A52B To: ;tag=z9hG4bKe07c679eD4D4AC1D CSeq: 1 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1597268538/3a5191f684220d8cafec187eb70491d3",opaque="0a8b502a62b338d7",algorithm=md5,qop="auth" Server: FPBX-14.0.13.26(13.35.0) Content-Length: 0 <--- Received SIP request (526 bytes) from UDP:108.41.213.129:63450 ---> ACK sip:205@155.138.217.63:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.173.52;branch=z9hG4bKe07c679eD4D4AC1D From: "201" ;tag=F03F1D54-E191A52B To: ;tag=z9hG4bKe07c679eD4D4AC1D CSeq: 1 ACK Call-ID: c903e3be18c6e663172c749c6bbdb6bc Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomVVX-VVX_411-UA/6.3.0.14929 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <--- Received SIP request (1407 bytes) from UDP:108.41.213.129:63450 ---> INVITE sip:205@155.138.217.63:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.173.52;branch=z9hG4bKfc1a0bd8D388BF From: "201" ;tag=F03F1D54-E191A52B To: CSeq: 2 INVITE Call-ID: c903e3be18c6e663172c749c6bbdb6bc Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomVVX-VVX_411-UA/6.3.0.14929 Accept-Language: en Supported: replaces,100rel Allow-Events: conference,talk,hold Authorization: Digest username="201", realm="asterisk", nonce="1597268538/3a5191f684220d8cafec187eb70491d3", qop=auth, cnonce="+zRrPWOprRb8DEY", nc=00000001, opaque="0a8b502a62b338d7", uri="sip:205@155.138.217.63:5060;user=phone", response="a0496e0554bbe7e27393d2a7450081d9", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 517 v=0 o=- 1597268538 1597268538 IN IP4 10.10.173.52 s=Polycom IP Phone c=IN IP4 10.10.173.52 t=0 0 a=sendrecv m=audio 2264 RTP/AVP 107 0 9 102 8 127 126 a=rtpmap:107 opus/48000/2 a=fmtp:107 maxplaybackrate=16000; sprop-maxcapturerate=16000; maxptime=20; ptime=20; maxaveragebitrate=24000; cbr=0; useinbandfec=0; usedtx=0 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=rtpmap:126 telephone-event/48000 == Setting global variable 'SIPDOMAIN' to '155.138.217.63' <--- Transmitting SIP response (332 bytes) to UDP:108.41.213.129:63450 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.173.52;rport=63450;received=108.41.213.129;branch=z9hG4bKfc1a0bd8D388BF Call-ID: c903e3be18c6e663172c749c6bbdb6bc From: "201" ;tag=F03F1D54-E191A52B To: CSeq: 2 INVITE Server: FPBX-14.0.13.26(13.35.0) Content-Length: 0 == Using SIP RTP Audio TOS bits 184 == Using SIP RTP Audio TOS bits 184 in TCLASS field. == Using SIP RTP Audio CoS mark 5 -- Executing [205@from-test-custom:1] Set("PJSIP/201-00000002", "PJSIP_MEDIA_OFFER(audio)=!all,g722") in new stack -- Executing [205@from-test-custom:2] Set("PJSIP/201-00000002", "PJSIP_SEND_SESSION_REFRESH()=invite") in new stack -- Executing [205@from-test-custom:3] Dial("PJSIP/201-00000002", "PJSIP/205,,b(gosub-jc-custom^s^1)") in new stack -- PJSIP/205-00000003 Internal Gosub(gosub-jc-custom,s,1) start -- Executing [s@gosub-jc-custom:1] Set("PJSIP/205-00000003", "PJSIP_MEDIA_OFFER(audio)=!all,g722") in new stack -- Executing [s@gosub-jc-custom:2] Set("PJSIP/205-00000003", "PJSIP_SEND_SESSION_REFRESH()=invite") in new stack -- Executing [s@gosub-jc-custom:3] Return("PJSIP/205-00000003", "") in new stack == Spawn extension (from-test-custom, 205, 1) exited non-zero on 'PJSIP/205-00000003' -- PJSIP/205-00000003 Internal Gosub(gosub-jc-custom,s,1) complete GOSUB_RETVAL= -- Called PJSIP/205 == Using SIP RTP Audio TOS bits 184 == Using SIP RTP Audio TOS bits 184 in TCLASS field. == Using SIP RTP Audio CoS mark 5 <--- Transmitting SIP request (961 bytes) to UDP:108.41.213.129:53142 ---> INVITE sip:205@108.41.213.129:53142 SIP/2.0 Via: SIP/2.0/UDP 155.138.217.63:5060;rport;branch=z9hG4bKPjacb03209-cec5-4c7c-ae19-eecb31f5f387 From: "201" ;tag=fb5b46dc-fa20-4fc6-a501-691646c9bac2 To: Contact: Call-ID: 7be0ceb6-0911-4941-9135-ea98542f7eef CSeq: 14336 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "201" Max-Forwards: 70 User-Agent: FPBX-14.0.13.26(13.35.0) Content-Type: application/sdp Content-Length: 239 v=0 o=- 484635969 484635969 IN IP4 155.138.217.63 s=Asterisk c=IN IP4 155.138.217.63 t=0 0 m=audio 13878 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <--- Received SIP response (349 bytes) from UDP:108.41.213.129:53142 ---> SIP/2.0 100 Trying To: From: "201" ;tag=fb5b46dc-fa20-4fc6-a501-691646c9bac2 Call-ID: 7be0ceb6-0911-4941-9135-ea98542f7eef CSeq: 14336 INVITE Via: SIP/2.0/UDP 155.138.217.63:5060;branch=z9hG4bKPjacb03209-cec5-4c7c-ae19-eecb31f5f387;rport=5060 Server: Cisco/SPA504G-7.6.2f Content-Length: 0 <--- Received SIP response (413 bytes) from UDP:108.41.213.129:53142 ---> SIP/2.0 180 Ringing To: ;tag=d727b84b37070c9bi1 From: "201" ;tag=fb5b46dc-fa20-4fc6-a501-691646c9bac2 Call-ID: 7be0ceb6-0911-4941-9135-ea98542f7eef CSeq: 14336 INVITE Via: SIP/2.0/UDP 155.138.217.63:5060;branch=z9hG4bKPjacb03209-cec5-4c7c-ae19-eecb31f5f387;rport=5060 Contact: Server: Cisco/SPA504G-7.6.2f Content-Length: 0 -- PJSIP/205-00000003 is ringing -- PJSIP/205-00000003 is ringing <--- Transmitting SIP response (585 bytes) to UDP:108.41.213.129:63450 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.173.52;rport=63450;received=108.41.213.129;branch=z9hG4bKfc1a0bd8D388BF Call-ID: c903e3be18c6e663172c749c6bbdb6bc From: "201" ;tag=F03F1D54-E191A52B To: ;tag=dd3444ef-ecad-4cbb-976d-8c44e874cec9 CSeq: 2 INVITE Server: FPBX-14.0.13.26(13.35.0) Contact: Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER P-Asserted-Identity: "205" Content-Length: 0 <--- Received SIP response (739 bytes) from UDP:108.41.213.129:53142 ---> SIP/2.0 200 OK To: ;tag=d727b84b37070c9bi1 From: "201" ;tag=fb5b46dc-fa20-4fc6-a501-691646c9bac2 Call-ID: 7be0ceb6-0911-4941-9135-ea98542f7eef CSeq: 14336 INVITE Via: SIP/2.0/UDP 155.138.217.63:5060;branch=z9hG4bKPjacb03209-cec5-4c7c-ae19-eecb31f5f387;rport=5060 Contact: Server: Cisco/SPA504G-7.6.2f Content-Length: 206 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 44523 44523 IN IP4 10.10.173.231 s=- c=IN IP4 10.10.173.231 t=0 0 m=audio 16414 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <--- Transmitting SIP request (413 bytes) to UDP:108.41.213.129:53142 ---> ACK sip:205@108.41.213.129:53142 SIP/2.0 Via: SIP/2.0/UDP 155.138.217.63:5060;rport;branch=z9hG4bKPjd2ebcb0a-1935-4160-92de-d3b4e91b53e6 From: "201" ;tag=fb5b46dc-fa20-4fc6-a501-691646c9bac2 To: ;tag=d727b84b37070c9bi1 Call-ID: 7be0ceb6-0911-4941-9135-ea98542f7eef CSeq: 14336 ACK Max-Forwards: 70 User-Agent: FPBX-14.0.13.26(13.35.0) Content-Length: 0 <--- Transmitting SIP request (1032 bytes) to UDP:108.41.213.129:53142 ---> INVITE sip:205@108.41.213.129:53142 SIP/2.0 Via: SIP/2.0/UDP 155.138.217.63:5060;rport;branch=z9hG4bKPj6f9a93b8-95af-4beb-8f28-6a3b689faeb9 From: "201" ;tag=fb5b46dc-fa20-4fc6-a501-691646c9bac2 To: ;tag=d727b84b37070c9bi1 Contact: Call-ID: 7be0ceb6-0911-4941-9135-ea98542f7eef CSeq: 14337 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "201" Max-Forwards: 70 User-Agent: FPBX-14.0.13.26(13.35.0) Content-Type: application/sdp Content-Length: 287 v=0 o=- 484635969 484635970 IN IP4 155.138.217.63 s=Asterisk c=IN IP4 155.138.217.63 t=0 0 m=audio 13878 RTP/AVP 0 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv -- PJSIP/205-00000003 answered PJSIP/201-00000002 <--- Transmitting SIP response (951 bytes) to UDP:108.41.213.129:63450 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.173.52;rport=63450;received=108.41.213.129;branch=z9hG4bKfc1a0bd8D388BF Call-ID: c903e3be18c6e663172c749c6bbdb6bc From: "201" ;tag=F03F1D54-E191A52B To: ;tag=dd3444ef-ecad-4cbb-976d-8c44e874cec9 CSeq: 2 INVITE Server: FPBX-14.0.13.26(13.35.0) Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Contact: Supported: 100rel, timer, replaces, norefersub P-Asserted-Identity: "205" Content-Type: application/sdp Content-Length: 289 v=0 o=- 1597268538 1597268540 IN IP4 155.138.217.63 s=Asterisk c=IN IP4 155.138.217.63 t=0 0 m=audio 18102 RTP/AVP 0 9 8 127 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=ptime:20 a=maxptime:150 a=sendrecv -- Channel PJSIP/205-00000003 joined 'simple_bridge' basic-bridge -- Channel PJSIP/201-00000002 joined 'simple_bridge' basic-bridge <--- Received SIP request (524 bytes) from UDP:108.41.213.129:63450 ---> ACK sip:155.138.217.63:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.173.52;branch=z9hG4bKe083181c44F0D113 From: "201" ;tag=F03F1D54-E191A52B To: ;tag=dd3444ef-ecad-4cbb-976d-8c44e874cec9 CSeq: 2 ACK Call-ID: c903e3be18c6e663172c749c6bbdb6bc Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomVVX-VVX_411-UA/6.3.0.14929 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <--- Received SIP response (647 bytes) from UDP:108.41.213.129:53142 ---> SIP/2.0 200 OK To: ;tag=d727b84b37070c9bi1 From: "201" ;tag=fb5b46dc-fa20-4fc6-a501-691646c9bac2 Call-ID: 7be0ceb6-0911-4941-9135-ea98542f7eef CSeq: 14337 INVITE Via: SIP/2.0/UDP 155.138.217.63:5060;branch=z9hG4bKPj6f9a93b8-95af-4beb-8f28-6a3b689faeb9;rport=5060 Contact: Server: Cisco/SPA504G-7.6.2f Content-Length: 206 Content-Type: application/sdp v=0 o=- 44523 44524 IN IP4 10.10.173.231 s=- c=IN IP4 10.10.173.231 t=0 0 m=audio 16414 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <--- Transmitting SIP request (413 bytes) to UDP:108.41.213.129:53142 ---> ACK sip:205@108.41.213.129:53142 SIP/2.0 Via: SIP/2.0/UDP 155.138.217.63:5060;rport;branch=z9hG4bKPj8836160b-6e12-4020-905b-33da0ac02046 From: "201" ;tag=fb5b46dc-fa20-4fc6-a501-691646c9bac2 To: ;tag=d727b84b37070c9bi1 Call-ID: 7be0ceb6-0911-4941-9135-ea98542f7eef CSeq: 14337 ACK Max-Forwards: 70 User-Agent: FPBX-14.0.13.26(13.35.0) Content-Length: 0 vultr*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0).