<--- SIP read from UDP:192.168.5.65:5060 ---> INVITE sip:202@172.16.16.91 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.65;rport;branch=z9hG4bKlusouaaz Max-Forwards: 70 To: From: "201" ;tag=ijpis Call-ID: gpjsgoymwtbbzyx@office CSeq: 153 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.9.0 Content-Length: 308 v=0 o=twinkle 1176550314 803481822 IN IP4 192.168.5.65 s=- c=IN IP4 192.168.5.65 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 14 lines) --- Sending to 192.168.5.65:5060 (no NAT) Sending to 192.168.5.65:5060 (no NAT) Using INVITE request as basis request - gpjsgoymwtbbzyx@office Found peer '201' for '201' from 192.168.5.65:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Found audio description format speex for ID 98 Found audio description format speex for ID 97 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|gsm|alaw|speex|speex16)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.5.65:8000 Looking for 202 in from-internal (domain 172.16.16.91) sip_route_dump: route/path hop: <--- Transmitting (NAT) to 192.168.5.65:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.65;branch=z9hG4bKlusouaaz;received=192.168.5.65;rport=5060 From: "201" ;tag=ijpis To: Call-ID: gpjsgoymwtbbzyx@office CSeq: 153 INVITE Server: Asterisk PBX 13.11.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [202@from-internal:1] Dial("SIP/201-00000000", "SIP/202") in new stack == Using SIP RTP CoS mark 5 Audio is at 22604 Adding codec alaw to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.7.31:15060: INVITE sip:202@192.168.7.31:15060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.91:5060;branch=z9hG4bK7afdfd3e;rport Max-Forwards: 70 From: "201" ;tag=as676cfea6 To: Contact: Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.11.2 Date: Wed, 05 Oct 2016 16:44:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 276 v=0 o=root 1463692269 1463692269 IN IP4 172.16.16.91 s=Asterisk PBX 13.11.2 c=IN IP4 172.16.16.91 t=0 0 m=audio 22604 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- -- Called SIP/202 <--- SIP read from UDP:192.168.7.31:15060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.91:5060;received=172.16.16.91;rport=5060;branch=z9hG4bK7afdfd3e To: From: "201" ;tag=as676cfea6 Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 102 INVITE Server: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.7.31:15060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.16.91:5060;received=172.16.16.91;rport=5060;branch=z9hG4bK7afdfd3e To: ;tag=vomgf From: "201" ;tag=as676cfea6 Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 102 INVITE Contact: Server: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/202-00000001 is ringing <--- Transmitting (NAT) to 192.168.5.65:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.5.65;branch=z9hG4bKlusouaaz;received=192.168.5.65;rport=5060 From: "201" ;tag=ijpis To: ;tag=as1417b6d5 Call-ID: gpjsgoymwtbbzyx@office CSeq: 153 INVITE Server: Asterisk PBX 13.11.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Really destroying SIP dialog '773085664@172.16.16.42' Method: OPTIONS <--- SIP read from UDP:192.168.7.31:15060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.91:5060;received=172.16.16.91;rport=5060;branch=z9hG4bK7afdfd3e To: ;tag=vomgf From: "201" ;tag=as676cfea6 Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Server: Twinkle/1.4.2 Supported: replaces,norefersub Content-Length: 206 v=0 o=twinkle 340368767 1490239799 IN IP4 192.168.7.31 s=- c=IN IP4 192.168.7.31 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (12 headers 10 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.7.31:8000 sip_route_dump: route/path hop: Transmitting (NAT) to 192.168.7.31:15060: ACK sip:202@192.168.7.31:15060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.91:5060;branch=z9hG4bK6200c3c6;rport Max-Forwards: 70 From: "201" ;tag=as676cfea6 To: ;tag=vomgf Contact: Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.11.2 Content-Length: 0 --- -- SIP/202-00000001 answered SIP/201-00000000 Audio is at 15018 Adding codec alaw to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.5.65:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.65;branch=z9hG4bKlusouaaz;received=192.168.5.65;rport=5060 From: "201" ;tag=ijpis To: ;tag=as1417b6d5 Call-ID: gpjsgoymwtbbzyx@office CSeq: 153 INVITE Server: Asterisk PBX 13.11.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 272 v=0 o=root 31336167 31336167 IN IP4 172.16.16.91 s=Asterisk PBX 13.11.2 c=IN IP4 172.16.16.91 t=0 0 m=audio 15018 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <------------> -- Channel SIP/202-00000001 joined 'simple_bridge' basic-bridge <7e89524c-36b6-452f-a9f6-eb4ed0d75eb5> -- Channel SIP/201-00000000 joined 'simple_bridge' basic-bridge <7e89524c-36b6-452f-a9f6-eb4ed0d75eb5> Retransmitting #1 (NAT) to 192.168.5.65:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.65;branch=z9hG4bKlusouaaz;received=192.168.5.65;rport=5060 From: "201" ;tag=ijpis To: ;tag=as1417b6d5 Call-ID: gpjsgoymwtbbzyx@office CSeq: 153 INVITE Server: Asterisk PBX 13.11.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 272 v=0 o=root 31336167 31336167 IN IP4 172.16.16.91 s=Asterisk PBX 13.11.2 c=IN IP4 172.16.16.91 t=0 0 m=audio 15018 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- > 0x28fb840 -- Probation passed - setting RTP source address to 192.168.7.31:8000 <--- SIP read from UDP:192.168.5.65:5060 ---> ACK sip:202@172.16.16.91:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.65;rport;branch=z9hG4bKghsrzafy Max-Forwards: 70 To: ;tag=as1417b6d5 From: "201" ;tag=ijpis Call-ID: gpjsgoymwtbbzyx@office CSeq: 153 ACK User-Agent: Twinkle/1.9.0 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.5.65:5060 ---> ACK sip:202@172.16.16.91:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.65;rport;branch=z9hG4bKghsrzafy Max-Forwards: 70 To: ;tag=as1417b6d5 From: "201" ;tag=ijpis Call-ID: gpjsgoymwtbbzyx@office CSeq: 153 ACK User-Agent: Twinkle/1.9.0 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- > 0x2b9f9f0 -- Probation passed - setting RTP source address to 192.168.5.65:8000 <--- SIP read from UDP:192.168.7.31:15060 ---> INVITE sip:201@172.16.16.91:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.31:15060;rport;branch=z9hG4bKmtjsjunr Max-Forwards: 70 To: "201" ;tag=as676cfea6 From: ;tag=vomgf Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 755 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.4.2 Content-Length: 320 v=0 o=twinkle 340368767 1490239800 IN IP4 192.168.7.31 s=- c=IN IP4 192.168.7.31 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendonly <-------------> --- (13 headers 15 lines) --- Sending to 192.168.7.31:15060 (NAT) Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Found audio description format speex for ID 98 Found audio description format speex for ID 97 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|gsm|alaw|speex|speex16)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.7.31:8000 <--- Transmitting (NAT) to 192.168.7.31:15060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.7.31:15060;branch=z9hG4bKmtjsjunr;received=192.168.7.31;rport=15060 From: ;tag=vomgf To: "201" ;tag=as676cfea6 Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 755 INVITE Server: Asterisk PBX 13.11.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 22604 Adding codec alaw to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.7.31:15060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.31:15060;branch=z9hG4bKmtjsjunr;received=192.168.7.31;rport=15060 From: ;tag=vomgf To: "201" ;tag=as676cfea6 Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 755 INVITE Server: Asterisk PBX 13.11.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 276 v=0 o=root 1463692269 1463692270 IN IP4 172.16.16.91 s=Asterisk PBX 13.11.2 c=IN IP4 172.16.16.91 t=0 0 m=audio 22604 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=recvonly <------------> -- Music class default requested but no musiconhold loaded. > 0x28fb840 -- Probation passed - setting RTP source address to 192.168.7.31:8000 <--- SIP read from UDP:192.168.7.31:15060 ---> ACK sip:201@172.16.16.91:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.31:15060;rport;branch=z9hG4bKdcdoxtcf Max-Forwards: 70 To: "201" ;tag=as676cfea6 From: ;tag=vomgf Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 755 ACK User-Agent: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Reliably Transmitting (NAT) to 192.168.7.31:15060: OPTIONS sip:202@192.168.7.31:15060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.91:5060;branch=z9hG4bK2fc070b8;rport Max-Forwards: 70 From: "asterisk" ;tag=as384b9daa To: Contact: Call-ID: 5050a1eb4f17f347437fc0461bc493b5@172.16.16.91:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.11.2 Date: Wed, 05 Oct 2016 16:44:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.7.31:15060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.91:5060;received=172.16.16.91;rport=5060;branch=z9hG4bK2fc070b8 To: ;tag=bcqqx From: "asterisk" ;tag=as384b9daa Call-ID: 5050a1eb4f17f347437fc0461bc493b5@172.16.16.91:5060 CSeq: 102 OPTIONS Accept: application/sdp Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Server: Twinkle/1.4.2 Supported: replaces,norefersub,100rel Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '5050a1eb4f17f347437fc0461bc493b5@172.16.16.91:5060' Method: OPTIONS <--- SIP read from UDP:192.168.7.31:15060 ---> INVITE sip:125@172.16.16.91 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.31:15060;rport;branch=z9hG4bKlrbmvtet Max-Forwards: 70 To: From: "202" ;tag=kawpp Call-ID: alspjympmhyetvt@debian-one CSeq: 746 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Subject: Call transfer - 201 Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.4.2 Content-Length: 307 v=0 o=twinkle 1996014822 83478050 IN IP4 192.168.7.31 s=- c=IN IP4 192.168.7.31 t=0 0 m=audio 8002 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 14 lines) --- Sending to 192.168.7.31:15060 (no NAT) Sending to 192.168.7.31:15060 (no NAT) Using INVITE request as basis request - alspjympmhyetvt@debian-one Found peer '202' for '202' from 192.168.7.31:15060 == Using SIP RTP CoS mark 5 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Found audio description format speex for ID 98 Found audio description format speex for ID 97 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|gsm|alaw|speex|speex16)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.7.31:8002 Looking for 125 in from-internal (domain 172.16.16.91) sip_route_dump: route/path hop: <--- Transmitting (NAT) to 192.168.7.31:15060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.7.31:15060;branch=z9hG4bKlrbmvtet;received=192.168.7.31;rport=15060 From: "202" ;tag=kawpp To: Call-ID: alspjympmhyetvt@debian-one CSeq: 746 INVITE Server: Asterisk PBX 13.11.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [125@from-internal:1] Answer("SIP/202-00000002", "") in new stack Audio is at 17350 Adding codec alaw to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.7.31:15060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.31:15060;branch=z9hG4bKlrbmvtet;received=192.168.7.31;rport=15060 From: "202" ;tag=kawpp To: ;tag=as76402a1d Call-ID: alspjympmhyetvt@debian-one CSeq: 746 INVITE Server: Asterisk PBX 13.11.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 276 v=0 o=root 1994806438 1994806438 IN IP4 172.16.16.91 s=Asterisk PBX 13.11.2 c=IN IP4 172.16.16.91 t=0 0 m=audio 17350 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <------------> <--- SIP read from UDP:192.168.7.31:15060 ---> ACK sip:125@172.16.16.91:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.31:15060;rport;branch=z9hG4bKacvnukwu Max-Forwards: 70 To: ;tag=as76402a1d From: "202" ;tag=kawpp Call-ID: alspjympmhyetvt@debian-one CSeq: 746 ACK User-Agent: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- > 0x2b530a0 -- Probation passed - setting RTP source address to 192.168.7.31:8002 -- Executing [125@from-internal:2] Wait("SIP/202-00000002", "5") in new stack -- Executing [125@from-internal:3] ExternalIVR("SIP/202-00000002", "/etc/asterisk/ivr/test1.sh") in new stack > ExternalIVR received application and arguments: /etc/asterisk/ivr/test1.sh > ExternalIVR received options: (null) > Parsing options from: [(null)] -- Answering channel and starting generator > got command 'S,/etc/asterisk/sounds/track1' <--- SIP read from UDP:192.168.7.31:15060 ---> REFER sip:201@172.16.16.91:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.31:15060;rport;branch=z9hG4bKmzjfmjdd Max-Forwards: 70 To: "201" ;tag=as676cfea6 From: ;tag=vomgf Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 756 REFER Contact: Refer-To: Referred-By: "202" User-Agent: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Call 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 125@from-internal by 202@172.16.16.91 <--- Transmitting (NAT) to 192.168.7.31:15060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.7.31:15060;branch=z9hG4bKmzjfmjdd;received=192.168.7.31;rport=15060 From: ;tag=vomgf To: "201" ;tag=as676cfea6 Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 756 REFER Server: Asterisk PBX 13.11.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Reliably Transmitting (NAT) to 192.168.7.31:15060: NOTIFY sip:202@192.168.7.31:15060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.91:5060;branch=z9hG4bK2a6cddc4;rport Max-Forwards: 70 From: "201" ;tag=as676cfea6 To: ;tag=vomgf Contact: Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 103 NOTIFY User-Agent: Asterisk PBX 13.11.2 Event: refer;id=756 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 16 SIP/2.0 200 OK --- -- Channel SIP/201-00000000 left 'simple_bridge' basic-bridge <7e89524c-36b6-452f-a9f6-eb4ed0d75eb5> -- Channel SIP/202-00000001 left 'simple_bridge' basic-bridge <7e89524c-36b6-452f-a9f6-eb4ed0d75eb5> Scheduling destruction of SIP dialog '210d9ca05415518861583f42270d16b7@172.16.16.91:5060' in 6400 ms (Method: REFER) <--- SIP read from UDP:192.168.7.31:15060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.91:5060;received=172.16.16.91;rport=5060;branch=z9hG4bK2a6cddc4 To: ;tag=vomgf From: "201" ;tag=as676cfea6 Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 103 NOTIFY Server: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.7.31:15060 ---> BYE sip:201@172.16.16.91:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.31:15060;rport;branch=z9hG4bKazouccpv Max-Forwards: 70 To: "201" ;tag=as676cfea6 From: ;tag=vomgf Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 757 BYE User-Agent: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 192.168.7.31:15060 (NAT) Scheduling destruction of SIP dialog '210d9ca05415518861583f42270d16b7@172.16.16.91:5060' in 6400 ms (Method: BYE) <--- Transmitting (NAT) to 192.168.7.31:15060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.7.31:15060;branch=z9hG4bKazouccpv;received=192.168.7.31;rport=15060 From: ;tag=vomgf To: "201" ;tag=as676cfea6 Call-ID: 210d9ca05415518861583f42270d16b7@172.16.16.91:5060 CSeq: 757 BYE Server: Asterisk PBX 13.11.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (from-internal, 202, 1) exited non-zero on 'SIP/202-00000002' Scheduling destruction of SIP dialog 'alspjympmhyetvt@debian-one' in 6400 ms (Method: ACK) Reliably Transmitting (NAT) to 192.168.7.31:15060: BYE sip:202@192.168.7.31:15060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.91:5060;branch=z9hG4bK51e13007;rport Max-Forwards: 70 From: ;tag=as76402a1d To: "202" ;tag=kawpp Call-ID: alspjympmhyetvt@debian-one CSeq: 102 BYE User-Agent: Asterisk PBX 13.11.2 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:192.168.7.31:15060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.91:5060;received=172.16.16.91;rport=5060;branch=z9hG4bK51e13007 To: "202" ;tag=kawpp From: ;tag=as76402a1d Call-ID: alspjympmhyetvt@debian-one CSeq: 102 BYE Server: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'alspjympmhyetvt@debian-one' Method: ACK Really destroying SIP dialog 'fjsxklzysqdmeuo@office' Method: REGISTER <--- SIP read from UDP:192.168.5.65:5060 ---> BYE sip:202@172.16.16.91:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.65;rport;branch=z9hG4bKcaukssvp Max-Forwards: 70 To: ;tag=as1417b6d5 From: "201" ;tag=ijpis Call-ID: gpjsgoymwtbbzyx@office CSeq: 154 BYE User-Agent: Twinkle/1.9.0 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 192.168.5.65:5060 (NAT) Scheduling destruction of SIP dialog 'gpjsgoymwtbbzyx@office' in 6400 ms (Method: BYE) <--- Transmitting (NAT) to 192.168.5.65:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.65;branch=z9hG4bKcaukssvp;received=192.168.5.65;rport=5060 From: "201" ;tag=ijpis To: ;tag=as1417b6d5 Call-ID: gpjsgoymwtbbzyx@office CSeq: 154 BYE Server: Asterisk PBX 13.11.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> -- Got check_hangup == Spawn extension (from-internal, 125, 3) exited non-zero on 'SIP/201-00000000' Really destroying SIP dialog '210d9ca05415518861583f42270d16b7@172.16.16.91:5060' Method: BYE Really destroying SIP dialog 'gpjsgoymwtbbzyx@office' Method: BYE