<-------------> [690/1947] --- (8 headers 0 lines) --- <--- SIP read from UDP:172.16.3.47:5060 ---> INVITE sip:+982191079724@77.104.118.87 SIP/2.0 Record-Route: Record-Route: Record-Route: Via: SIP/2.0/UDP 172.16.3.47;branch=z9hG4bKfd.64e4a0ebcfdeae32f4ca34948a8acfeb.0 Via: SIP/2.0/UDP 172.16.3.72;branch=z9hG4bKfd.a8b652d1cc67d55112e215a7cae505fc.0 Via: SIP/2.0/UDP 172.22.132.235:11000;received=172.22.132.235;rport=11000;branch=z9hG4bK67vy8FZZvS45a Max-Forwards: 68 From: ;tag=gcySKe01N9B3r To: Call-ID: 027d3ee873218b8abbc81f5ae29ad5a3ec4a CSeq: 22980326 INVITE Contact: User-Agent: 2600hz Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-descript ion, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 292 X-FS-Support: update_display,send_info Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1595051214 1595051215 IN IP4 172.16.3.72 s=FreeSWITCH c=IN IP4 172.16.3.72 t=0 0 m=audio 33026 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=rtpmap:13 CN/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:33027 a=ptime:20 --- (8 headers 0 lines) --- == Using SIP RTP CoS mark 5 Audio is at 12390 Adding codec ulaw to SDP Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 172.16.3.47:5060: INVITE sip:+982191079724@172.16.3.47:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.3.16:5060;branch=z9hG4bK46475606;rport Max-Forwards: 70 From: ;tag=as6476e354 To: Contact: Call-ID: 3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon CSeq: 102 INVITE User-Agent: Respina SW Date: Sat, 18 Jul 2020 10:30:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 261 v=0 o=root 617581892 617581892 IN IP4 172.16.3.16 s=RespinaSW c=IN IP4 172.16.3.16 t=0 0 m=audio 12390 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- -- Called SIP/+982191079724 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:172.16.3.47:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 172.16.3.16:5060;branch=z9hG4bK46475606;rport=5060;received=172.16.3.16 From: ;tag=as6476e354 To: Call-ID: 3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon CSeq: 102 INVITE Server: kamailio (4.4.5 (x86_64/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:172.16.3.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.3.16:5060;received=172.16.3.16;branch=z9hG4bK46475606;rport=5060 Record-Route: Record-Route: Record-Route: Record-Route: From: ;tag=as6476e354 To: ;tag=B4Ft55NXUKptF Call-ID: 3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon CSeq: 102 INVITE Contact: User-Agent: 2600hz Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-descript ion, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 916 Remote-Party-ID: "undefined" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1595044128 1595044129 IN IP4 172.16.3.72 s=FreeSWITCH c=IN IP4 172.16.3.72 t=0 0 a=msid-semantic: WMS pJeLcD6iTGpgkhgk3JjbOOC9qEvaNRC3 m=audio 33036 RTP/AVP 0 101 a=ice-ufrag:GCTlgmlwQGxuoZnF a=ice-pwd:wRYxiqhf78nW6GED8nOfz0Cf a=candidate:4415515794 1 udp 659136 172.22.132.229 24108 typ host generation 0 a=candidate:4415515794 2 udp 659135 172.22.132.229 24109 typ host generation 0 a=ssrc:2601969844 cname:qHEtbABU7R1iY7sR a=ssrc:2601969844 msid:pJeLcD6iTGpgkhgk3JjbOOC9qEvaNRC3 a0 a=ssrc:2601969844 mslabel:pJeLcD6iTGpgkhgk3JjbOOC9qEvaNRC3 a=ssrc:2601969844 label:pJeLcD6iTGpgkhgk3JjbOOC9qEvaNRC3a0 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:33037 a=ptime:20 a=candidate:K6lstDg6SjdEgcq2 1 UDP 2130706431 172.16.3.72 33036 typ host a=candidate:K6lstDg6SjdEgcq2 2 UDP 2130706430 172.16.3.72 33037 typ host a=end-of-candidates -------------> [315/1947] --- (20 headers 24 lines) --- Got SDP version 1595044129 and unique parts [FreeSWITCH 1595044128 IN IP4 172.16.3.72] == Using UDPTL CoS mark 5 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|alaw), 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|) > 0x7f3df0059480 -- Strict RTP learning after remote address set to: 172.16.3.72:33036 Peer audio RTP is at port 172.16.3.72:33036 sip_route_dump: route/path hop: sip_route_dump: route/path hop: sip_route_dump: route/path hop: sip_route_dump: route/path hop: Transmitting (NAT) to 172.16.3.47:5060: ACK sip:+982191079724@172.22.132.229:11000;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.3.16:5060;branch=z9hG4bK74460c23;rport Route: ,,, Max-Forwards: 70 From: ;tag=as6476e354 To: ;tag=B4Ft55NXUKptF Contact: Call-ID: 3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon CSeq: 102 ACK User-Agent: Respina SW Content-Length: 0 --- -- SIP/+982191079724-0000001a answered SIP/+982191079726-00000019 Audio is at 10684 Adding codec ulaw to SDP Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 172.16.3.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.3.47;branch=z9hG4bKfd.64e4a0ebcfdeae32f4ca34948a8acfeb.0;received=172.16.3.47;rport=5060 Via: SIP/2.0/UDP 172.16.3.72;branch=z9hG4bKfd.a8b652d1cc67d55112e215a7cae505fc.0 Via: SIP/2.0/UDP 172.22.132.235:11000;received=172.22.132.235;rport=11000;branch=z9hG4bK67vy8FZZvS45a Record-Route: Record-Route: Record-Route: From: ;tag=gcySKe01N9B3r To: ;tag=as316dc0a9 Call-ID: 027d3ee873218b8abbc81f5ae29ad5a3ec4a CSeq: 22980326 INVITE Server: Respina SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 1475923560 1475923560 IN IP4 172.16.3.16 s=RespinaSW c=IN IP4 172.16.3.16 t=0 0 m=audio 10684 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <------------> -- Channel SIP/+982191079724-0000001a joined 'simple_bridge' basic-bridge -- Channel SIP/+982191079726-00000019 joined 'simple_bridge' basic-bridge <--- SIP read from UDP:172.16.3.47:5060 ---> ACK sip:+982191079724@172.16.3.16:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.3.47;branch=z9hG4bKfd.1675f21150718234589feed97431f660.0 Via: SIP/2.0/UDP 172.16.3.72;branch=z9hG4bKfd.99bc5ded371d9fdafb836ff46f007187.0 Via: SIP/2.0/UDP 172.22.132.235:11000;received=172.22.132.235;rport=11000;branch=z9hG4bK7gpQaBg3S2trp Max-Forwards: 68 From: ;tag=gcySKe01N9B3r To: ;tag=as316dc0a9 Call-ID: 027d3ee873218b8abbc81f5ae29ad5a3ec4a CSeq: 22980326 ACK Contact: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- > 0x7f3e0c03dac0 -- Strict RTP switching to RTP target address 172.16.3.72:33026 as source > 0x7f3df0059480 -- Strict RTP switching to RTP target address 172.16.3.72:33036 as source <--- SIP read from UDP:172.16.3.47:5060 ---> INVITE sip:+982191079726@172.16.3.16:5060;alias=172.16.3.47~5060~1 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 172.16.3.47;branch=z9hG4bKf5e5.2ab6765c814878660bc100b684258e1e.0 Via: SIP/2.0/UDP 172.16.3.72;branch=z9hG4bKf5e5.1fbfc8869aeb3fb6d083811c6f84197b.0 Via: SIP/2.0/UDP 172.22.132.229;rport=5060;branch=z9hG4bKf5e5.ccee7e283f8b08aa6dd3fb8f1b74f6eb.0 Via: SIP/2.0/UDP 172.22.132.229:11000;received=172.22.132.229;rport=11000;branch=z9hG4bK7egZreBQHtyyD Max-Forwards: 47 From: ;tag=B4Ft55NXUKptF To: ;tag=as6476e354 Call-ID: 3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon CSeq: 22980326 INVITE Contact: User-Agent: 2600hz Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Content-Type: application/sdp Content-Length: 338v=0 o=FreeSWITCH 1595044128 1595044130 IN IP4 172.16.3.72 s=FreeSWITCH c=IN IP4 172.16.3.72 t=0 0 m=image 33036 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=sendrecv a=ptime:20 <-------------> [186/1947] --- (17 headers 15 lines) --- Sending to 172.16.3.47:5060 (NAT) Comparing SDP version 1595044129 -> 1595044130 and unique parts [FreeSWITCH 1595044128 IN IP4 172.16.3.72] -> [FreeSWITCH 1595 044128 IN IP4 172.16.3.72] Got T.38 offer in SDP in dialog 3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon Capabilities: us - (ulaw|alaw), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. <--- Transmitting (NAT) to 172.16.3.47:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.3.47;branch=z9hG4bKf5e5.2ab6765c814878660bc100b684258e1e.0;received=172.16.3.47;rport=5060 Via: SIP/2.0/UDP 172.16.3.72;branch=z9hG4bKf5e5.1fbfc8869aeb3fb6d083811c6f84197b.0 Via: SIP/2.0/UDP 172.22.132.229;rport=5060;branch=z9hG4bKf5e5.ccee7e283f8b08aa6dd3fb8f1b74f6eb.0 Via: SIP/2.0/UDP 172.22.132.229:11000;received=172.22.132.229;rport=11000;branch=z9hG4bK7egZreBQHtyyD Record-Route: From: ;tag=B4Ft55NXUKptF To: ;tag=as6476e354 Call-ID: 3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon CSeq: 22980326 INVITE Server: Respina SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> == Using UDPTL CoS mark 5 Reliably Transmitting (NAT) to 172.16.3.47:5060: INVITE sip:mod_sofia@172.22.132.235:11000;alias=172.22.132.235~11000~1 SIP/2.0 Via: SIP/2.0/UDP 172.16.3.16:5060;branch=z9hG4bK739d3186;rport Route: ,, Max-Forwards: 70 From: ;tag=as316dc0a9 To: ;tag=gcySKe01N9B3r Contact: Call-ID: 027d3ee873218b8abbc81f5ae29ad5a3ec4a CSeq: 102 INVITE User-Agent: Respina SW 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: 282 v=0 o=root 1475923560 1475923561 IN IP4 172.16.3.16 s=RespinaSW c=IN IP4 172.16.3.16 t=0 0 m=image 4674 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:397 a=T38FaxUdpEC:t38UDPRedundancy--- <--- SIP read from UDP:172.16.3.47:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 172.16.3.16:5060;branch=z9hG4bK739d3186;rport=5060;received=172.16.3.16 From: ;tag=as316dc0a9 To: ;tag=gcySKe01N9B3r Call-ID: 027d3ee873218b8abbc81f5ae29ad5a3ec4a CSeq: 102 INVITE Server: kamailio (4.4.5 (x86_64/linux)) Content-Length: 0<-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:172.16.3.47:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.16.3.16:5060;received=172.16.3.16;branch=z9hG4bK739d3186;rport=5060 From: ;tag=as316dc0a9 To: ;tag=gcySKe01N9B3r Call-ID: 027d3ee873218b8abbc81f5ae29ad5a3ec4a CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Transmitting (NAT) to 172.16.3.47:5060: ACK sip:mod_sofia@172.22.132.235:11000;alias=172.22.132.235~11000~1 SIP/2.0 Via: SIP/2.0/UDP 172.16.3.16:5060;branch=z9hG4bK739d3186;rport Route: ,, Max-Forwards: 70 From: ;tag=as316dc0a9 To: ;tag=gcySKe01N9B3r Contact: Call-ID: 027d3ee873218b8abbc81f5ae29ad5a3ec4a CSeq: 102 ACK User-Agent: Respina SW Content-Length: 0 -- -- Channel SIP/+982191079726-00000019 left 'simple_bridge' basic-bridge == Spawn extension (public, +982191079724, 4) exited non-zero on 'SIP/+982191079726-00000019' -- Channel SIP/+982191079724-0000001a left 'simple_bridge' basic-bridge Scheduling destruction of SIP dialog '3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '027d3ee873218b8abbc81f5ae29ad5a3ec4a' in 32000 ms (Method: ACK) Reliably Transmitting (NAT) to 172.16.3.47:5060: BYE sip:mod_sofia@172.22.132.235:11000;alias=172.22.132.235~11000~1 SIP/2.0 Via: SIP/2.0/UDP 172.16.3.16:5060;branch=z9hG4bK1eaa355d;rport Route: ,, Max-Forwards: 70 From: ;tag=as316dc0a9 To: ;tag=gcySKe01N9B3r Call-ID: 027d3ee873218b8abbc81f5ae29ad5a3ec4a CSeq: 103 BYE User-Agent: Respina SW Reason: Q.850;cause=1 X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 Content-Length: 0 --- <--- SIP read from UDP:172.16.3.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.3.16:5060;received=172.16.3.16;branch=z9hG4bK1eaa355d;rport=5060 From: ;tag=as316dc0a9 To: ;tag=gcySKe01N9B3r Call-ID: 027d3ee873218b8abbc81f5ae29ad5a3ec4a CSeq: 103 BYE User-Agent: 2600hz Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '027d3ee873218b8abbc81f5ae29ad5a3ec4a' Method: ACK<--- Reliably Transmitting (NAT) to 172.16.3.47:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 172.16.3.47;branch=z9hG4bKf5e5.2ab6765c814878660bc100b684258e1e.0;received=172.16.3.47;rport=5060 Via: SIP/2.0/UDP 172.16.3.72;branch=z9hG4bKf5e5.1fbfc8869aeb3fb6d083811c6f84197b.0 Via: SIP/2.0/UDP 172.22.132.229;rport=5060;branch=z9hG4bKf5e5.ccee7e283f8b08aa6dd3fb8f1b74f6eb.0 Via: SIP/2.0/UDP 172.22.132.229:11000;received=172.22.132.229;rport=11000;branch=z9hG4bK7egZreBQHtyyD From: ;tag=B4Ft55NXUKptF To: ;tag=as6476e354 Call-ID: 3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon CSeq: 22980326 INVITE Server: Respina SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Reason: Q.850;cause=1 Content-Length: 0 <------------> <--- SIP read from UDP:172.16.3.47:5060 ---> ACK sip:+982191079726@172.16.3.16:5060;alias=172.16.3.47~5060~1 SIP/2.0 Via: SIP/2.0/UDP 172.16.3.47;branch=z9hG4bKf5e5.2ab6765c814878660bc100b684258e1e.0 Max-Forwards: 47 From: ;tag=B4Ft55NXUKptF To: ;tag=as6476e354 Call-ID: 3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon CSeq: 22980326 ACK Content-Length: 0<-------------> --- (8 headers 0 lines) --- Reliably Transmitting (NAT) to 172.16.3.47:5060: BYE sip:+982191079724@172.22.132.229:11000;alias=172.22.132.229~5060~1;transport=udp;alias=172.22.132.229~11000~1 SIP/2.0 Via: SIP/2.0/UDP 172.16.3.16:5060;branch=z9hG4bK14883a37;rport Route: ,,, Max-Forwards: 70 From: ;tag=as6476e354 To: ;tag=B4Ft55NXUKptF Call-ID: 3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon CSeq: 103 BYE User-Agent: Respina SW Reason: Q.850;cause=1 X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 Content-Length: 0 --- Scheduling destruction of SIP dialog '3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon' in 32000 ms (Method: ACK) <--- SIP read from UDP:172.16.3.47:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 172.16.3.16:5060;received=172.16.3.16;branch=z9hG4bK14883a37;rport=5060 From: ;tag=as6476e354 To: ;tag=B4Ft55NXUKptF Call-ID: 3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon CSeq: 103 BYE Content-Length: 0<-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '3b6b24c1465c9c862cec22f0198178f0@testdev.nexfon' Method: ACK trunk2-sw*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups