[Sep 29 14:26:27] Asterisk 15.4.0 built by root @ testbed-sw-1 on a x86_64 running Linux on 2018-06-17 10:37:09 UTC [Sep 29 14:26:27] VERBOSE[62459] logger.c: Asterisk Queue Logger restarted [Sep 29 14:26:41] VERBOSE[61987] chan_sip.c: <--- SIP read from UDP:172.16.0.14:5060 ---> INVITE sip:942020@testbedpr.test.info SIP/2.0 Record-Route: Record-Route: Record-Route: Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK633c.159e5cec575c28872874512ed8e71eb9.0 Via: SIP/2.0/UDP 172.16.0.29;branch=z9hG4bK633c.0233ba3d3e96bc13a8584b98fccc16c2.0 Via: SIP/2.0/UDP 10.0.5.4:1182;received=10.0.5.22;branch=z9hG4bK0767183f;rport=1026 Max-Forwards: 68 From: ;tag=as696e0599 To: Contact: Call-ID: 40388a5b09bef1175646d1fd040158fa@testbed CSeq: 103 INVITE User-Agent: Asterisk PBX 14.4.0 Date: Sat, 29 Sep 2018 10:56:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 312 v=0 o=root 1093938844 1093938845 IN IP4 172.16.0.29 s=testSW c=IN IP4 172.16.0.29 t=0 0 m=audio 31828 RTP/AVP 0 8 18 101 a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-16 a=sendrecv a=rtcp:31829 <-------------> [Sep 29 14:26:41] VERBOSE[61987] chan_sip.c: --- (19 headers 15 lines) --- [Sep 29 14:26:41] VERBOSE[61987] chan_sip.c: Sending to 172.16.0.14:5060 (NAT) [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Sending to 172.16.0.14:5060 (NAT) [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Using INVITE request as basis request - 40388a5b09bef1175646d1fd040158fa@testbed [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '69b1c1fb216ade5c337bf1f03960c9d4@testbed' in 32000 ms (Method: NOTIFY) [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Reliably Transmitting (NAT) to 172.16.0.14:5060: NOTIFY sip:949999@172.16.0.14:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.15:5060;branch=z9hG4bK1aa9185c;rport Max-Forwards: 70 From: "asterisk" ;tag=as1d0e607e To: Contact: Call-ID: 69b1c1fb216ade5c337bf1f03960c9d4@testbed CSeq: 102 NOTIFY User-Agent: test SW Event: message-summary Content-Type: application/simple-message-summary Content-Length: 87 Messages-Waiting: no Message-Account: sip:asterisk@testbed Voice-Message: 0/0 (0/0) --- [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Found peer '949999' for '949999' from 172.16.0.14:5060 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] netsock2.c: Using SIP VIDEO CoS mark 6 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] netsock2.c: Using SIP RTP CoS mark 5 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Found RTP audio format 0 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Found RTP audio format 8 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Found RTP audio format 18 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Found RTP audio format 101 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Found audio description format G729 for ID 18 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Capabilities: us - (ulaw|alaw|opus|vp8|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729) [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Sep 29 14:26:41] VERBOSE[61987][C-00000001] res_rtp_asterisk.c: 0x7f156c01b1a0 -- Strict RTP learning after remote address set to: 172.16.0.29:31828 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Peer audio RTP is at port 172.16.0.29:31828 [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Peer doesn't provide video [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: Looking for 942020 in public (domain testbedpr.test.info) [Sep 29 14:26:41] VERBOSE[61987][C-00000001] sip/route.c: sip_route_dump: route/path hop: [Sep 29 14:26:41] VERBOSE[61987][C-00000001] sip/route.c: sip_route_dump: route/path hop: [Sep 29 14:26:41] VERBOSE[61987][C-00000001] sip/route.c: sip_route_dump: route/path hop: [Sep 29 14:26:41] VERBOSE[61987][C-00000001] chan_sip.c: <--- Transmitting (NAT) to 172.16.0.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK633c.159e5cec575c28872874512ed8e71eb9.0;received=172.16.0.14;rport=5060 Via: SIP/2.0/UDP 172.16.0.29;branch=z9hG4bK633c.0233ba3d3e96bc13a8584b98fccc16c2.0 Via: SIP/2.0/UDP 10.0.5.4:1182;received=10.0.5.22;branch=z9hG4bK0767183f;rport=1026 Record-Route: Record-Route: Record-Route: From: ;tag=as696e0599 To: Call-ID: 40388a5b09bef1175646d1fd040158fa@testbed CSeq: 103 INVITE Server: test SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Sep 29 14:26:41] VERBOSE[62460][C-00000001] pbx.c: Executing [942020@public:1] NoOp("SIP/949999-00000000", "") in new stack [Sep 29 14:26:41] VERBOSE[62460][C-00000001] pbx.c: Executing [942020@public:2] Stasis("SIP/949999-00000000", "switchmanager,incoming,942020,unknown,unknown,none") in new stack [Sep 29 14:26:41] VERBOSE[61987] chan_sip.c: <--- SIP read from UDP:172.16.0.14:5060 ---> SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 172.16.0.15:5060;received=172.16.0.15;branch=z9hG4bK1aa9185c;rport=5060 From: "asterisk" ;tag=as1d0e607e To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.3cb2 Call-ID: 69b1c1fb216ade5c337bf1f03960c9d4@testbed CSeq: 102 NOTIFY Server: kamailio (4.4.5 (x86_64/linux)) Content-Length: 0 <-------------> [Sep 29 14:26:41] VERBOSE[61987] chan_sip.c: --- (8 headers 0 lines) --- [Sep 29 14:26:41] VERBOSE[62463] netsock2.c: Using SIP VIDEO CoS mark 6 [Sep 29 14:26:41] VERBOSE[62463] netsock2.c: Using SIP RTP CoS mark 5 [Sep 29 14:26:41] VERBOSE[62464] chan_sip.c: Audio is at 19222 [Sep 29 14:26:41] VERBOSE[62464] chan_sip.c: Adding codec ulaw to SDP [Sep 29 14:26:41] VERBOSE[62464] chan_sip.c: Adding codec alaw to SDP [Sep 29 14:26:41] VERBOSE[62464] chan_sip.c: Adding codec opus to SDP [Sep 29 14:26:41] VERBOSE[62464] chan_sip.c: Adding codec g729 to SDP [Sep 29 14:26:41] VERBOSE[62464] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Sep 29 14:26:41] VERBOSE[62464] chan_sip.c: Reliably Transmitting (NAT) to 172.16.0.14:5060: INVITE sip:98942020@lb-host SIP/2.0 Via: SIP/2.0/UDP 172.16.0.15:5060;branch=z9hG4bK53561f2b;rport Max-Forwards: 70 From: ;tag=as0b78761e To: Contact: Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 102 INVITE User-Agent: test SW Date: Sat, 29 Sep 2018 10:56:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 326 v=0 o=root 543991548 543991548 IN IP4 172.16.0.15 s=testSW c=IN IP4 172.16.0.15 t=0 0 m=audio 19222 RTP/AVP 0 8 107 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:107 opus/48000/2 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- [Sep 29 14:26:41] VERBOSE[62464] dial.c: Called 98942020@LB [Sep 29 14:26:41] VERBOSE[61987] chan_sip.c: <--- SIP read from UDP:172.16.0.14:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 172.16.0.15:5060;branch=z9hG4bK53561f2b;rport=5060;received=172.16.0.15 From: ;tag=as0b78761e To: Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 102 INVITE Server: kamailio (4.4.5 (x86_64/linux)) Content-Length: 0 <-------------> [Sep 29 14:26:41] VERBOSE[61987] chan_sip.c: --- (8 headers 0 lines) --- [Sep 29 14:26:41] VERBOSE[62460][C-00000001] chan_sip.c: <--- Transmitting (NAT) to 172.16.0.14:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK633c.159e5cec575c28872874512ed8e71eb9.0;received=172.16.0.14;rport=5060 Via: SIP/2.0/UDP 172.16.0.29;branch=z9hG4bK633c.0233ba3d3e96bc13a8584b98fccc16c2.0 Via: SIP/2.0/UDP 10.0.5.4:1182;received=10.0.5.22;branch=z9hG4bK0767183f;rport=1026 Record-Route: Record-Route: Record-Route: From: ;tag=as696e0599 To: ;tag=as28d6dd4a Call-ID: 40388a5b09bef1175646d1fd040158fa@testbed CSeq: 103 INVITE Server: test SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: <--- SIP read from UDP:172.16.0.14:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.0.15:5060;received=172.16.0.15;branch=z9hG4bK53561f2b;rport=5060 Record-Route: From: ;tag=as0b78761e To: ;tag=as4d44d9f8 Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 102 INVITE Server: Asterisk PBX 14.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 340 v=0 o=root 1510249861 1510249861 IN IP4 192.168.100.10 s=Asterisk PBX 14.6.0 c=IN IP4 192.168.100.10 t=0 0 m=audio 16176 RTP/AVP 0 8 107 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:107 opus/48000/2 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv <-------------> [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: --- (15 headers 15 lines) --- [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Found RTP audio format 0 [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Found RTP audio format 8 [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Found RTP audio format 107 [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Found RTP audio format 18 [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Found RTP audio format 101 [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Found audio description format PCMU for ID 0 [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Found audio description format PCMA for ID 8 [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Found audio description format opus for ID 107 [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Found audio description format G729 for ID 18 [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Found audio description format telephone-event for ID 101 [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Capabilities: us - (ulaw|alaw|opus|vp8|g729), peer - audio=(ulaw|alaw|g729|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|opus|g729) [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Sep 29 14:26:42] VERBOSE[61987] res_rtp_asterisk.c: 0x7f1574006340 -- Strict RTP learning after remote address set to: 192.168.100.10:16176 [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Peer audio RTP is at port 192.168.100.10:16176 [Sep 29 14:26:42] VERBOSE[61987] sip/route.c: sip_route_dump: route/path hop: [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: Transmitting (NAT) to 172.16.0.14:5060: ACK sip:98942020@192.168.100.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.15:5060;branch=z9hG4bK5f5098ae;rport Route: Max-Forwards: 70 From: ;tag=as0b78761e To: ;tag=as4d44d9f8 Contact: Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 102 ACK User-Agent: test SW Content-Length: 0 --- [Sep 29 14:26:42] VERBOSE[62464] dial.c: SIP/LB-00000001 answered [Sep 29 14:26:42] VERBOSE[62464] ari/resource_channels.c: Launching Stasis(switchmanager,outbound) on SIP/LB-00000001 [Sep 29 14:26:42] VERBOSE[62460][C-00000001] chan_sip.c: Audio is at 13714 [Sep 29 14:26:42] VERBOSE[62460][C-00000001] chan_sip.c: Adding codec ulaw to SDP [Sep 29 14:26:42] VERBOSE[62460][C-00000001] chan_sip.c: Adding codec alaw to SDP [Sep 29 14:26:42] VERBOSE[62460][C-00000001] chan_sip.c: Adding codec g729 to SDP [Sep 29 14:26:42] VERBOSE[62460][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Sep 29 14:26:42] VERBOSE[62460][C-00000001] chan_sip.c: <--- Reliably Transmitting (NAT) to 172.16.0.14:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK633c.159e5cec575c28872874512ed8e71eb9.0;received=172.16.0.14;rport=5060 Via: SIP/2.0/UDP 172.16.0.29;branch=z9hG4bK633c.0233ba3d3e96bc13a8584b98fccc16c2.0 Via: SIP/2.0/UDP 10.0.5.4:1182;received=10.0.5.22;branch=z9hG4bK0767183f;rport=1026 Record-Route: Record-Route: Record-Route: From: ;tag=as696e0599 To: ;tag=as28d6dd4a Call-ID: 40388a5b09bef1175646d1fd040158fa@testbed CSeq: 103 INVITE Server: test SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 298 v=0 o=root 1255155792 1255155792 IN IP4 172.16.0.15 s=testSW c=IN IP4 172.16.0.15 t=0 0 m=audio 13714 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: <--- SIP read from UDP:172.16.0.14:5060 ---> ACK sip:942020@172.16.0.15:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK633c.6f49f3bc2f0044c5a4aad7fd95a0950a.0 Via: SIP/2.0/UDP 172.16.0.29;branch=z9hG4bK633c.d9b52d452fc5a03e4b69249472cc6b11.0 Via: SIP/2.0/UDP 10.0.5.4:1182;received=10.0.5.22;branch=z9hG4bK20d0d3a9;rport=1026 Max-Forwards: 68 From: ;tag=as696e0599 To: ;tag=as28d6dd4a Contact: Call-ID: 40388a5b09bef1175646d1fd040158fa@testbed CSeq: 103 ACK User-Agent: Asterisk PBX 14.4.0 Content-Length: 0 <-------------> [Sep 29 14:26:42] VERBOSE[61987] chan_sip.c: --- (12 headers 0 lines) --- [Sep 29 14:26:42] VERBOSE[62466][C-00000001] bridge_channel.c: Channel SIP/949999-00000000 joined 'simple_bridge' stasis-bridge <5f1cc272-3814-4b82-8826-59fd8c7ae5ce> [Sep 29 14:26:42] VERBOSE[62467] bridge_channel.c: Channel SIP/LB-00000001 joined 'simple_bridge' stasis-bridge <5f1cc272-3814-4b82-8826-59fd8c7ae5ce> [Sep 29 14:26:42] VERBOSE[62470] bridge_channel.c: Channel Recorder/ARI-00000000;2 joined 'simple_bridge' stasis-bridge <5f1cc272-3814-4b82-8826-59fd8c7ae5ce> [Sep 29 14:26:42] VERBOSE[62470] bridge.c: Bridge 5f1cc272-3814-4b82-8826-59fd8c7ae5ce: switching from simple_bridge technology to softmix [Sep 29 14:26:42] VERBOSE[62472][C-00000001] app.c: x=0, open writing: /var/spool/asterisk/recording/cdr/188/949999_LB_1538218602_757882 format: wav, 0x7f1554001bd8 [Sep 29 14:26:42] VERBOSE[62467] res_rtp_asterisk.c: 0x7f1574006340 -- Strict RTP switching to RTP target address 192.168.100.10:16176 as source [Sep 29 14:26:45] VERBOSE[62466][C-00000001] res_rtp_asterisk.c: 0x7f156c01b1a0 -- Strict RTP switching to RTP target address 172.16.0.29:31828 as source [Sep 29 14:26:45] WARNING[62466][C-00000001] chan_iax2.c: Resyncing the jb. last_delay 0, this delay 2810, threshold 1000, new offset -2810 [Sep 29 14:26:45] VERBOSE[61987] chan_sip.c: <--- SIP read from UDP:172.16.0.14:5060 ---> INVITE sip:949999@172.16.0.15:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK81d4.77dfb82ec4b4f9786a25bf7bb07d8ed8.0 Via: SIP/2.0/UDP 192.168.100.10:5060;received=192.168.100.10;branch=z9hG4bK33f54e3b;rport=5060 Max-Forwards: 69 From: ;tag=as4d44d9f8 To: ;tag=as0b78761e Contact: Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.6.0 Session-Expires: 1800;refresher=uac Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 271 v=0 o=root 1510249861 1510249862 IN IP4 192.168.100.10 s=Asterisk PBX 14.6.0 c=IN IP4 192.168.100.10 t=0 0 m=image 4978 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Sep 29 14:26:45] VERBOSE[61987] chan_sip.c: --- (16 headers 11 lines) --- [Sep 29 14:26:45] VERBOSE[61987] chan_sip.c: Sending to 172.16.0.14:5060 (NAT) [Sep 29 14:26:45] VERBOSE[61987] netsock2.c: Using UDPTL CoS mark 5 [Sep 29 14:26:45] VERBOSE[61987] chan_sip.c: Got T.38 offer in SDP in dialog 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 [Sep 29 14:26:45] VERBOSE[61987] chan_sip.c: Capabilities: us - (ulaw|alaw|opus|vp8|g729), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing) [Sep 29 14:26:45] VERBOSE[61987] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) [Sep 29 14:26:45] VERBOSE[61987] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. [Sep 29 14:26:45] VERBOSE[61987] chan_sip.c: <--- Transmitting (NAT) to 172.16.0.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK81d4.77dfb82ec4b4f9786a25bf7bb07d8ed8.0;received=172.16.0.14;rport=5060 Via: SIP/2.0/UDP 192.168.100.10:5060;received=192.168.100.10;branch=z9hG4bK33f54e3b;rport=5060 From: ;tag=as4d44d9f8 To: ;tag=as0b78761e Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 102 INVITE Server: test SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: Content-Length: 0 <------------> [Sep 29 14:26:46] VERBOSE[62466][C-00000001] res_rtp_asterisk.c: 0x7f156c01b1a0 -- Strict RTP learning complete - Locking on source address 172.16.0.29:31828 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: <--- Reliably Transmitting (NAT) to 172.16.0.14:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK81d4.77dfb82ec4b4f9786a25bf7bb07d8ed8.0;received=172.16.0.14;rport=5060 Via: SIP/2.0/UDP 192.168.100.10:5060;received=192.168.100.10;branch=z9hG4bK33f54e3b;rport=5060 From: ;tag=as4d44d9f8 To: ;tag=as0b78761e Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 102 INVITE Server: test SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Reason: Q.850;cause=16 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <------------> [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: <--- SIP read from UDP:172.16.0.14:5060 ---> ACK sip:949999@172.16.0.15:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK81d4.77dfb82ec4b4f9786a25bf7bb07d8ed8.0 Max-Forwards: 69 From: ;tag=as4d44d9f8 To: ;tag=as0b78761e Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 102 ACK Content-Length: 0 <-------------> [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: --- (8 headers 0 lines) --- [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: <--- SIP read from UDP:172.16.0.14:5060 ---> INVITE sip:949999@172.16.0.15:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK91d4.d8c20932ebd451b062311ef175a2d993.0 Via: SIP/2.0/UDP 192.168.100.10:5060;received=192.168.100.10;branch=z9hG4bK1204b57f;rport=5060 Max-Forwards: 69 From: ;tag=as4d44d9f8 To: ;tag=as0b78761e Contact: Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 14.6.0 Session-Expires: 1800;refresher=uac Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 340 v=0 o=root 1510249861 1510249863 IN IP4 192.168.100.10 s=Asterisk PBX 14.6.0 c=IN IP4 192.168.100.10 t=0 0 m=audio 16176 RTP/AVP 0 8 107 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:107 opus/48000/2 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv <-------------> [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: --- (16 headers 15 lines) --- [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Sending to 172.16.0.14:5060 (NAT) [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Found RTP audio format 0 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Found RTP audio format 8 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Found RTP audio format 107 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Found RTP audio format 18 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Found RTP audio format 101 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Found audio description format PCMU for ID 0 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Found audio description format PCMA for ID 8 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Found audio description format opus for ID 107 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Found audio description format G729 for ID 18 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Found audio description format telephone-event for ID 101 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Capabilities: us - (ulaw|alaw|opus|vp8|g729), peer - audio=(ulaw|alaw|g729|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|opus|g729) [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Peer audio RTP is at port 192.168.100.10:16176 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: <--- Transmitting (NAT) to 172.16.0.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK91d4.d8c20932ebd451b062311ef175a2d993.0;received=172.16.0.14;rport=5060 Via: SIP/2.0/UDP 192.168.100.10:5060;received=192.168.100.10;branch=z9hG4bK1204b57f;rport=5060 From: ;tag=as4d44d9f8 To: ;tag=as0b78761e Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 103 INVITE Server: test SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: Content-Length: 0 <------------> [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Audio is at 19222 [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Adding codec ulaw to SDP [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Adding codec alaw to SDP [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Adding codec opus to SDP [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Adding codec g729 to SDP [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: <--- Reliably Transmitting (NAT) to 172.16.0.14:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK91d4.d8c20932ebd451b062311ef175a2d993.0;received=172.16.0.14;rport=5060 Via: SIP/2.0/UDP 192.168.100.10:5060;received=192.168.100.10;branch=z9hG4bK1204b57f;rport=5060 From: ;tag=as4d44d9f8 To: ;tag=as0b78761e Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 103 INVITE Server: test SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: Content-Type: application/sdp Require: timer Content-Length: 326 v=0 o=root 543991548 543991549 IN IP4 172.16.0.15 s=testSW c=IN IP4 172.16.0.15 t=0 0 m=audio 19222 RTP/AVP 0 8 107 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:107 opus/48000/2 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv <------------> [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: <--- SIP read from UDP:172.16.0.14:5060 ---> ACK sip:949999@172.16.0.15:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK91d4.f2716bf64149861d88d1239f7bd0b11e.0 Via: SIP/2.0/UDP 192.168.100.10:5060;received=192.168.100.10;branch=z9hG4bK45593dda;rport=5060 Max-Forwards: 69 From: ;tag=as4d44d9f8 To: ;tag=as0b78761e Contact: Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 14.6.0 Content-Length: 0 <-------------> [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: --- (11 headers 0 lines) --- [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: <--- SIP read from UDP:172.16.0.14:5060 ---> BYE sip:949999@172.16.0.15:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK61d4.458382c7b3d99a32edef4845c37196cc.0 Via: SIP/2.0/UDP 192.168.100.10:5060;received=192.168.100.10;branch=z9hG4bK12ce4685;rport=5060 Max-Forwards: 69 From: ;tag=as4d44d9f8 To: ;tag=as0b78761e Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 14.6.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: --- (12 headers 0 lines) --- [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Sending to 172.16.0.14:5060 (NAT) [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Scheduling destruction of SIP dialog '4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060' in 32000 ms (Method: BYE) [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: <--- Transmitting (NAT) to 172.16.0.14:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.0.14;branch=z9hG4bK61d4.458382c7b3d99a32edef4845c37196cc.0;received=172.16.0.14;rport=5060 Via: SIP/2.0/UDP 192.168.100.10:5060;received=192.168.100.10;branch=z9hG4bK12ce4685;rport=5060 From: ;tag=as4d44d9f8 To: ;tag=as0b78761e Call-ID: 4ee3ce0e2fa8b40b3dd8ba7b5eb9064e@172.16.0.15:5060 CSeq: 104 BYE Server: test SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Sep 29 14:26:50] VERBOSE[62467] bridge_channel.c: Channel SIP/LB-00000001 left 'softmix' stasis-bridge <5f1cc272-3814-4b82-8826-59fd8c7ae5ce> [Sep 29 14:26:50] VERBOSE[62467] bridge.c: Bridge 5f1cc272-3814-4b82-8826-59fd8c7ae5ce: switching from softmix technology to simple_bridge [Sep 29 14:26:50] VERBOSE[62472][C-00000001] app.c: Message ended by control [Sep 29 14:26:50] VERBOSE[62466][C-00000001] bridge_channel.c: Channel SIP/949999-00000000 left 'simple_bridge' stasis-bridge <5f1cc272-3814-4b82-8826-59fd8c7ae5ce> [Sep 29 14:26:50] VERBOSE[62470] bridge_channel.c: Channel Recorder/ARI-00000000;2 left 'simple_bridge' stasis-bridge <5f1cc272-3814-4b82-8826-59fd8c7ae5ce> [Sep 29 14:26:50] VERBOSE[62460][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '40388a5b09bef1175646d1fd040158fa@testbed' in 32000 ms (Method: ACK) [Sep 29 14:26:50] VERBOSE[62460][C-00000001] chan_sip.c: Reliably Transmitting (NAT) to 172.16.0.14:5060: BYE sip:949999@10.0.5.4:1182;alias=10.0.5.22~1026~1 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.15:5060;branch=z9hG4bK6a380143;rport Route: ,, Max-Forwards: 70 From: ;tag=as28d6dd4a To: ;tag=as696e0599 Call-ID: 40388a5b09bef1175646d1fd040158fa@testbed CSeq: 102 BYE User-Agent: test SW Reason: Q.850;cause=16 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: <--- SIP read from UDP:172.16.0.14:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.0.15:5060;received=172.16.0.15;branch=z9hG4bK6a380143;rport=5060 From: ;tag=as28d6dd4a To: ;tag=as696e0599 Call-ID: 40388a5b09bef1175646d1fd040158fa@testbed CSeq: 102 BYE Server: Asterisk PBX 14.4.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: --- (10 headers 0 lines) --- [Sep 29 14:26:50] VERBOSE[61987][C-00000001] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Sep 29 14:26:50] VERBOSE[61987] chan_sip.c: Really destroying SIP dialog '40388a5b09bef1175646d1fd040158fa@testbed' Method: ACK [Sep 29 14:26:56] VERBOSE[62459] asterisk.c: Remote UNIX connection disconnected [Sep 29 14:27:12] VERBOSE[61932] asterisk.c: Remote UNIX connection [Sep 29 14:27:13] VERBOSE[61987] chan_sip.c: Really destroying SIP dialog '69b1c1fb216ade5c337bf1f03960c9d4@testbed' Method: NOTIFY [Sep 29 14:27:15] VERBOSE[61987] chan_sip.c: Reloading SIP