-- Called 99995474470533@default -- Executing [99995474470533@default:1] Dial("Local/99995474470533@default-00000006;2", "SIP/5474470533@kamailio,,tTo") in new stack == Using SIP RTP CoS mark 5 Audio is at 47536 Adding codec opus to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 127.0.0.1:5060: INVITE sip:5474470533@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK7e99ce53;rport Max-Forwards: 70 From: "S2106022200068600051" ;tag=as5ec3ee2e To: Contact: Call-ID: 4eb142d9604a97676d64a372199769dc@127.0.0.1:5070 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.17.2-vici Date: Wed, 02 Jun 2021 14:00:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Remote-Party-ID: "S2106022200068600051" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 306 v=0 o=root 210085128 210085128 IN IP4 127.0.0.1 s=Asterisk PBX 13.17.2-vici c=IN IP4 127.0.0.1 t=0 0 m=audio 47536 RTP/AVP 107 0 101 a=rtpmap:107 opus/48000/2 a=fmtp:107 useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:20 a=sendrecv --- -- Called SIP/5474470533@kamailio <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK7e99ce53;rport=5070;received=127.0.0.1 From: "S2106022200068600051" ;tag=as5ec3ee2e To: Call-ID: 4eb142d9604a97676d64a372199769dc@127.0.0.1:5070 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 127.0.0.1:5070;rport=5070;received=127.0.0.1;branch=z9hG4bK7e99ce53 Record-Route: Record-Route: Call-ID: 4eb142d9604a97676d64a372199769dc@127.0.0.1:5070 From: "S2106022200068600051" ;tag=as5ec3ee2e To: ;tag=KJ6m1dAMFLw63qd6Nx30ohXhaDDbxG3X CSeq: 102 INVITE Contact: "5474470533" Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 <-------------> --- (11 headers 0 lines) --- sip_route_dump: route/path hop: sip_route_dump: route/path hop: -- SIP/kamailio-00000005 is ringing -- Local/99995474470533@default-00000006;1 is ringing == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from 127.0.0.1 <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5070;rport=5070;received=127.0.0.1;branch=z9hG4bK7e99ce53 Record-Route: Record-Route: Call-ID: 4eb142d9604a97676d64a372199769dc@127.0.0.1:5070 From: "S2106022200068600051" ;tag=as5ec3ee2e To: ;tag=KJ6m1dAMFLw63qd6Nx30ohXhaDDbxG3X CSeq: 102 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: "5474470533" Supported: replaces, 100rel, norefersub Content-Type: application/sdp Content-Length: 491 Session-Expires: 90;refresher=uac v=0 o=- 3831631206 3831631207 IN IP4 10.100.1.11 s=pjmedia b=AS:117 t=0 0 a=X-nat:0 m=audio 30070 RTP/AVP 107 101 c=IN IP4 10.100.1.11 b=TIAS:96000 a=ssrc:1930347664 cname:6995a62a5fd0458d a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:107 useinbandfec=1 a=fmtp:101 0-16 a=sendrecv a=rtcp:30071 a=ptime:20 a=candidate:hVGvyh9hGu8X5XwX 1 UDP 2130706431 10.100.1.11 30070 typ host a=candidate:hVGvyh9hGu8X5XwX 2 UDP 2130706430 10.100.1.11 30071 typ host <-------------> --- (14 headers 19 lines) --- Found RTP audio format 107 Found RTP audio format 101 Found audio description format opus for ID 107 Found audio description format telephone-event for ID 101 Capabilities: us - (opus|ulaw), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) > 0x18d76f0 -- Strict RTP learning after remote address set to: 10.100.1.11:30070 Peer audio RTP is at port 10.100.1.11:30070 sip_route_dump: route/path hop: sip_route_dump: route/path hop: Transmitting (NAT) to 127.0.0.1:5060: ACK sip:5474470533@115.132.133.184:59009;ob SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK396c894d;rport Route: , Max-Forwards: 70 From: "S2106022200068600051" ;tag=as5ec3ee2e To: ;tag=KJ6m1dAMFLw63qd6Nx30ohXhaDDbxG3X Contact: Call-ID: 4eb142d9604a97676d64a372199769dc@127.0.0.1:5070 CSeq: 102 ACK User-Agent: Asterisk PBX 13.17.2-vici Content-Length: 0