-- Executing [01476292507@testdial:1] Dial("PJSIP/squiresvi-000000ef", "PJSIP/custom/sip:901@pbx.example.com,,b(testaddhandler^set_handler^1)") in new stack -- PJSIP/custom-000000f0 Internal Gosub(testaddhandler,set_handler,1) start -- Executing [set_handler@testaddhandler:1] NoOp("PJSIP/custom-000000f0", "Setting handler") in new stack -- Executing [set_handler@testaddhandler:2] Set("PJSIP/custom-000000f0", "CHANNEL(hangup_handler_push)=testhandler,outbound_handler,1") in new stack -- Executing [set_handler@testaddhandler:3] Return("PJSIP/custom-000000f0", "") in new stack == Spawn extension (default, 01476292507, 1) exited non-zero on 'PJSIP/custom-000000f0' -- PJSIP/custom-000000f0 Internal Gosub(testaddhandler,set_handler,1) complete GOSUB_RETVAL= -- Called PJSIP/custom/sip:901@pbx.example.com -- PJSIP/custom-000000f0 is ringing (callee answers) [Jun 13 19:46:06] NOTICE[6197]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'OPTIONS' from '"asterisk" ' failed for '10.20.40.164:5060' (callid: 46c6ccfc696d70ee596220c3367be0cb@10.20.40.164) - No matching endpoint found > 0x7fdc780269a0 -- Strict RTP learning after remote address set to: 10.20.40.164:11050 -- PJSIP/custom-000000f0 answered PJSIP/squiresvi-000000ef > 0x7fdc7801e0e0 -- Strict RTP learning after remote address set to: 10.20.41.37:4000 -- Channel PJSIP/custom-000000f0 joined 'simple_bridge' basic-bridge <736affd8-a001-48d7-a75b-1442b0b746e0> -- Channel PJSIP/squiresvi-000000ef joined 'simple_bridge' basic-bridge <736affd8-a001-48d7-a75b-1442b0b746e0> > Bridge 736affd8-a001-48d7-a75b-1442b0b746e0: switching from simple_bridge technology to native_rtp > Remotely bridged 'PJSIP/squiresvi-000000ef' and 'PJSIP/custom-000000f0' - media will flow directly between them > 0x7fdc780269a0 -- Strict RTP learning after remote address set to: 10.20.40.164:11050 > 0x7fdc7801e0e0 -- Strict RTP switching to RTP target address 10.20.41.37:4000 as source (callee hangs up) -- Channel PJSIP/custom-000000f0 left 'native_rtp' basic-bridge <736affd8-a001-48d7-a75b-1442b0b746e0> -- Channel PJSIP/squiresvi-000000ef left 'native_rtp' basic-bridge <736affd8-a001-48d7-a75b-1442b0b746e0> -- PJSIP/custom-000000f0 Internal Gosub(testhandler,outbound_handler,1) start -- Executing [outbound_handler@testhandler:1] NoOp("PJSIP/custom-000000f0", "## Hangup handler") in new stack -- Executing [outbound_handler@testhandler:2] NoOp("PJSIP/custom-000000f0", "## CHANNEL=PJSIP/custom-000000f0") in new stack == Spawn extension (testdial, 01476292507, 1) exited non-zero on 'PJSIP/squiresvi-000000ef' -- Executing [outbound_handler@testhandler:3] NoOp("PJSIP/custom-000000f0", "## HANGUPCAUSE_KEYS=PJSIP/custom-000000f0,PJSIP/squiresvi-000000ef") in new stack -- Executing [outbound_handler@testhandler:4] NoOp("PJSIP/custom-000000f0", "## HANGUPCAUSE=SIP 200 OK") in new stack -- Executing [outbound_handler@testhandler:5] Return("PJSIP/custom-000000f0", "") in new stack == Spawn extension (default, , 1) exited non-zero on 'PJSIP/custom-000000f0' -- PJSIP/custom-000000f0 Internal Gosub(testhandler,outbound_handler,1) complete GOSUB_RETVAL= U 10.20.41.27:5060 -> 10.20.40.164:5060 #382 INVITE sip:901@pbx.example.com SIP/2.0. Via: SIP/2.0/UDP 10.20.41.27:5060;rport;branch=z9hG4bKPj99026f29-4eb9-4905-9c55-0462659980d2. From: " " ;tag=3ab0831a-af3e-43e9-a4d5-0cfefe87efd6. To: . Contact: . Call-ID: a0197842-bd9f-4cf2-973e-a542bb64da09. CSeq: 22718 INVITE. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE. Supported: 100rel, timer, replaces, norefersub. Session-Expires: 1800. Min-SE: 90. Max-Forwards: 70. User-Agent: Asterisk PBX certified/13.21-cert2. Content-Type: application/sdp. Content-Length: 396. . v=0. o=- 1937807452 1937807452 IN IP4 10.20.41.27. s=Asterisk. c=IN IP4 10.20.41.27. t=0 0. m=audio 13510 RTP/AVP 8 0 3 9 110 117 119 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:9 G722/8000. a=rtpmap:110 speex/8000. a=rtpmap:117 speex/16000. a=rtpmap:119 speex/32000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:60. a=sendrecv. U 10.20.40.164:5060 -> 10.20.41.27:5060 #383 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.20.41.27:5060;branch=z9hG4bKPj99026f29-4eb9-4905-9c55-0462659980d2;received=10.20.41.27;rport=5060. From: " " ;tag=3ab0831a-af3e-43e9-a4d5-0cfefe87efd6. To: . Call-ID: a0197842-bd9f-4cf2-973e-a542bb64da09. CSeq: 22718 INVITE. Server: Asterisk PBX 1.6.2.6. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: . Content-Length: 0. . U 10.20.40.164:5060 -> 10.20.41.27:5060 #388 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.20.41.27:5060;branch=z9hG4bKPj99026f29-4eb9-4905-9c55-0462659980d2;received=10.20.41.27;rport=5060. From: " " ;tag=3ab0831a-af3e-43e9-a4d5-0cfefe87efd6. To: ;tag=as03777070. Call-ID: a0197842-bd9f-4cf2-973e-a542bb64da09. CSeq: 22718 INVITE. Server: Asterisk PBX 1.6.2.6. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: . Content-Length: 0. . U 10.20.40.164:5060 -> 10.20.41.27:5060 #401 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.20.41.27:5060;branch=z9hG4bKPj99026f29-4eb9-4905-9c55-0462659980d2;received=10.20.41.27;rport=5060. From: " " ;tag=3ab0831a-af3e-43e9-a4d5-0cfefe87efd6. To: ;tag=as03777070. Call-ID: a0197842-bd9f-4cf2-973e-a542bb64da09. CSeq: 22718 INVITE. Server: Asterisk PBX 1.6.2.6. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: . Content-Type: application/sdp. Content-Length: 312. . v=0. o=root 1547026156 1547026156 IN IP4 10.20.40.164. s=Asterisk PBX 1.6.2.6. c=IN IP4 10.20.40.164. t=0 0. m=audio 11050 RTP/AVP 3 0 8 101. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 10.20.41.27:5060 -> 10.20.40.164:5060 #402 ACK sip:901@10.20.40.164 SIP/2.0. Via: SIP/2.0/UDP 10.20.41.27:5060;rport;branch=z9hG4bKPj8be1abda-3b54-47c5-86e8-6a845546b477. From: " " ;tag=3ab0831a-af3e-43e9-a4d5-0cfefe87efd6. To: ;tag=as03777070. Call-ID: a0197842-bd9f-4cf2-973e-a542bb64da09. CSeq: 22718 ACK. Max-Forwards: 70. User-Agent: Asterisk PBX certified/13.21-cert2. Content-Length: 0. . U 10.20.41.27:5060 -> 10.20.40.164:5060 #404 INVITE sip:901@10.20.40.164 SIP/2.0. Via: SIP/2.0/UDP 10.20.41.27:5060;rport;branch=z9hG4bKPj2e8abe87-c676-4209-b8a1-2b975cbaf85b. From: " " ;tag=3ab0831a-af3e-43e9-a4d5-0cfefe87efd6. To: ;tag=as03777070. Contact: . Call-ID: a0197842-bd9f-4cf2-973e-a542bb64da09. CSeq: 22719 INVITE. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE. Supported: 100rel, timer, replaces, norefersub. Session-Expires: 1800;refresher=uas. Min-SE: 90. Max-Forwards: 70. User-Agent: Asterisk PBX certified/13.21-cert2. Content-Type: application/sdp. Content-Length: 236. . v=0. o=- 1937807452 1937807453 IN IP4 10.20.41.27. s=Asterisk. c=IN IP4 10.20.41.37. t=0 0. m=audio 4000 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. U 10.20.40.164:5060 -> 10.20.41.27:5060 #405 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.20.41.27:5060;branch=z9hG4bKPj2e8abe87-c676-4209-b8a1-2b975cbaf85b;received=10.20.41.27;rport=5060. From: " " ;tag=3ab0831a-af3e-43e9-a4d5-0cfefe87efd6. To: ;tag=as03777070. Call-ID: a0197842-bd9f-4cf2-973e-a542bb64da09. CSeq: 22719 INVITE. Server: Asterisk PBX 1.6.2.6. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: . Content-Length: 0. . U 10.20.40.164:5060 -> 10.20.41.27:5060 #406 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.20.41.27:5060;branch=z9hG4bKPj2e8abe87-c676-4209-b8a1-2b975cbaf85b;received=10.20.41.27;rport=5060. From: " " ;tag=3ab0831a-af3e-43e9-a4d5-0cfefe87efd6. To: ;tag=as03777070. Call-ID: a0197842-bd9f-4cf2-973e-a542bb64da09. CSeq: 22719 INVITE. Server: Asterisk PBX 1.6.2.6. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: . Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 1547026156 1547026157 IN IP4 10.20.40.164. s=Asterisk PBX 1.6.2.6. c=IN IP4 10.20.40.164. t=0 0. m=audio 11050 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 10.20.41.27:5060 -> 10.20.40.164:5060 #407 ACK sip:901@10.20.40.164 SIP/2.0. Via: SIP/2.0/UDP 10.20.41.27:5060;rport;branch=z9hG4bKPjee955d99-a9e4-494b-9dcb-48d33dc3373b. From: " " ;tag=3ab0831a-af3e-43e9-a4d5-0cfefe87efd6. To: ;tag=as03777070. Call-ID: a0197842-bd9f-4cf2-973e-a542bb64da09. CSeq: 22719 ACK. Max-Forwards: 70. User-Agent: Asterisk PBX certified/13.21-cert2. Content-Length: 0. . U 10.20.40.164:5060 -> 10.20.41.27:5060 #419 BYE sip:asterisk@10.20.41.27:5060 SIP/2.0. Via: SIP/2.0/UDP 10.20.40.164:5060;branch=z9hG4bK56d101e4;rport. Max-Forwards: 70. From: ;tag=as03777070. To: " " ;tag=3ab0831a-af3e-43e9-a4d5-0cfefe87efd6. Call-ID: a0197842-bd9f-4cf2-973e-a542bb64da09. CSeq: 102 BYE. User-Agent: Asterisk PBX 1.6.2.6. X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Content-Length: 0. . U 10.20.41.27:5060 -> 10.20.40.164:5060 #420 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.20.40.164:5060;rport=5060;received=10.20.40.164;branch=z9hG4bK56d101e4. Call-ID: a0197842-bd9f-4cf2-973e-a542bb64da09. From: ;tag=as03777070. To: " " ;tag=3ab0831a-af3e-43e9-a4d5-0cfefe87efd6. CSeq: 102 BYE. Server: Asterisk PBX certified/13.21-cert2. Content-Length: 0. .