asterisk*CLI> sip set debug on SIP Debugging enabled Retransmitting #4 (no NAT) to 10.1.52.250:5060: OPTIONS sip:10.1.52.250 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK6c5d7c26 Max-Forwards: 70 From: "asterisk" ;tag=as7c919ae9 To: Contact: Call-ID: 6a8c15396b0765ea160076137e41ae39@10.15.101.125:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 14.2.1 Date: Fri, 24 Feb 2017 15:13:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '6a8c15396b0765ea160076137e41ae39@10.15.101.125:5060' Method: OPTIONS <--- SIP read from UDP:10.15.101.103:5060 ---> INVITE sip:7000@10.15.101.125 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.103:5060;branch=z9hG4bK519043737;rport From: ;tag=239881640 To: Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 20 INVITE Contact: Max-Forwards: 70 User-Agent: Grandstream GXP1405 1.0.8.9 Privacy: none P-Preferred-Identity: Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 407 v=0 o=ext7010 8000 8000 IN IP4 10.15.101.103 s=SIP Call c=IN IP4 10.15.101.103 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (16 headers 19 lines) --- Sending to 10.15.101.103:5060 (no NAT) Sending to 10.15.101.103:5060 (no NAT) Using INVITE request as basis request - 1518768481-5060-3@BA.BF.BAB.BAD Found peer 'ext7010' for 'ext7010' from 10.15.101.103:5060 <--- Reliably Transmitting (no NAT) to 10.15.101.103:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.15.101.103:5060;branch=z9hG4bK519043737;received=10.15.101.103;rport=5060 From: ;tag=239881640 To: ;tag=as5c99606d Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 20 INVITE Server: Asterisk PBX 14.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a7f1ac8" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1518768481-5060-3@BA.BF.BAB.BAD' in 6400 ms (Method: INVITE) <--- SIP read from UDP:10.15.101.103:5060 ---> ACK sip:7000@10.15.101.125 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.103:5060;branch=z9hG4bK519043737;rport From: ;tag=239881640 To: ;tag=as5c99606d Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 20 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:10.15.101.103:5060 ---> INVITE sip:7000@10.15.101.125 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.103:5060;branch=z9hG4bK1333506714;rport From: ;tag=239881640 To: Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 21 INVITE Contact: Authorization: Digest username="ext7010", realm="asterisk", nonce="7a7f1ac8", uri="sip:7000@10.15.101.125", response="117b5fbad448de973d13dfc40c400e15", algorithm=MD5 Max-Forwards: 70 User-Agent: Grandstream GXP1405 1.0.8.9 Privacy: none P-Preferred-Identity: Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 407 v=0 o=ext7010 8000 8000 IN IP4 10.15.101.103 s=SIP Call c=IN IP4 10.15.101.103 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (17 headers 19 lines) --- Sending to 10.15.101.103:5060 (no NAT) Using INVITE request as basis request - 1518768481-5060-3@BA.BF.BAB.BAD Found peer 'ext7010' for 'ext7010' from 10.15.101.103:5060 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 9 Found RTP audio format 97 Found RTP audio format 2 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G723 for ID 4 Found audio description format G729 for ID 18 Found audio description format G722 for ID 9 Found audio description format iLBC for ID 97 Found audio description format G726-32 for ID 2 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|g726|g723|alaw|g722|g729|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.15.101.103:5004 Looking for 7000 in National_Access (domain 10.15.101.125) sip_route_dump: route/path hop: <--- Transmitting (no NAT) to 10.15.101.103:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.101.103:5060;branch=z9hG4bK1333506714;received=10.15.101.103;rport=5060 From: ;tag=239881640 To: Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 21 INVITE Server: Asterisk PBX 14.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 19438 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.15.101.111:53071: INVITE sip:ext7000@10.15.101.111:53071;rinstance=061bec11ccaff80c SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK01fd258d Max-Forwards: 70 From: "Boss" ;tag=as23fbbcd3 To: Contact: Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.2.1 Date: Fri, 24 Feb 2017 15:13:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "Boss" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 286 v=0 o=root 736425571 736425571 IN IP4 10.15.101.125 s=Asterisk PBX 14.2.1 c=IN IP4 10.15.101.125 t=0 0 m=audio 19438 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:10.15.101.111:53071 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK01fd258d Contact: To: ;tag=605a2500 From: "Boss";tag=as23fbbcd3 Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 102 INVITE User-Agent: 3CXPhone 6.0.26523.0 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip_route_dump: route/path hop: Retransmitting #1 (no NAT) to 10.15.101.111:53071: INVITE sip:ext7000@10.15.101.111:53071;rinstance=061bec11ccaff80c SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK01fd258d Max-Forwards: 70 From: "Boss" ;tag=as23fbbcd3 To: Contact: Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.2.1 Date: Fri, 24 Feb 2017 15:13:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "Boss" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 286 v=0 o=root 736425571 736425571 IN IP4 10.15.101.125 s=Asterisk PBX 14.2.1 c=IN IP4 10.15.101.125 t=0 0 m=audio 19438 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- <--- Transmitting (no NAT) to 10.15.101.103:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.15.101.103:5060;branch=z9hG4bK1333506714;received=10.15.101.103;rport=5060 From: ;tag=239881640 To: ;tag=as355a2b3f Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 21 INVITE Server: Asterisk PBX 14.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> <--- SIP read from UDP:10.15.101.111:53071 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK01fd258d Contact: To: ;tag=605a2500 From: "Boss";tag=as23fbbcd3 Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 102 INVITE User-Agent: 3CXPhone 6.0.26523.0 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip_route_dump: route/path hop: <--- SIP read from UDP:10.15.101.111:53071 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK01fd258d Contact: To: ;tag=605a2500 From: "Boss";tag=as23fbbcd3 Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 6.0.26523.0 Content-Length: 258 v=0 o=3cxVCE 290218605 100179675 IN IP4 10.15.101.111 s=3cxVCE Audio Call c=IN IP4 10.15.101.111 t=0 0 m=audio 40024 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.15.101.111:40024 sip_route_dump: route/path hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.15.101.111:53071 Transmitting (no NAT) to 10.15.101.111:53071: ACK sip:ext7000@10.15.101.111:53071;rinstance=061bec11ccaff80c SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK4d6d47d0 Max-Forwards: 70 From: "Boss" ;tag=as23fbbcd3 To: ;tag=605a2500 Contact: Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 14.2.1 Content-Length: 0 --- Audio is at 16162 Adding codec ulaw to SDP Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.15.101.103:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.101.103:5060;branch=z9hG4bK1333506714;received=10.15.101.103;rport=5060 From: ;tag=239881640 To: ;tag=as355a2b3f Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 21 INVITE Server: Asterisk PBX 14.2.1 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: 265 v=0 o=root 1254584659 1254584659 IN IP4 10.15.101.125 s=Asterisk PBX 14.2.1 c=IN IP4 10.15.101.125 t=0 0 m=audio 16162 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 10.15.101.111:53071 Audio is at 19438 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.15.101.111:53071: INVITE sip:ext7000@10.15.101.111:53071;rinstance=061bec11ccaff80c SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK6dfeabf0 Max-Forwards: 70 From: "Boss" ;tag=as23fbbcd3 To: ;tag=605a2500 Contact: Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 14.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "Boss" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 285 v=0 o=root 736425571 736425572 IN IP4 10.15.101.103 s=Asterisk PBX 14.2.1 c=IN IP4 10.15.101.103 t=0 0 m=audio 5004 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:10.15.101.103:5060 ---> ACK sip:7000@10.15.101.125:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.103:5060;branch=z9hG4bK2061851976;rport From: ;tag=239881640 To: ;tag=as355a2b3f Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 21 ACK Contact: Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXP1405 1.0.8.9 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.15.101.103:5060 Audio is at 16162 Adding codec ulaw to SDP Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.15.101.103:5060: INVITE sip:ext7010@10.15.101.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK2721c49c;rport Max-Forwards: 70 From: ;tag=as355a2b3f To: ;tag=239881640 Contact: Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 102 INVITE User-Agent: Asterisk PBX 14.2.1 Session-Expires: 1800;refresher=uac Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 265 v=0 o=root 1254584659 1254584660 IN IP4 10.15.101.111 s=Asterisk PBX 14.2.1 c=IN IP4 10.15.101.111 t=0 0 m=audio 40024 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- Retransmitting #1 (no NAT) to 10.15.101.111:53071: INVITE sip:ext7000@10.15.101.111:53071;rinstance=061bec11ccaff80c SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK6dfeabf0 Max-Forwards: 70 From: "Boss" ;tag=as23fbbcd3 To: ;tag=605a2500 Contact: Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 14.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "Boss" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 285 v=0 o=root 736425571 736425572 IN IP4 10.15.101.103 s=Asterisk PBX 14.2.1 c=IN IP4 10.15.101.103 t=0 0 m=audio 5004 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:10.15.101.111:53071 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK6dfeabf0 Contact: To: ;tag=605a2500 From: "Boss";tag=as23fbbcd3 Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 6.0.26523.0 Content-Length: 258 v=0 o=3cxVCE 290218605 100179676 IN IP4 10.15.101.111 s=3cxVCE Audio Call c=IN IP4 10.15.101.111 t=0 0 m=audio 40024 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.15.101.111:40024 set_destination: Parsing for address/port to send to set_destination: set destination to 10.15.101.111:53071 Transmitting (no NAT) to 10.15.101.111:53071: ACK sip:ext7000@10.15.101.111:53071;rinstance=061bec11ccaff80c SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK64d748fb Max-Forwards: 70 From: "Boss" ;tag=as23fbbcd3 To: ;tag=605a2500 Contact: Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 14.2.1 Content-Length: 0 --- Retransmitting #1 (no NAT) to 10.15.101.103:5060: INVITE sip:ext7010@10.15.101.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK2721c49c;rport Max-Forwards: 70 From: ;tag=as355a2b3f To: ;tag=239881640 Contact: Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 102 INVITE User-Agent: Asterisk PBX 14.2.1 Session-Expires: 1800;refresher=uac Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 265 v=0 o=root 1254584659 1254584660 IN IP4 10.15.101.111 s=Asterisk PBX 14.2.1 c=IN IP4 10.15.101.111 t=0 0 m=audio 40024 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:10.15.101.103:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK2721c49c;rport=5060 From: ;tag=as355a2b3f To: ;tag=239881640 Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 102 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP1405 1.0.8.9 Require: timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:10.15.101.103:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK2721c49c;rport=5060 From: ;tag=as355a2b3f To: ;tag=239881640 Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 102 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP1405 1.0.8.9 Require: timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:10.15.101.103:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK2721c49c;rport=5060 From: ;tag=as355a2b3f To: ;tag=239881640 Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 102 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP1405 1.0.8.9 Session-Expires: 1800;refresher=uac Require: timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 240 v=0 o=ext7010 8000 8001 IN IP4 10.15.101.103 s=SIP Call c=IN IP4 10.15.101.103 t=0 0 m=audio 5004 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (14 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.15.101.103:5004 set_destination: Parsing for address/port to send to set_destination: set destination to 10.15.101.103:5060 Transmitting (no NAT) to 10.15.101.103:5060: ACK sip:ext7010@10.15.101.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK4d0b2214;rport Max-Forwards: 70 From: ;tag=as355a2b3f To: ;tag=239881640 Contact: Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 102 ACK User-Agent: Asterisk PBX 14.2.1 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.15.101.111:53071 Audio is at 19438 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.15.101.111:53071: INVITE sip:ext7000@10.15.101.111:53071;rinstance=061bec11ccaff80c SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK4436dfb2 Max-Forwards: 70 From: "Boss" ;tag=as23fbbcd3 To: ;tag=605a2500 Contact: Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 14.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "Boss" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 285 v=0 o=root 736425571 736425573 IN IP4 10.15.101.103 s=Asterisk PBX 14.2.1 c=IN IP4 10.15.101.103 t=0 0 m=audio 5004 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:10.15.101.111:53071 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK4436dfb2 Contact: To: ;tag=605a2500 From: "Boss";tag=as23fbbcd3 Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 6.0.26523.0 Content-Length: 258 v=0 o=3cxVCE 290218605 100179677 IN IP4 10.15.101.111 s=3cxVCE Audio Call c=IN IP4 10.15.101.111 t=0 0 m=audio 40024 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.15.101.111:40024 set_destination: Parsing for address/port to send to set_destination: set destination to 10.15.101.111:53071 Transmitting (no NAT) to 10.15.101.111:53071: ACK sip:ext7000@10.15.101.111:53071;rinstance=061bec11ccaff80c SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK76d9c7b9 Max-Forwards: 70 From: "Boss" ;tag=as23fbbcd3 To: ;tag=605a2500 Contact: Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 14.2.1 Content-Length: 0 --- <--- SIP read from UDP:10.15.101.111:53071 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK4436dfb2 Contact: To: ;tag=605a2500 From: "Boss";tag=as23fbbcd3 Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 6.0.26523.0 Content-Length: 258 v=0 o=3cxVCE 290218605 100179677 IN IP4 10.15.101.111 s=3cxVCE Audio Call c=IN IP4 10.15.101.111 t=0 0 m=audio 40024 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (12 headers 11 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.15.101.111:53071 Transmitting (no NAT) to 10.15.101.111:53071: ACK sip:ext7000@10.15.101.111:53071;rinstance=061bec11ccaff80c SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK35764a3e Max-Forwards: 70 From: "Boss" ;tag=as23fbbcd3 To: ;tag=605a2500 Contact: Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 14.2.1 Content-Length: 0 --- <--- SIP read from UDP:10.15.101.111:53071 ---> <-------------> Reliably Transmitting (no NAT) to 10.1.52.250:5060: OPTIONS sip:10.1.52.250 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK294edecb Max-Forwards: 70 From: "asterisk" ;tag=as00e4c61f To: Contact: Call-ID: 6c6d2b0962800d2d14007b26655c2fcc@10.15.101.125:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 14.2.1 Date: Fri, 24 Feb 2017 15:13:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #1 (no NAT) to 10.1.52.250:5060: OPTIONS sip:10.1.52.250 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK294edecb Max-Forwards: 70 From: "asterisk" ;tag=as00e4c61f To: Contact: Call-ID: 6c6d2b0962800d2d14007b26655c2fcc@10.15.101.125:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 14.2.1 Date: Fri, 24 Feb 2017 15:13:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.15.101.111:53071 ---> BYE sip:7010@10.15.101.125:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.111:53071;branch=z9hG4bK-d8754z-634660209f3b5b09-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Boss";tag=as23fbbcd3 From: ;tag=605a2500 Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 2 BYE User-Agent: 3CXPhone 6.0.26523.0 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.15.101.111:53071 (no NAT) Scheduling destruction of SIP dialog '7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060' in 7040 ms (Method: BYE) <--- Transmitting (no NAT) to 10.15.101.111:53071 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.101.111:53071;branch=z9hG4bK-d8754z-634660209f3b5b09-1---d8754z-;received=10.15.101.111;rport=53071 From: ;tag=605a2500 To: "Boss";tag=as23fbbcd3 Call-ID: 7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060 CSeq: 2 BYE Server: Asterisk PBX 14.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 10.15.101.103:5060 Audio is at 16162 Adding codec ulaw to SDP Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.15.101.103:5060: INVITE sip:ext7010@10.15.101.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK4bee62fd;rport Max-Forwards: 70 From: ;tag=as355a2b3f To: ;tag=239881640 Contact: Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 103 INVITE User-Agent: Asterisk PBX 14.2.1 Session-Expires: 1800;refresher=uac Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 265 v=0 o=root 1254584659 1254584661 IN IP4 10.15.101.125 s=Asterisk PBX 14.2.1 c=IN IP4 10.15.101.125 t=0 0 m=audio 16162 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- Scheduling destruction of SIP dialog '1518768481-5060-3@BA.BF.BAB.BAD' in 6400 ms (Method: ACK) <--- SIP read from UDP:10.15.101.103:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK4bee62fd;rport=5060 From: ;tag=as355a2b3f To: ;tag=239881640 Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 103 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP1405 1.0.8.9 Require: timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:10.15.101.103:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK4bee62fd;rport=5060 From: ;tag=as355a2b3f To: ;tag=239881640 Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 103 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP1405 1.0.8.9 Session-Expires: 1800;refresher=uac Require: timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 240 v=0 o=ext7010 8000 8002 IN IP4 10.15.101.103 s=SIP Call c=IN IP4 10.15.101.103 t=0 0 m=audio 5004 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (14 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.15.101.103:5004 set_destination: Parsing for address/port to send to set_destination: set destination to 10.15.101.103:5060 Transmitting (no NAT) to 10.15.101.103:5060: ACK sip:ext7010@10.15.101.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK301867eb;rport Max-Forwards: 70 From: ;tag=as355a2b3f To: ;tag=239881640 Contact: Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 103 ACK User-Agent: Asterisk PBX 14.2.1 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.15.101.103:5060 Reliably Transmitting (no NAT) to 10.15.101.103:5060: BYE sip:ext7010@10.15.101.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK457a64b0;rport Max-Forwards: 70 From: ;tag=as355a2b3f To: ;tag=239881640 Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 104 BYE User-Agent: Asterisk PBX 14.2.1 Proxy-Authorization: Digest username="ext7010", realm="asterisk", algorithm=MD5, uri="sip:10.15.101.125", nonce="7a7f1ac8", response="fede99b46622e91fea586541d757fd0f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '1518768481-5060-3@BA.BF.BAB.BAD' in 6400 ms (Method: ACK) <--- SIP read from UDP:10.15.101.103:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK457a64b0;rport=5060 From: ;tag=as355a2b3f To: ;tag=239881640 Call-ID: 1518768481-5060-3@BA.BF.BAB.BAD CSeq: 104 BYE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP1405 1.0.8.9 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '1518768481-5060-3@BA.BF.BAB.BAD' Method: ACK Retransmitting #2 (no NAT) to 10.1.52.250:5060: OPTIONS sip:10.1.52.250 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK294edecb Max-Forwards: 70 From: "asterisk" ;tag=as00e4c61f To: Contact: Call-ID: 6c6d2b0962800d2d14007b26655c2fcc@10.15.101.125:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 14.2.1 Date: Fri, 24 Feb 2017 15:13:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 10.15.101.111:53071: OPTIONS sip:ext7000@10.15.101.111:53071;rinstance=061bec11ccaff80c SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK43789b03 Max-Forwards: 70 From: "asterisk" ;tag=as072157c3 To: Contact: Call-ID: 79ca60c033e43772517da32b1d6d3c2a@10.15.101.125:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 14.2.1 Date: Fri, 24 Feb 2017 15:13:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.15.101.111:53071 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK43789b03 Contact: To: ;tag=b74e8859 From: "asterisk";tag=as072157c3 Call-ID: 79ca60c033e43772517da32b1d6d3c2a@10.15.101.125:5060 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Supported: replaces Allow-Events: presence, message-summary, tunnel-info Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '79ca60c033e43772517da32b1d6d3c2a@10.15.101.125:5060' Method: OPTIONS Retransmitting #3 (no NAT) to 10.1.52.250:5060: OPTIONS sip:10.1.52.250 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK294edecb Max-Forwards: 70 From: "asterisk" ;tag=as00e4c61f To: Contact: Call-ID: 6c6d2b0962800d2d14007b26655c2fcc@10.15.101.125:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 14.2.1 Date: Fri, 24 Feb 2017 15:13:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (no NAT) to 10.1.52.250:5060: OPTIONS sip:10.1.52.250 SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK294edecb Max-Forwards: 70 From: "asterisk" ;tag=as00e4c61f To: Contact: Call-ID: 6c6d2b0962800d2d14007b26655c2fcc@10.15.101.125:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 14.2.1 Date: Fri, 24 Feb 2017 15:13:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '6c6d2b0962800d2d14007b26655c2fcc@10.15.101.125:5060' Method: OPTIONS Reliably Transmitting (no NAT) to 10.15.101.111:34831: OPTIONS sip:ext7001@10.15.101.111:34831;rinstance=1d0e160518e474f1;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK255c78c1 Max-Forwards: 70 From: "asterisk" ;tag=as00756d01 To: Contact: Call-ID: 57e39225194f43cf12f3994116e3139a@10.15.101.125:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 14.2.1 Date: Fri, 24 Feb 2017 15:13:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.15.101.111:34831 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.101.125:5060;branch=z9hG4bK255c78c1 Contact: To: ;tag=534c9545 From: "asterisk" ;tag=as00756d01 Call-ID: 57e39225194f43cf12f3994116e3139a@10.15.101.125:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '57e39225194f43cf12f3994116e3139a@10.15.101.125:5060' Method: OPTIONS Really destroying SIP dialog '7a36a1e971e1907d7e9c647f1e312605@10.15.101.125:5060' Method: BYE asterisk*CLI> sip set debug off SIP Debugging Disabled