<--- SIP read from UDP:10.8.15.45:5063 ---> INVITE sip:014509615003@10.9.0.94:5060 SIP/2.0 Via: SIP/2.0/UDP 10.8.15.45:5063;branch=z9hG4bK2021811806 From: "100" ;tag=1997694706 To: Call-ID: 169586255@10.8.15.45 CSeq: 1 INVITE Contact: Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T48G 35.72.0.6 Supported: replaces Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 277 v=0 o=- 20045 20045 IN IP4 10.8.15.45 s=SDP data c=IN IP4 10.8.15.45 t=0 0 m=audio 5054 RTP/AVP 0 18 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv <-------------> --- (14 headers 14 lines) --- Sending to 10.8.15.45:5063 (no NAT) Sending to 10.8.15.45:5063 (no NAT) Using INVITE request as basis request - 169586255@10.8.15.45 Found peer '100' for '100' from 10.8.15.45:5063 <--- Reliably Transmitting (no NAT) to 10.8.15.45:5063 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.8.15.45:5063;branch=z9hG4bK2021811806;received=10.8.15.45 From: "100" ;tag=1997694706 To: ;tag=as00cfa8c4 Call-ID: 169586255@10.8.15.45 CSeq: 1 INVITE Server: Asterisk PBX 15.3.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6563a378" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '169586255@10.8.15.45' in 32000 ms (Method: INVITE) <--- SIP read from UDP:10.8.15.45:5063 ---> ACK sip:014509615003@10.9.0.94:5060 SIP/2.0 Via: SIP/2.0/UDP 10.8.15.45:5063;branch=z9hG4bK2021811806 From: "100" ;tag=1997694706 To: ;tag=as00cfa8c4 Call-ID: 169586255@10.8.15.45 CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:10.8.15.45:5063 ---> INVITE sip:014509615003@10.9.0.94:5060 SIP/2.0 Via: SIP/2.0/UDP 10.8.15.45:5063;branch=z9hG4bK118354455 From: "100" ;tag=1997694706 To: Call-ID: 169586255@10.8.15.45 CSeq: 2 INVITE Contact: Authorization: Digest username="100", realm="asterisk", nonce="6563a378", uri="sip:014509615003@10.9.0.94:5060", response="4dd56d2c977113c0c23cc4f44591f1b9", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T48G 35.72.0.6 Supported: replaces Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 277 v=0 o=- 20045 20045 IN IP4 10.8.15.45 s=SDP data c=IN IP4 10.8.15.45 t=0 0 m=audio 5054 RTP/AVP 0 18 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv <-------------> --- (15 headers 14 lines) --- Sending to 10.8.15.45:5063 (no NAT) Using INVITE request as basis request - 169586255@10.8.15.45 Found peer '100' for '100' from 10.8.15.45:5063 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 9 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw), peer - audio=(ulaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) > 0x7f781000a1a0 -- Strict RTP learning after remote address set to: 10.8.15.45:5054 Peer audio RTP is at port 10.8.15.45:5054 Looking for 014509615003 in phones_outgoing (domain 10.9.0.94) sip_route_dump: route/path hop: <--- Transmitting (no NAT) to 10.8.15.45:5063 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.8.15.45:5063;branch=z9hG4bK118354455;received=10.8.15.45 From: "100" ;tag=1997694706 To: Call-ID: 169586255@10.8.15.45 CSeq: 2 INVITE Server: Asterisk PBX 15.3.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [014509615003@phones_outgoing:1] Dial("SIP/100-00000000", "SIP/4509615003@provider1") in new stack == Using SIP RTP CoS mark 5 Audio is at 24570 Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.12.187:5080: INVITE sip:4509615003@linsanity.mtl.broadsoft.com SIP/2.0 Via: SIP/2.0/TLS 10.9.0.94:5061;branch=z9hG4bK0c057a77 Max-Forwards: 70 From: "100" ;tag=as37e1711f To: Contact: Call-ID: 266a700e12b5695c6d4b2fe60fbe6ad7@linsanity.mtl.broadsoft.com CSeq: 102 INVITE User-Agent: Asterisk PBX 15.3.0 Date: Thu, 29 Mar 2018 11:16:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "100" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 330 v=0 o=root 1859053209 1859053209 IN IP4 10.9.0.94 s=Asterisk PBX 15.3.0 c=IN IP4 10.9.0.94 t=0 0 m=audio 24570 RTP/SAVP 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BAvH6zSSvR6IDwYm/qwMkqhfDftoBWUGZ6ChpPld a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- -- Called SIP/4509615003@provider1 <--- SIP read from TLS:192.168.12.187:5080 ---> SIP/2.0 180 Ringing Via:SIP/2.0/TLS 10.9.0.94:5061;branch=z9hG4bK0c057a77 From:"100";tag=as37e1711f To:;tag=100147382-1522321118456 Call-ID:266a700e12b5695c6d4b2fe60fbe6ad7@linsanity.mtl.broadsoft.com CSeq:102 INVITE Supported:altc Contact: Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE X-BroadWorks-Correlation-Info:a4d36cf3-c4ab-4d68-b6ec-1457d0b9642c Content-Length:0 <-------------> --- (11 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/provider1-00000001 is ringing <--- Transmitting (no NAT) to 10.8.15.45:5063 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.8.15.45:5063;branch=z9hG4bK118354455;received=10.8.15.45 From: "100" ;tag=1997694706 To: ;tag=as0318e391 Call-ID: 169586255@10.8.15.45 CSeq: 2 INVITE Server: Asterisk PBX 15.3.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> <--- SIP read from TLS:192.168.12.187:5080 ---> SIP/2.0 200 OK Via:SIP/2.0/TLS 10.9.0.94:5061;branch=z9hG4bK0c057a77 From:"100";tag=as37e1711f To:;tag=100147382-1522321118456 Call-ID:266a700e12b5695c6d4b2fe60fbe6ad7@linsanity.mtl.broadsoft.com CSeq:102 INVITE Supported:altc Contact: Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:application/btbc-session-info,application/conference-info+xml,application/dtmf-relay,application/media_control+xml,application/sdp X-BroadWorks-Correlation-Info:a4d36cf3-c4ab-4d68-b6ec-1457d0b9642c Content-Type:application/sdp Content-Length:305 v=0 o=BroadWorks 19431 1 IN IP4 192.168.12.192 s=- c=IN IP4 192.168.12.192 t=0 0 m=audio 13020 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:3156087703 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:b89Z+NLqJ8QTx52BlK6tSWpXGtOxUPFPEQhRpNn0 <-------------> --- (13 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) > 0x7f77d4005c00 -- Strict RTP learning after remote address set to: 192.168.12.192:13020 Peer audio RTP is at port 192.168.12.192:13020 sip_route_dump: route/path hop: Transmitting (no NAT) to 192.168.12.187:5080: ACK sip:192.168.12.187:5080;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.9.0.94:5061;branch=z9hG4bK1280e99b Max-Forwards: 70 From: "100" ;tag=as37e1711f To: ;tag=100147382-1522321118456 Contact: Call-ID: 266a700e12b5695c6d4b2fe60fbe6ad7@linsanity.mtl.broadsoft.com CSeq: 102 ACK User-Agent: Asterisk PBX 15.3.0 Content-Length: 0 --- -- SIP/provider1-00000001 answered SIP/100-00000000 Audio is at 13354 Adding codec ulaw to SDP <--- Reliably Transmitting (no NAT) to 10.8.15.45:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.8.15.45:5063;branch=z9hG4bK118354455;received=10.8.15.45 From: "100" ;tag=1997694706 To: ;tag=as0318e391 Call-ID: 169586255@10.8.15.45 CSeq: 2 INVITE Server: Asterisk PBX 15.3.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 187 v=0 o=root 828583263 828583263 IN IP4 10.9.0.94 s=Asterisk PBX 15.3.0 c=IN IP4 10.9.0.94 t=0 0 m=audio 13354 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:150 a=sendrecv <------------> -- Channel SIP/provider1-00000001 joined 'simple_bridge' basic-bridge <26686fd3-9def-4316-bb0f-baf4f30a3706> -- Channel SIP/100-00000000 joined 'simple_bridge' basic-bridge <26686fd3-9def-4316-bb0f-baf4f30a3706> > 0x7f77d4005c00 -- Strict RTP switching to RTP target address 192.168.12.192:13020 as source <--- SIP read from UDP:10.8.15.45:5063 ---> ACK sip:014509615003@10.9.0.94:5060 SIP/2.0 Via: SIP/2.0/UDP 10.8.15.45:5063;branch=z9hG4bK920046021 From: "100" ;tag=1997694706 To: ;tag=as0318e391 Call-ID: 169586255@10.8.15.45 CSeq: 2 ACK Contact: Max-Forwards: 70 User-Agent: Yealink SIP-T48G 35.72.0.6 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- > 0x7f781000a1a0 -- Strict RTP switching to RTP target address 10.8.15.45:5054 as source > 0x7f781000a1a0 -- Strict RTP learning complete - Locking on source address 10.8.15.45:5054 > 0x7f77d4005c00 -- Strict RTP learning complete - Locking on source address 192.168.12.192:13020 <--- SIP read from UDP:10.8.15.45:5063 ---> BYE sip:014509615003@10.9.0.94:5060 SIP/2.0 Via: SIP/2.0/UDP 10.8.15.45:5063;branch=z9hG4bK510856582 From: "100" ;tag=1997694706 To: ;tag=as0318e391 Call-ID: 169586255@10.8.15.45 CSeq: 3 BYE Contact: Authorization: Digest username="100", realm="asterisk", nonce="6563a378", uri="sip:014509615003@10.9.0.94:5060", response="a9f1726806c2c46849bb7285681df54a", algorithm=MD5 Max-Forwards: 70 User-Agent: Yealink SIP-T48G 35.72.0.6 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.8.15.45:5063 (no NAT) Scheduling destruction of SIP dialog '169586255@10.8.15.45' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 10.8.15.45:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.8.15.45:5063;branch=z9hG4bK510856582;received=10.8.15.45 From: "100" ;tag=1997694706 To: ;tag=as0318e391 Call-ID: 169586255@10.8.15.45 CSeq: 3 BYE Server: Asterisk PBX 15.3.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> -- Channel SIP/100-00000000 left 'simple_bridge' basic-bridge <26686fd3-9def-4316-bb0f-baf4f30a3706> -- Channel SIP/provider1-00000001 left 'simple_bridge' basic-bridge <26686fd3-9def-4316-bb0f-baf4f30a3706> Scheduling destruction of SIP dialog '266a700e12b5695c6d4b2fe60fbe6ad7@linsanity.mtl.broadsoft.com' in 32000 ms (Method: INVITE) == Spawn extension (phones_outgoing, 014509615003, 1) exited non-zero on 'SIP/100-00000000' Reliably Transmitting (no NAT) to 192.168.12.187:5080: BYE sip:192.168.12.187:5080;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.9.0.94:5061;branch=z9hG4bK2368557e Max-Forwards: 70 From: "100" ;tag=as37e1711f To: ;tag=100147382-1522321118456 Call-ID: 266a700e12b5695c6d4b2fe60fbe6ad7@linsanity.mtl.broadsoft.com CSeq: 103 BYE User-Agent: Asterisk PBX 15.3.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from TLS:192.168.12.187:5080 ---> SIP/2.0 200 OK Via:SIP/2.0/TLS 10.9.0.94:5061;branch=z9hG4bK2368557e From:"100";tag=as37e1711f To:;tag=100147382-1522321118456 Call-ID:266a700e12b5695c6d4b2fe60fbe6ad7@linsanity.mtl.broadsoft.com CSeq:103 BYE Content-Length:0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '266a700e12b5695c6d4b2fe60fbe6ad7@linsanity.mtl.broadsoft.com' Method: INVITE <--- SIP read from UDP:10.8.15.45:5063 ---> <-------------> Asterisk cleanly ending (0). Executing last minute cleanups == Destroying musiconhold processes == Manager unregistered action DBGet == Manager unregistered action DBPut == Manager unregistered action DBDel == Manager unregistered action DBDelTree [root@vlin-094 asterisk]#