<div dir="ltr"><div dir="ltr"><br></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Jan 12, 2022 at 5:09 PM Jerry Geis <<a href="mailto:jerry.geis@gmail.com">jerry.geis@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr">I am running 18.8.0 -  videosupport is enabled. I get video calls no problem.<div><br></div><div>However when I make a call file to a soft phone and include:</div><div>Codecs: ulaw,h264<br></div><div>in the call file...</div><div><br></div><div>sip show channels - shows:</div><div>4013c15f1f4cdff  (ulaw|h264)      No       Tx: ACK  </div><div>so clearly the caller has h264.</div><div><br></div><div>Then when I "automatically" request another softphone to join my conf bridge...</div><div>the soft phone rings, and answers - all I get is audio and sip show channels for that device:</div><div>5c77cf1455e4afc  (ulaw)           No       Tx: ACK </div><div><br></div><div>How do I get Video in the confbridge ?</div><div><br></div><div>Thanks</div><div><br></div><div>Jerry <br></div></div></blockquote><div><br></div><div><br></div><div><br></div><div>hi Josh,</div><div><br></div><div>here is the sip debug... It shows the the first call negotiate video - but the second call to bring the end video device into the conf - no video negotitation.</div><div><br></div><div>Audio is at 15542<br>Adding codec ulaw to SDP<br>Adding codec alaw to SDP<br>Adding codec gsm to SDP<br></div><div><br></div><div>Thanks,</div><div><br></div><div>Jerry</div><div><br></div><div><br></div><div>Asterisk 18.8.0, Copyright (C) 1999 - 2021, Sangoma Technologies Corporation and others.<br>Created by Mark Spencer <<a href="mailto:markster@digium.com">markster@digium.com</a>><br>Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.<br>This is free software, with components licensed under the GNU General Public<br>License version 2 and other licenses; you are welcome to redistribute it under<br>certain conditions. Type 'core show license' for details.<br>=========================================================================<br>Running as user 'silentm'<br>Running under group 'silentm'<br>Connected to Asterisk 18.8.0 currently running on DevKaufer (pid = 597669)<br>Really destroying SIP dialog 'c24843e8-d7f1-0740-08dd-8b79fe39a15a' Method: REGISTER<br><br><--- SIP read from UDP:<a href="http://192.168.2.22:5060">192.168.2.22:5060</a> ---><br><br><br><br><-------------><br>  == Using SIP VIDEO CoS mark 6<br>  == Using SIP RTP CoS mark 5<br>Audio is at 17816<br>Video is at <a href="http://192.168.1.6:10746">192.168.1.6:10746</a><br>Adding video codec vp8 to SDP<br>Adding codec ulaw to SDP<br>Adding codec opus to SDP<br>Reliably Transmitting (NAT) to <a href="http://192.168.1.6:48124">192.168.1.6:48124</a>:<br>INVITE sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss SIP/2.0<br>Via: SIP/2.0/WS 192.168.1.6:5060;branch=z9hG4bK73b689b9;rport<br>Max-Forwards: 70<br>From: "Mason Kaufer 34" <<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as101db932<br>To: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss><br>Contact: <sip:527@192.168.1.6:5060;transport=ws><br>Call-ID: <a href="http://4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060">4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 18.8.0<br>Date: Thu, 13 Jan 2022 13:46:18 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>Content-Length: 1106<br><br>v=0<br>o=root 1174630673 1174630673 IN IP4 192.168.1.6<br>s=Asterisk PBX 18.8.0<br>c=IN IP4 192.168.1.6<br>b=CT:5120<br>t=0 0<br>m=audio 17816 UDP/TLS/RTP/SAVPF 0 107<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:107 opus/48000/2<br>a=maxptime:60<br>a=ice-ufrag:4ff9bfd157a3896a6bc7f86d312dde00<br>a=ice-pwd:2c8b8f052875a1cd7096d71478ff3567<br>a=candidate:Hc0a80106 1 UDP 2130706431 192.168.1.6 17816 typ host<br>a=candidate:Hc0a80106 2 UDP 2130706430 192.168.1.6 17817 typ host<br>a=connection:new<br>a=setup:passive<br>a=fingerprint:SHA-256 0D:4D:96:97:A7:EB:A2:62:4E:43:10:C9:64:9E:5D:43:C5:00:08:2D:D7:1D:01:0C:83:99:B9:49:77:64:24:AF<br>a=rtcp-mux<br>a=sendrecv<br>m=video 10746 UDP/TLS/RTP/SAVPF 100<br>a=ice-ufrag:0c2c1ae221c4578666475d5455d11e6f<br>a=ice-pwd:7eadf35c40a33c2f2bd87431669b60b2<br>a=candidate:Hc0a80106 1 UDP 2130706431 192.168.1.6 10746 typ host<br>a=candidate:Hc0a80106 2 UDP 2130706430 192.168.1.6 10747 typ host<br>a=connection:new<br>a=setup:passive<br>a=fingerprint:SHA-256 0D:4D:96:97:A7:EB:A2:62:4E:43:10:C9:64:9E:5D:43:C5:00:08:2D:D7:1D:01:0C:83:99:B9:49:77:64:24:AF<br>a=rtpmap:100 VP8/90000<br>a=rtcp-fb:* ccm fir<br>a=rtcp-mux<br>a=sendrecv<br><br>---<br>    -- Called mason.kaufer.visualcampus<br><br><--- SIP read from WS:<a href="http://192.168.1.6:48124">192.168.1.6:48124</a> ---><br>SIP/2.0 100 Trying (sent from the Transaction Layer)<br>Via: SIP/2.0/WS 192.168.1.6:5060;rport=5060;branch=z9hG4bK73b689b9<br>From: "Mason Kaufer 34"<<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as101db932<br>To: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss><br>Call-ID: <a href="http://4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060">4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060</a><br>CSeq: 102 INVITE<br>Content-Length: 0<br><br><br><-------------><br>--- (7 headers 0 lines) ---<br><br><--- SIP read from WS:<a href="http://192.168.1.6:48124">192.168.1.6:48124</a> ---><br>SIP/2.0 180 Ringing<br>Via: SIP/2.0/WS 192.168.1.6:5060;rport=5060;branch=z9hG4bK73b689b9<br>From: "Mason Kaufer 34"<<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as101db932<br>To: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;tag=HULiDWhvD78SNfAPBUqC<br>Contact: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;transport=wss><br>Call-ID: <a href="http://4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060">4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060</a><br>CSeq: 102 INVITE<br>Content-Length: 0<br>Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE<br><br><br><-------------><br>--- (9 headers 0 lines) ---<br>sip_route_dump: route/path hop: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;transport=wss><br>    -- SIP/mason.kaufer.visualcampus-0000004b is ringing<br>       > 0x7f8eac0141f0 -- Strict RTP learning after remote address set to: <a href="http://192.168.1.6:56634">192.168.1.6:56634</a><br>       > 0x7f8eac00b800 -- Strict RTP learning after remote address set to: <a href="http://192.168.1.6:32953">192.168.1.6:32953</a><br><br><--- SIP read from WS:<a href="http://192.168.1.6:48124">192.168.1.6:48124</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/WS 192.168.1.6:5060;rport=5060;branch=z9hG4bK73b689b9<br>From: "Mason Kaufer 34"<<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as101db932<br>To: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;tag=HULiDWhvD78SNfAPBUqC<br>Contact: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;transport=wss><br>Call-ID: <a href="http://4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060">4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060</a><br>CSeq: 102 INVITE<br>Content-Type: application/sdp<br>Content-Length: 1824<br>Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE<br><br>v=0<br>o=- 6700265476590515000 2 IN IP4 127.0.0.1<br>s=Cloudonix WebRTC Client - chrome<br>t=0 0<br>a=msid-semantic: WMS NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE<br>m=audio 56634 UDP/TLS/RTP/SAVPF 0 107<br>c=IN IP4 192.168.1.6<br>a=rtcp:9 IN IP4 0.0.0.0<br>a=candidate:505434299 1 udp 2122260223 192.168.1.6 56634 typ host generation 0 network-id 1<br>a=ice-ufrag:tt5f<br>a=ice-pwd:t6HFMwvAhMBcsLzbNv6ZsBN7<br>a=ice-options:trickle<br>a=fingerprint:sha-256 BC:DF:CE:46:D5:23:0D:50:52:1D:9A:E8:5C:ED:66:B9:4D:8A:73:8C:83:3C:20:75:8E:BC:D5:19:A4:28:50:74<br>a=setup:active<br>a=mid:0<br>a=sendrecv<br>a=msid:NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE eca112c3-1a46-4c88-8ab0-822a8db6f24e<br>a=rtcp-mux<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:107 opus/48000/2<br>a=fmtp:107 minptime=10;useinbandfec=1<br>a=ssrc:902560899 cname:g/TRvw9o4VgxF0Qi<br>a=ssrc:902560899 msid:NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE eca112c3-1a46-4c88-8ab0-822a8db6f24e<br>a=ssrc:902560899 mslabel:NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE<br>a=ssrc:902560899 label:eca112c3-1a46-4c88-8ab0-822a8db6f24e<br>m=video 32953 UDP/TLS/RTP/SAVPF 100<br>c=IN IP4 192.168.1.6<br>a=rtcp:9 IN IP4 0.0.0.0<br>a=candidate:505434299 1 udp 2122260223 192.168.1.6 32953 typ host generation 0 network-id 1<br>a=ice-ufrag:iQuk<br>a=ice-pwd:V3WS4tt1M2TwSqCs+sNWzhXP<br>a=ice-options:trickle<br>a=fingerprint:sha-256 BC:DF:CE:46:D5:23:0D:50:52:1D:9A:E8:5C:ED:66:B9:4D:8A:73:8C:83:3C:20:75:8E:BC:D5:19:A4:28:50:74<br>a=setup:active<br>a=mid:1<br>a=sendrecv<br>a=msid:NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE 1ed3f295-431e-492d-a1a3-cfa356566ea1<br>a=rtcp-mux<br>a=rtpmap:100 VP8/90000<br>a=rtcp-fb:100 ccm fir<br>a=ssrc:3172627963 cname:g/TRvw9o4VgxF0Qi<br>a=ssrc:3172627963 msid:NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE 1ed3f295-431e-492d-a1a3-cfa356566ea1<br>a=ssrc:3172627963 mslabel:NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE<br>a=ssrc:3172627963 label:1ed3f295-431e-492d-a1a3-cfa356566ea1<br><br><-------------><br>--- (10 headers 44 lines) ---<br>Got SDP version 2 and unique parts [- 6700265476590515000 IN IP4 127.0.0.1]<br>Found RTP audio format 0<br>Found RTP audio format 107<br>Found audio description format PCMU for ID 0<br>Found audio description format opus for ID 107<br>Found RTP video format 100<br>Found video description format VP8 for ID 100<br>Capabilities: us - (ulaw|opus|vp8|h264), peer - audio=(ulaw|opus)/video=(vp8)/text=(nothing), combined - (ulaw|opus|vp8)<br>Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)<br>Peer audio RTP is at port <a href="http://192.168.1.6:56634">192.168.1.6:56634</a><br>Peer video RTP is at port <a href="http://192.168.1.6:32953">192.168.1.6:32953</a><br>sip_route_dump: route/path hop: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;transport=wss><br>Transmitting (NAT) to <a href="http://192.168.1.6:48124">192.168.1.6:48124</a>:<br>ACK sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;transport=wss SIP/2.0<br>Via: SIP/2.0/WS 192.168.1.6:5060;branch=z9hG4bK65afda31;rport<br>Max-Forwards: 70<br>From: "Mason Kaufer 34" <<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as101db932<br>To: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;tag=HULiDWhvD78SNfAPBUqC<br>Contact: <sip:527@192.168.1.6:5060;transport=ws><br>Call-ID: <a href="http://4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060">4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060</a><br>CSeq: 102 ACK<br>User-Agent: Asterisk PBX 18.8.0<br>Content-Length: 0<br><br><br>---<br>    -- SIP/mason.kaufer.visualcampus-0000004b answered<br>    -- Executing [smvoice_callprogress@smvoice-dialout:1] GotoIf("SIP/mason.kaufer.visualcampus-0000004b", "0?smvoice-analog,s,1") in new stack<br>    -- Executing [smvoice_callprogress@smvoice-dialout:2] GotoIf("SIP/mason.kaufer.visualcampus-0000004b", "1?smvoice_callprogress,4:smvoice_callprogress,3") in new stack<br>    -- Goto (smvoice-dialout,smvoice_callprogress,4)<br>    -- Executing [smvoice_callprogress@smvoice-dialout:4] AGI("SIP/mason.kaufer.visualcampus-0000004b", "smvoice,-digium_asterisk") in new stack<br>    -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice<br>       > 0x7f8eac00b800 -- Strict RTP learning after ICE completion<br>       > 0x7f8eac0141f0 -- Strict RTP learning after ICE completion<br>       > 0x7f8eac00b800 -- Strict RTP learning after remote address set to: <a href="http://192.168.1.6:32953">192.168.1.6:32953</a><br>       > 0x7f8eac0141f0 -- Strict RTP learning after remote address set to: <a href="http://192.168.1.6:56634">192.168.1.6:56634</a><br>       > 0x7f8eac0141f0 -- Strict RTP switching to RTP target address <a href="http://192.168.1.6:56634">192.168.1.6:56634</a> as source<br>       > 0x7f8eac00b800 -- Strict RTP switching to RTP target address <a href="http://192.168.1.6:32953">192.168.1.6:32953</a> as source<br>    -- <SIP/mason.kaufer.visualcampus-0000004b>AGI Script smvoice completed, returning 0<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:1] Set("SIP/mason.kaufer.visualcampus-0000004b", "agi_use_meetme=0") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:2] Set("SIP/mason.kaufer.visualcampus-0000004b", "agi_use_confbridge=1") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:3] AGI("SIP/mason.kaufer.visualcampus-0000004b", "smvoice,-digium_success,-pa_list") in new stack<br>    -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice<br>    -- <SIP/mason.kaufer.visualcampus-0000004b>AGI Script smvoice completed, returning 0<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:4] Wait("SIP/mason.kaufer.visualcampus-0000004b", "1") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:5] NoOp("SIP/mason.kaufer.visualcampus-0000004b", "SETTING SPEAK LIVE GAIN") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:6] Set("SIP/mason.kaufer.visualcampus-0000004b", "VOLUME(TX)=0") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:7] Set("SIP/mason.kaufer.visualcampus-0000004b", "VOLUME(RX)=0") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:8] NoOp("SIP/mason.kaufer.visualcampus-0000004b", "START SPEAK LIVE OPTIONS") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:9] GotoIf("SIP/mason.kaufer.visualcampus-0000004b", "1?smvoice-local-public-address,app_confbridge_call_out,skip_speak_live_delay") in new stack<br>    -- Goto (smvoice-local-public-address,app_confbridge_call_out,11)<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:11] NoOp("SIP/mason.kaufer.visualcampus-0000004b", "Skipped speak live delay") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:12] GotoIf("SIP/mason.kaufer.visualcampus-0000004b", "1?smvoice-local-public-address,app_confbridge_call_out,skip_speak_live_preamble") in new stack<br>    -- Goto (smvoice-local-public-address,app_confbridge_call_out,18)<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:18] NoOp("SIP/mason.kaufer.visualcampus-0000004b", "Skipped speak live preamble") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:19] NoOp("SIP/mason.kaufer.visualcampus-0000004b", "END SPEAK LIVE OPTIONS") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:20] NoOp("SIP/mason.kaufer.visualcampus-0000004b", "START SPEAK LIVE BEEP") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:21] GotoIf("SIP/mason.kaufer.visualcampus-0000004b", "0?smvoice-local-public-address,app_confbridge_call_out,skip_speak_live_beep") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:22] Playback("SIP/mason.kaufer.visualcampus-0000004b", "beep") in new stack<br>    -- <SIP/mason.kaufer.visualcampus-0000004b> Playing 'beep.gsm' (language 'en')<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:23] NoOp("SIP/mason.kaufer.visualcampus-0000004b", "END SPEAK LIVE BEEP") in new stack<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:24] GotoIf("SIP/mason.kaufer.visualcampus-0000004b", "1?smvoice-local-public-address,app_confbridge_call_out,skip_record") in new stack<br>    -- Goto (smvoice-local-public-address,app_confbridge_call_out,26)<br>    -- Executing [app_confbridge_call_out@smvoice-local-public-address:26] ConfBridge("SIP/mason.kaufer.visualcampus-0000004b", "PA0003,LayeredSolutionsConfBridge,LayeredSolutionsConfUser") in new stack<br>    -- Channel CBAnn/PA0003-00000ae7;2 joined 'softmix' base-bridge <920968b7-3db6-4d15-ab7f-f123d585d98e><br>    -- Channel SIP/mason.kaufer.visualcampus-0000004b joined 'softmix' base-bridge <920968b7-3db6-4d15-ab7f-f123d585d98e><br>  == Using SIP VIDEO CoS mark 6<br>  == Using SIP RTP CoS mark 5<br>Audio is at 15542<br>Adding codec ulaw to SDP<br>Adding codec alaw to SDP<br>Adding codec gsm to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br>Reliably Transmitting (no NAT) to <a href="http://192.168.1.102:5063">192.168.1.102:5063</a>:<br>INVITE <a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK4fa302bf<br>Max-Forwards: 70<br>From: "Mason Kaufer 34" <<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as3729f342<br>To: <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>><br>Contact: <<a href="http://sip:527@192.168.1.6:5060">sip:527@192.168.1.6:5060</a>><br>Call-ID: <a href="http://240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060">240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 18.8.0<br>Date: Thu, 13 Jan 2022 13:46:21 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE<br>Supported: replaces, timer<br>Alert-Info: Ring Answer<br>Content-Type: application/sdp<br>Content-Length: 284<br><br>v=0<br>o=root 1569896537 1569896537 IN IP4 192.168.1.6<br>s=Asterisk PBX 18.8.0<br>c=IN IP4 192.168.1.6<br>t=0 0<br>m=audio 15542 RTP/AVP 0 8 3 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=maxptime:150<br>a=sendrecv<br><br>---<br>    -- Called 5124<br><br><--- SIP read from UDP:<a href="http://192.168.1.102:5063">192.168.1.102:5063</a> ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK4fa302bf<br>From: "Mason Kaufer 34" <<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as3729f342<br>To: <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>><br>Call-ID: <a href="http://240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060">240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060</a><br>CSeq: 102 INVITE<br>User-Agent: TOA N-SP80VS1 21.192.1.167 0005F9300106<br>Content-Length: 0<br><br><br><-------------><br>--- (8 headers 0 lines) ---<br><br><--- SIP read from UDP:<a href="http://192.168.1.102:5063">192.168.1.102:5063</a> ---><br>SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK4fa302bf<br>From: "Mason Kaufer 34" <<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as3729f342<br>To: <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>>;tag=735442138<br>Call-ID: <a href="http://240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060">240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060</a><br>CSeq: 102 INVITE<br>Contact: <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>><br>Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE<br>User-Agent: TOA N-SP80VS1 21.192.1.167 0005F9300106<br>Allow-Events: talk,hold,conference,refer,check-sync<br>P-Asserted-Identity: "5124"<<a href="mailto:sip%3A5124@192.168.1.6">sip:5124@192.168.1.6</a>><br>Privacy: none<br>Content-Length: 0<br><br><br><-------------><br>--- (13 headers 0 lines) ---<br>sip_route_dump: route/path hop: <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>><br>    -- SIP/5124-0000004c is ringing<br><br><--- SIP read from UDP:<a href="http://192.168.1.102:5063">192.168.1.102:5063</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK4fa302bf<br>From: "Mason Kaufer 34" <<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as3729f342<br>To: <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>>;tag=735442138<br>Call-ID: <a href="http://240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060">240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060</a><br>CSeq: 102 INVITE<br>Contact: <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>><br>Content-Type: application/sdp<br>Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE<br>User-Agent: TOA N-SP80VS1 21.192.1.167 0005F9300106<br>Content-Length:   210<br><br>v=0<br>o=5124 5000 5000 IN IP4 192.168.1.102<br>s=Talk<br>c=IN IP4 192.168.1.102<br>t=0 0<br>m=audio 11868 RTP/AVP 0 101<br>a=ptime:20<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=sendrecv<br><br><-------------><br>--- (11 headers 11 lines) ---<br>Got SDP version 5000 and unique parts [5124 5000 IN IP4 192.168.1.102]<br>Found RTP audio format 0<br>Found RTP audio format 101<br>Found audio description format PCMU for ID 0<br>Found audio description format telephone-event for ID 101<br>Capabilities: us - (h264|ulaw|alaw|gsm|vp8), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)<br>Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)<br>       > 0x7f8f400292c0 -- Strict RTP learning after remote address set to: <a href="http://192.168.1.102:11868">192.168.1.102:11868</a><br>Peer audio RTP is at port <a href="http://192.168.1.102:11868">192.168.1.102:11868</a><br>sip_route_dump: route/path hop: <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>><br>set_destination: Parsing <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>> for address/port to send to<br>set_destination: set destination to <a href="http://192.168.1.102:5063">192.168.1.102:5063</a><br>Transmitting (no NAT) to <a href="http://192.168.1.102:5063">192.168.1.102:5063</a>:<br>ACK <a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK18a38671<br>Max-Forwards: 70<br>From: "Mason Kaufer 34" <<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as3729f342<br>To: <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>>;tag=735442138<br>Contact: <<a href="http://sip:527@192.168.1.6:5060">sip:527@192.168.1.6:5060</a>><br>Call-ID: <a href="http://240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060">240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060</a><br>CSeq: 102 ACK<br>User-Agent: Asterisk PBX 18.8.0<br>Content-Length: 0<br><br><br>---<br>    -- SIP/5124-0000004c answered<br>    -- Executing [smvoice_pa_app_confbridge_twoway@smvoice-transfers:1] GotoIf("SIP/5124-0000004c", "1?skip_dtmf:use_dtmf") in new stack<br>    -- Goto (smvoice-transfers,smvoice_pa_app_confbridge_twoway,4)<br>    -- Executing [smvoice_pa_app_confbridge_twoway@smvoice-transfers:4] ConfBridge("SIP/5124-0000004c", "PA0003,LayeredSolutionsConfBridge,LayeredSolutionsConfUser") in new stack<br>    -- Channel SIP/5124-0000004c joined 'softmix' base-bridge <920968b7-3db6-4d15-ab7f-f123d585d98e><br>Reliably Transmitting (NAT) to <a href="http://192.168.1.6:48124">192.168.1.6:48124</a>:<br>OPTIONS sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss SIP/2.0<br>Via: SIP/2.0/WS 192.168.1.6:5060;branch=z9hG4bK3c0c1808;rport<br>Max-Forwards: 70<br>From: "asterisk" <<a href="mailto:sip%3Aasterisk@192.168.1.6">sip:asterisk@192.168.1.6</a>>;tag=as0ceb5b65<br>To: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss><br>Contact: <sip:asterisk@192.168.1.6:5060;transport=ws><br>Call-ID: <a href="http://3f760a323f6d68fa68274e1c6512bfa4@192.168.1.6:5060">3f760a323f6d68fa68274e1c6512bfa4@192.168.1.6:5060</a><br>CSeq: 102 OPTIONS<br>User-Agent: Asterisk PBX 18.8.0<br>Date: Thu, 13 Jan 2022 13:46:23 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE<br>Supported: replaces, timer<br>Content-Length: 0<br><br><br>---<br><br><--- SIP read from WS:<a href="http://192.168.1.6:48124">192.168.1.6:48124</a> ---><br>SIP/2.0 405 Method Not Allowed<br>Via: SIP/2.0/WS 192.168.1.6:5060;rport=5060;branch=z9hG4bK3c0c1808<br>From: "asterisk"<<a href="mailto:sip%3Aasterisk@192.168.1.6">sip:asterisk@192.168.1.6</a>>;tag=as0ceb5b65<br>To: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss><br>Call-ID: <a href="http://3f760a323f6d68fa68274e1c6512bfa4@192.168.1.6:5060">3f760a323f6d68fa68274e1c6512bfa4@192.168.1.6:5060</a><br>CSeq: 102 OPTIONS<br>Content-Length: 0<br><br><br><-------------><br>--- (7 headers 0 lines) ---<br>       > 0x7f8f400292c0 -- Strict RTP switching to RTP target address <a href="http://192.168.1.102:11868">192.168.1.102:11868</a> as source<br>       > 0x7f8eac00b800 -- Strict RTP learning complete - Locking on source address <a href="http://192.168.1.6:32953">192.168.1.6:32953</a><br>       > 0x7f8eac0141f0 -- Strict RTP learning complete - Locking on source address <a href="http://192.168.1.6:56634">192.168.1.6:56634</a><br>Really destroying SIP dialog '<a href="http://3f760a323f6d68fa68274e1c6512bfa4@192.168.1.6:5060">3f760a323f6d68fa68274e1c6512bfa4@192.168.1.6:5060</a>' Method: OPTIONS<br>       > 0x7f8f400292c0 -- Strict RTP learning complete - Locking on source address <a href="http://192.168.1.102:11868">192.168.1.102:11868</a><br><br><--- SIP read from WS:<a href="http://192.168.1.6:48124">192.168.1.6:48124</a> ---><br>BYE sip:527@192.168.1.6:5060;transport=ws SIP/2.0<br>Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKXZvId1AmTWItdTSFURnuLNJdEg8esTwa;rport<br>From: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid>;tag=HULiDWhvD78SNfAPBUqC<br>To: "Mason Kaufer 34"<<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as101db932<br>Call-ID: <a href="http://4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060">4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060</a><br>CSeq: 60034 BYE<br>Content-Length: 0<br>Max-Forwards: 70<br>User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04<br>Organization: Doubango Telecom<br><br><br><-------------><br>--- (11 headers 0 lines) ---<br>Scheduling destruction of SIP dialog '<a href="http://4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060">4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060</a>' in 6400 ms (Method: BYE)<br><br><--- Transmitting (NAT) to <a href="http://192.168.1.6:48124">192.168.1.6:48124</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKXZvId1AmTWItdTSFURnuLNJdEg8esTwa;received=192.168.1.6;rport=48124<br>From: <sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid>;tag=HULiDWhvD78SNfAPBUqC<br>To: "Mason Kaufer 34"<<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as101db932<br>Call-ID: <a href="http://4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060">4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060</a><br>CSeq: 60034 BYE<br>Server: Asterisk PBX 18.8.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE<br>Supported: replaces, timer<br>Content-Length: 0<br><br><br><------------><br>    -- Channel SIP/mason.kaufer.visualcampus-0000004b left 'softmix' base-bridge <920968b7-3db6-4d15-ab7f-f123d585d98e><br>    -- Executing [h@smvoice-local-public-address:1] NoOp("SIP/mason.kaufer.visualcampus-0000004b", "agi_pa_meetme=PA0003 agi_use_meetme0 agi_use_confbridge=1") in new stack<br>    -- Executing [h@smvoice-local-public-address:2] GotoIf("SIP/mason.kaufer.visualcampus-0000004b", "1?h_app_conference,1") in new stack<br>    -- Goto (smvoice-local-public-address,h_app_conference,1)<br>    -- Executing [h_app_conference@smvoice-local-public-address:1] GotoIf("SIP/mason.kaufer.visualcampus-0000004b", "1?h_app_confbridge,1") in new stack<br>    -- Goto (smvoice-local-public-address,h_app_confbridge,1)<br>    -- Executing [h_app_confbridge@smvoice-local-public-address:1] AGI("SIP/mason.kaufer.visualcampus-0000004b", "smvoice,-digium_success,-pa_done,241,PA0003") in new stack<br>    -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice<br>    -- Channel CBAnn/PA0003-00000ae7;2 left 'softmix' base-bridge <920968b7-3db6-4d15-ab7f-f123d585d98e><br>    -- Executing [h@smvoice-transfers:1] GotoIf("SIP/5124-0000004c", "1?h,s,4") in new stack<br>    -- Goto (h,s,4)<br>Scheduling destruction of SIP dialog '<a href="http://240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060">240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060</a>' in 32000 ms (Method: INVITE)<br>[Jan 13 08:46:28] NOTICE[1097928]: manager.c:4499 action_hangup: Request to hangup non-existent channel: SIP/5124-0000004c<br>set_destination: Parsing <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>> for address/port to send to<br>set_destination: set destination to <a href="http://192.168.1.102:5063">192.168.1.102:5063</a><br>Reliably Transmitting (no NAT) to <a href="http://192.168.1.102:5063">192.168.1.102:5063</a>:<br>BYE <a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK6e2de9fa<br>Max-Forwards: 70<br>From: "Mason Kaufer 34" <<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as3729f342<br>To: <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>>;tag=735442138<br>Call-ID: <a href="http://240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060">240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060</a><br>CSeq: 103 BYE<br>User-Agent: Asterisk PBX 18.8.0<br>X-Asterisk-HangupCause: Normal Clearing<br>X-Asterisk-HangupCauseCode: 16<br>Content-Length: 0<br><br><br>---<br><br><--- SIP read from UDP:<a href="http://192.168.1.102:5063">192.168.1.102:5063</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK6e2de9fa<br>From: "Mason Kaufer 34" <<a href="mailto:sip%3A527@192.168.1.6">sip:527@192.168.1.6</a>>;tag=as3729f342<br>To: <<a href="http://sip:5124@192.168.1.102:5063">sip:5124@192.168.1.102:5063</a>>;tag=735442138<br>Call-ID: <a href="http://240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060">240dde9e3a94137916b97b2d60264c53@192.168.1.6:5060</a><br>CSeq: 103 BYE<br>User-Agent: TOA N-SP80VS1 21.192.1.167 0005F9300106<br>Content-Length: 0<br></div><div> </div></div></div>