[2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: <--- SIP read from TCP:192.168.125.116:5062 ---> INVITE sip:10081@172.17.9.75 SIP/2.0 Accept: application/sdp, message/sipfrag, multipart/mixed Via: SIP/2.0/TCP 192.168.125.116:5062;branch=z9hG4bKa0d92dff8493d475e Max-Forwards: 70 From: "1100203" ;tag=693ca13dce To: Call-ID: a317c3d69eca9877 CSeq: 603883884 INVITE Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE Allow-Events: refer Contact: Min-SE: 90 Session-Expires: 1800 Supported: replaces, timer User-Agent: Media5-fone/2.14.1.194 Content-Disposition: session Content-Type: application/sdp Content-Length: 304 v=0 o=MxSIP 359101844977660356 359101844977660357 IN IP4 192.168.125.116 s=SIP Call c=IN IP4 192.168.125.116 t=0 0 m=audio 10000 RTP/AVP 0 8 96 125 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 iLBC/8000 a=rtpmap:125 telephone-event/8000 a=fmtp:96 mode=30 a=fmtp:125 0-15 a=sendrecv <-------------> [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: --- (18 headers 13 lines) --- [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Sending to 192.168.125.116:5062 (NAT) [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Using INVITE request as basis request - a317c3d69eca9877 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Found peer '1100203' for '1100203' from 192.168.125.116:5062 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.125.116:5062 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.125.116:5062;branch=z9hG4bKa0d92dff8493d475e;received=192.168.125.116;rport=5062 From: "1100203" ;tag=693ca13dce To: ;tag=as62a68c78 Call-ID: a317c3d69eca9877 CSeq: 603883884 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="32c72d32" Content-Length: 0 <------------> [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Scheduling destruction of SIP dialog 'a317c3d69eca9877' in 8512 ms (Method: INVITE) [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: <--- SIP read from TCP:192.168.125.116:5062 ---> ACK sip:10081@172.17.9.75 SIP/2.0 Via: SIP/2.0/TCP 192.168.125.116:5062;branch=z9hG4bKa0d92dff8493d475e Max-Forwards: 70 From: "1100203" ;tag=693ca13dce To: ;tag=as62a68c78 Call-ID: a317c3d69eca9877 CSeq: 603883884 ACK User-Agent: Media5-fone/2.14.1.194 Content-Length: 0 <-------------> [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: --- (9 headers 0 lines) --- [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: <--- SIP read from TCP:192.168.125.116:5062 ---> INVITE sip:10081@172.17.9.75 SIP/2.0 Accept: application/sdp, message/sipfrag, multipart/mixed Via: SIP/2.0/TCP 192.168.125.116:5062;branch=z9hG4bK938bfc937287e8715 Max-Forwards: 70 From: "1100203" ;tag=693ca13dce To: Call-ID: a317c3d69eca9877 CSeq: 603883885 INVITE Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE Allow-Events: refer Authorization: Digest username="1100203",realm="asterisk",nonce="32c72d32",uri="sip:10081@172.17.9.75",response="3f765e7106887bcd5a4655f9ffee86fd",algorithm=MD5 Contact: Min-SE: 90 Session-Expires: 1800 Supported: replaces, timer User-Agent: Media5-fone/2.14.1.194 Content-Disposition: session Content-Type: application/sdp Content-Length: 304 v=0 o=MxSIP 359101844977660356 359101844977660357 IN IP4 192.168.125.116 s=SIP Call c=IN IP4 192.168.125.116 t=0 0 m=audio 10000 RTP/AVP 0 8 96 125 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 iLBC/8000 a=rtpmap:125 telephone-event/8000 a=fmtp:96 mode=30 a=fmtp:125 0-15 a=sendrecv <-------------> [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: --- (19 headers 13 lines) --- [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Sending to 192.168.125.116:5062 (NAT) [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Using INVITE request as basis request - a317c3d69eca9877 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Found peer '1100203' for '1100203' from 192.168.125.116:5062 [2012-06-06 10:52:57] VERBOSE[18326] netsock2.c: == Using SIP RTP CoS mark 5 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Found RTP audio format 0 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Found RTP audio format 8 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Found RTP audio format 96 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Found RTP audio format 125 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Found audio description format PCMU for ID 0 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Found audio description format PCMA for ID 8 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Found audio description format iLBC for ID 96 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Found audio description format telephone-event for ID 125 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x40c (ulaw|alaw|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Peer audio RTP is at port 192.168.125.116:10000 [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: Looking for 10081 in DLPN_DialPlan1 (domain 172.17.9.75) [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: list_route: hop: [2012-06-06 10:52:57] VERBOSE[18326] chan_sip.c: <--- Transmitting (NAT) to 192.168.125.116:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.125.116:5062;branch=z9hG4bK938bfc937287e8715;received=192.168.125.116;rport=5062 From: "1100203" ;tag=693ca13dce To: Call-ID: a317c3d69eca9877 CSeq: 603883885 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 180;refresher=uas Contact: Content-Length: 0 <------------> [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [10081@DLPN_DialPlan1:1] Macro("SIP/1100203-00000002", "stdexten,10081,SIP/10081") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:1] NoOp("SIP/1100203-00000002", "Old Delete Location") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:2] Set("SIP/1100203-00000002", "LOOPCNT=100") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:3] NoOp("SIP/1100203-00000002", "Call to = SIP/10081") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:4] NoOp("SIP/1100203-00000002", "Originating Context = DLPN_DialPlan1") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:5] NoOp("SIP/1100203-00000002", "Caller ID is: 1100203") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:6] Set("SIP/1100203-00000002", "ORIGCONTEXT=macro-stdexten") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:7] NoOp("SIP/1100203-00000002", "Loops count is: 100") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:8] GotoIf("SIP/1100203-00000002", "1?:lbl_MAXLOOPS_0") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:9] GotoIf("SIP/1100203-00000002", "1?:lbl_RECORDNAME") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:10] NoOp("SIP/1100203-00000002", "10081@DLPN_Dialplan1 has state NOT_INUSE") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:11] Set("SIP/1100203-00000002", "CDR(userfield)=WEB_SERVICE_LOOP") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:12] Set("SIP/1100203-00000002", "STATE1=NOT_INUSE") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:13] GotoIf("SIP/1100203-00000002", "0?lbl_WebServiceCall_0:") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:14] GotoIf("SIP/1100203-00000002", "0?lbl_ON_PHONE_0:") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:15] GotoIf("SIP/1100203-00000002", "0?lbl_ON_PHONE_0:") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:16] Set("SIP/1100203-00000002", "CDR(userfield)=DIALING") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:17] GotoIf("SIP/1100203-00000002", "1?:lbl_DIAL-FROMIVR") in new stack [2012-06-06 10:52:57] WARNING[18354] ast_expr2.y: non-numeric argument [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:18] GotoIf("SIP/1100203-00000002", "1?:lbl_DIAL-FROMSHIP") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:19] GotoIf("SIP/1100203-00000002", "0?:lbl_DIAL-FROMSIP") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Goto (macro-stdexten,s,20) [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:20] Set("SIP/1100203-00000002", "GOSUB_RESULT=FROMSIP") in new stack [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:21] Progress("SIP/1100203-00000002", "") in new stack [2012-06-06 10:52:57] VERBOSE[18354] chan_sip.c: Audio is at 18178 [2012-06-06 10:52:57] VERBOSE[18354] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2012-06-06 10:52:57] VERBOSE[18354] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2012-06-06 10:52:57] VERBOSE[18354] chan_sip.c: <--- Transmitting (NAT) to 192.168.125.116:5062 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/TCP 192.168.125.116:5062;branch=z9hG4bK938bfc937287e8715;received=192.168.125.116;rport=5062 From: "1100203" ;tag=693ca13dce To: ;tag=as52322d33 Call-ID: a317c3d69eca9877 CSeq: 603883885 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 180;refresher=uas Contact: Content-Type: application/sdp Content-Length: 222 v=0 o=root 1471857749 1471857749 IN IP4 172.17.9.75 s=Asterisk c=IN IP4 172.17.9.75 t=0 0 m=audio 18178 RTP/AVP 0 125 a=rtpmap:0 PCMU/8000 a=rtpmap:125 telephone-event/8000 a=fmtp:125 0-16 a=ptime:20 a=sendrecv <------------> [2012-06-06 10:52:57] VERBOSE[18354] pbx.c: -- Executing [s@macro-stdexten:22] Dial("SIP/1100203-00000002", "SIP/10081,20,r") in new stack [2012-06-06 10:52:57] VERBOSE[18354] netsock2.c: == Using SIP RTP CoS mark 5 [2012-06-06 10:52:57] VERBOSE[18354] chan_sip.c: Audio is at 18786 [2012-06-06 10:52:57] VERBOSE[18354] chan_sip.c: Adding codec 0x2 (gsm) to SDP [2012-06-06 10:52:57] VERBOSE[18354] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2012-06-06 10:52:57] VERBOSE[18354] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.250.107:55163: INVITE sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK784a82f1 Max-Forwards: 70 From: "James Smith" ;tag=as7e97a879 To: Contact: Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 102 INVITE User-Agent: Asterisk Date: Wed, 06 Jun 2012 15:52:57 GMT Session-Expires: 180 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 219 v=0 o=root 115440865 115440865 IN IP4 172.17.9.75 s=Asterisk c=IN IP4 172.17.9.75 t=0 0 m=audio 18786 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2012-06-06 10:52:57] VERBOSE[18354] app_dial.c: -- Called SIP/10081 [2012-06-06 10:52:57] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK784a82f1 Contact: To: ;tag=872b0261 From: "James Smith";tag=as7e97a879 Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 102 INVITE User-Agent: 3CXPhone 6.0.20943.0 Content-Length: 0 <-------------> [2012-06-06 10:52:57] VERBOSE[18344] chan_sip.c: --- (9 headers 0 lines) --- [2012-06-06 10:52:57] VERBOSE[18344] chan_sip.c: list_route: hop: [2012-06-06 10:52:57] VERBOSE[18354] app_dial.c: -- SIP/10081-00000003 is ringing [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: <--- SIP read from TCP:192.168.125.116:5062 ---> REGISTER sip:172.17.9.75:5060 SIP/2.0 Accept: application/reginfo+xml, multipart/related Via: SIP/2.0/TCP 192.168.125.116:5062;branch=z9hG4bK8fc2266ba663b087a Max-Forwards: 70 From: "1100203" ;tag=40be9cfd79 To: "1100203" Call-ID: ab24a856abcb2313 CSeq: 40879079 REGISTER Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE Allow-Events: refer Authorization: Digest username="1100203",realm="asterisk",nonce="512555ae",uri="sip:172.17.9.75:5060",response="45a2973c765ef897ecac3f3d570605be",algorithm=MD5 Contact: ;expires=120 Supported: 100rel User-Agent: Media5-fone/2.14.1.194 Content-Length: 0 <-------------> [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: --- (15 headers 0 lines) --- [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: Sending to 192.168.125.116:5062 (NAT) [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: <--- Transmitting (NAT) to 192.168.125.116:5062 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.125.116:5062;branch=z9hG4bK8fc2266ba663b087a;received=192.168.125.116;rport=5062 From: "1100203" ;tag=40be9cfd79 To: "1100203" ;tag=as4af5fac3 Call-ID: ab24a856abcb2313 CSeq: 40879079 REGISTER Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0ff0686c" Content-Length: 0 <------------> [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: Scheduling destruction of SIP dialog 'ab24a856abcb2313' in 32000 ms (Method: REGISTER) [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: <--- SIP read from TCP:192.168.125.116:5062 ---> REGISTER sip:172.17.9.75:5060 SIP/2.0 Accept: application/reginfo+xml, multipart/related Via: SIP/2.0/TCP 192.168.125.116:5062;branch=z9hG4bK478bfac77b3b0ae26 Max-Forwards: 70 From: "1100203" ;tag=40be9cfd79 To: "1100203" Call-ID: ab24a856abcb2313 CSeq: 40879080 REGISTER Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE Allow-Events: refer Authorization: Digest username="1100203",realm="asterisk",nonce="0ff0686c",uri="sip:172.17.9.75:5060",response="9aa43f7aa2a07000be772b6ab543104c",algorithm=MD5 Contact: ;expires=120 Supported: 100rel User-Agent: Media5-fone/2.14.1.194 Content-Length: 0 <-------------> [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: --- (15 headers 0 lines) --- [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: Sending to 192.168.125.116:5062 (NAT) [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: Reliably Transmitting (NAT) to 192.168.125.116:5062: OPTIONS sip:1100203@192.168.125.116:5062 SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK56d9adc5;rport Max-Forwards: 70 From: "asterisk" ;tag=as5696564e To: Contact: Call-ID: 603006707459086b5e75badd1e0fff2b@172.17.9.75:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Date: Wed, 06 Jun 2012 15:52:58 GMT Session-Expires: 180 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: <--- Transmitting (NAT) to 192.168.125.116:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.125.116:5062;branch=z9hG4bK478bfac77b3b0ae26;received=192.168.125.116;rport=5062 From: "1100203" ;tag=40be9cfd79 To: "1100203" ;tag=as4af5fac3 Call-ID: ab24a856abcb2313 CSeq: 40879080 REGISTER Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Wed, 06 Jun 2012 15:52:58 GMT Content-Length: 0 <------------> [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: Scheduling destruction of SIP dialog 'ab24a856abcb2313' in 32000 ms (Method: REGISTER) [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: <--- SIP read from TCP:192.168.125.116:5062 ---> SIP/2.0 200 OK Accept: application/reginfo+xml, application/rlmi+xml, application/sdp, application/simple-message-summary, application/watcherinfo+xml, message/sipfrag, multipart/mixed Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK56d9adc5;rport=5060;received=172.17.9.75 From: "asterisk" ;tag=as5696564e To: ;tag=1215907228 Call-ID: 603006707459086b5e75badd1e0fff2b@172.17.9.75:5060 CSeq: 102 OPTIONS Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE Allow-Events: refer Contact: Server: Media5-fone/2.14.1.194 Supported: eventlist, replaces, timer Content-Length: 0 <-------------> [2012-06-06 10:52:58] VERBOSE[18326] chan_sip.c: --- (13 headers 0 lines) --- [2012-06-06 10:52:59] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog '603006707459086b5e75badd1e0fff2b@172.17.9.75:5060' Method: OPTIONS [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK784a82f1 Contact: To: ;tag=872b0261 From: "James Smith";tag=as7e97a879 Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 6.0.20943.0 Content-Length: 214 v=0 o=3cxVCE 403730880 272306010 IN IP4 192.168.250.107 s=3cxVCE Audio Call c=IN IP4 192.168.250.107 t=0 0 m=audio 40014 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: --- (12 headers 9 lines) --- [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: Found RTP audio format 3 [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: Found RTP audio format 101 [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: Found audio description format GSM for ID 3 [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: Found audio description format telephone-event for ID 101 [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: Peer audio RTP is at port 192.168.250.107:40014 [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: list_route: hop: [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: set_destination: Parsing for address/port to send to [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: set_destination: set destination to 192.168.250.107:55163 [2012-06-06 10:53:01] VERBOSE[18344] chan_sip.c: Transmitting (no NAT) to 192.168.250.107:55163: ACK sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK67d97cde Max-Forwards: 70 From: "James Smith" ;tag=as7e97a879 To: ;tag=872b0261 Contact: Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- [2012-06-06 10:53:01] VERBOSE[18354] app_dial.c: -- SIP/10081-00000003 answered SIP/1100203-00000002 [2012-06-06 10:53:01] VERBOSE[18354] chan_sip.c: Audio is at 18178 [2012-06-06 10:53:01] VERBOSE[18354] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2012-06-06 10:53:01] VERBOSE[18354] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2012-06-06 10:53:01] VERBOSE[18354] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.125.116:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.125.116:5062;branch=z9hG4bK938bfc937287e8715;received=192.168.125.116;rport=5062 From: "1100203" ;tag=693ca13dce To: ;tag=as52322d33 Call-ID: a317c3d69eca9877 CSeq: 603883885 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 180;refresher=uas Contact: Content-Type: application/sdp Content-Length: 222 v=0 o=root 1471857749 1471857750 IN IP4 172.17.9.75 s=Asterisk c=IN IP4 172.17.9.75 t=0 0 m=audio 18178 RTP/AVP 0 125 a=rtpmap:0 PCMU/8000 a=rtpmap:125 telephone-event/8000 a=fmtp:125 0-16 a=ptime:20 a=sendrecv <------------> [2012-06-06 10:53:01] VERBOSE[18326] chan_sip.c: <--- SIP read from TCP:192.168.125.116:5062 ---> ACK sip:10081@172.17.9.75:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 192.168.125.116:5062;branch=z9hG4bKc5347d2794f1b89fc Max-Forwards: 70 From: "1100203" ;tag=693ca13dce To: ;tag=as52322d33 Call-ID: a317c3d69eca9877 CSeq: 603883885 ACK Allow-Events: refer Authorization: Digest username="1100203",realm="asterisk",nonce="32c72d32",uri="sip:10081@172.17.9.75",response="3f765e7106887bcd5a4655f9ffee86fd",algorithm=MD5 Contact: User-Agent: Media5-fone/2.14.1.194 Content-Length: 0 <-------------> [2012-06-06 10:53:01] VERBOSE[18326] chan_sip.c: --- (12 headers 0 lines) --- [2012-06-06 10:53:14] VERBOSE[18315] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.250.107:55163: OPTIONS sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK304a8f21 Max-Forwards: 70 From: "asterisk" ;tag=as7ac4bb16 To: Contact: Call-ID: 200b5dd032433f865f017f7d79500073@172.17.9.75:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Date: Wed, 06 Jun 2012 15:53:14 GMT Session-Expires: 180 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2012-06-06 10:53:15] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK304a8f21 Contact: To: ;tag=ba2c2928 From: "asterisk";tag=as7ac4bb16 Call-ID: 200b5dd032433f865f017f7d79500073@172.17.9.75:5060 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 <-------------> [2012-06-06 10:53:15] VERBOSE[18344] chan_sip.c: --- (13 headers 0 lines) --- [2012-06-06 10:53:15] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog '200b5dd032433f865f017f7d79500073@172.17.9.75:5060' Method: OPTIONS [2012-06-06 10:53:30] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog 'ab24a856abcb2313' Method: REGISTER [2012-06-06 10:53:58] VERBOSE[18315] chan_sip.c: Reliably Transmitting (NAT) to 192.168.125.116:5062: OPTIONS sip:1100203@192.168.125.116:5062 SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK0bc7a682;rport Max-Forwards: 70 From: "asterisk" ;tag=as67d7af0f To: Contact: Call-ID: 424b63490eeaf2e57ab2d8c362495136@172.17.9.75:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Date: Wed, 06 Jun 2012 15:53:58 GMT Session-Expires: 180 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2012-06-06 10:54:02] NOTICE[18315] chan_sip.c: Peer '1100203' is now UNREACHABLE! Last qualify: 55 [2012-06-06 10:54:02] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog '424b63490eeaf2e57ab2d8c362495136@172.17.9.75:5060' Method: OPTIONS [2012-06-06 10:54:02] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> REGISTER sip:172.17.9.75:5060 SIP/2.0 Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-c65748188f030509-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Test Account" From: "Test Account";tag=ae076621 Call-ID: YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM. CSeq: 3 REGISTER Expires: 120 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Supported: replaces User-Agent: 3CXPhone 6.0.20943.0 Authorization: Digest username="10081",realm="asterisk",nonce="222bb238",uri="sip:172.17.9.75:5060",response="a8a7beb30277dcf3b37bc4c1e644b425",algorithm=MD5 Content-Length: 0 <-------------> [2012-06-06 10:54:02] VERBOSE[18344] chan_sip.c: --- (14 headers 0 lines) --- [2012-06-06 10:54:02] VERBOSE[18344] chan_sip.c: Sending to 192.168.250.107:55163 (NAT) [2012-06-06 10:54:02] VERBOSE[18344] chan_sip.c: <--- Transmitting (no NAT) to 192.168.250.107:55163 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-c65748188f030509-1---d8754z-;received=192.168.250.107;rport=55163 From: "Test Account";tag=ae076621 To: "Test Account";tag=as2abc2159 Call-ID: YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM. CSeq: 3 REGISTER Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1671504c" Content-Length: 0 <------------> [2012-06-06 10:54:02] VERBOSE[18344] chan_sip.c: Scheduling destruction of SIP dialog 'YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM.' in 32000 ms (Method: REGISTER) [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> REGISTER sip:172.17.9.75:5060 SIP/2.0 Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-693bce10af24c129-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Test Account" From: "Test Account";tag=ae076621 Call-ID: YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM. CSeq: 4 REGISTER Expires: 120 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Supported: replaces User-Agent: 3CXPhone 6.0.20943.0 Authorization: Digest username="10081",realm="asterisk",nonce="1671504c",uri="sip:172.17.9.75:5060",response="5a466c4a8eb867a9306f3c61ac60e777",algorithm=MD5 Content-Length: 0 <-------------> [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: --- (14 headers 0 lines) --- [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: Sending to 192.168.250.107:55163 (no NAT) [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.250.107:55163: OPTIONS sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK31f8472b Max-Forwards: 70 From: "asterisk" ;tag=as313058eb To: Contact: Call-ID: 41cc574873db3a803682efa922a37685@172.17.9.75:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Date: Wed, 06 Jun 2012 15:54:03 GMT Session-Expires: 180 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: <--- Transmitting (no NAT) to 192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-693bce10af24c129-1---d8754z-;received=192.168.250.107;rport=55163 From: "Test Account";tag=ae076621 To: "Test Account";tag=as2abc2159 Call-ID: YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM. CSeq: 4 REGISTER Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Wed, 06 Jun 2012 15:54:03 GMT Content-Length: 0 <------------> [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: Scheduling destruction of SIP dialog 'YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM.' in 32000 ms (Method: REGISTER) [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK31f8472b Contact: To: ;tag=60672729 From: "asterisk";tag=as313058eb Call-ID: 41cc574873db3a803682efa922a37685@172.17.9.75:5060 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 <-------------> [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: --- (13 headers 0 lines) --- [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SUBSCRIBE sip:asterisk@172.17.9.75:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-ed7f387bb933140d-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Test Account";tag=as6ce6c6b8 From: "Test Account";tag=e4488a14 Call-ID: YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA. CSeq: 4 SUBSCRIBE Expires: 120 User-Agent: 3CXPhone 6.0.20943.0 Authorization: Digest username="10081",realm="asterisk",nonce="63119892",uri="sip:asterisk@172.17.9.75:5060;transport=TCP",response="3978c5e2cb76f8a84900ca28948c18bb",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: --- (13 headers 0 lines) --- [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: Found peer '10081' for '10081' from 192.168.250.107:55163 [2012-06-06 10:54:03] NOTICE[18344] chan_sip.c: Correct auth, but based on stale nonce received from '"Test Account";tag=e4488a14' [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: <--- Transmitting (no NAT) to 192.168.250.107:55163 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-ed7f387bb933140d-1---d8754z-;received=192.168.250.107;rport=55163 From: "Test Account";tag=e4488a14 To: "Test Account";tag=as6ce6c6b8 Call-ID: YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA. CSeq: 4 SUBSCRIBE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="55f424db", stale=true Content-Length: 0 <------------> [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: Scheduling destruction of SIP dialog 'YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA.' in 16128 ms (Method: SUBSCRIBE) [2012-06-06 10:54:03] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog '41cc574873db3a803682efa922a37685@172.17.9.75:5060' Method: OPTIONS [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SUBSCRIBE sip:asterisk@172.17.9.75:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-9e6dad35c30e146e-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Test Account";tag=as6ce6c6b8 From: "Test Account";tag=e4488a14 Call-ID: YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA. CSeq: 5 SUBSCRIBE Expires: 120 User-Agent: 3CXPhone 6.0.20943.0 Authorization: Digest username="10081",realm="asterisk",nonce="55f424db",uri="sip:asterisk@172.17.9.75:5060;transport=TCP",response="2d0ab7cdf64143675acaae4bb2be29ce",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: --- (13 headers 0 lines) --- [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: Found peer '10081' for '10081' from 192.168.250.107:55163 [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: Scheduling destruction of SIP dialog 'YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA.' in 130000 ms (Method: SUBSCRIBE) [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: <--- Transmitting (no NAT) to 192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-9e6dad35c30e146e-1---d8754z-;received=192.168.250.107;rport=55163 From: "Test Account";tag=e4488a14 To: "Test Account";tag=as6ce6c6b8 Call-ID: YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA. CSeq: 5 SUBSCRIBE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Content-Length: 0 <------------> [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.250.107:63492: NOTIFY sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK71bfceca;rport Max-Forwards: 70 Route: From: "asterisk" ;tag=as6ce6c6b8 To: ;tag=e4488a14 Contact: Call-ID: YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA. CSeq: 103 NOTIFY User-Agent: Asterisk Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 105 Messages-Waiting: no Message-Account: sip:asterisk@172.17.9.75;transport=TCP Voice-Message: 0/0 (0/0) --- [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK71bfceca;rport=5060 Contact: To: ;tag=e4488a14 From: "asterisk";tag=as6ce6c6b8 Call-ID: YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA. CSeq: 103 NOTIFY User-Agent: 3CXPhone 6.0.20943.0 Content-Length: 0 <-------------> [2012-06-06 10:54:03] VERBOSE[18344] chan_sip.c: --- (9 headers 0 lines) --- [2012-06-06 10:54:12] VERBOSE[18315] chan_sip.c: Reliably Transmitting (NAT) to 192.168.125.116:5062: OPTIONS sip:1100203@192.168.125.116:5062 SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK3e59e54b;rport Max-Forwards: 70 From: "asterisk" ;tag=as366915a0 To: Contact: Call-ID: 6a7cd7997815fb1648eee2b05fa2dbdb@172.17.9.75:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Date: Wed, 06 Jun 2012 15:54:12 GMT Session-Expires: 180 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2012-06-06 10:54:16] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog '6a7cd7997815fb1648eee2b05fa2dbdb@172.17.9.75:5060' Method: OPTIONS [2012-06-06 10:54:26] VERBOSE[18315] chan_sip.c: Reliably Transmitting (NAT) to 192.168.125.116:5062: OPTIONS sip:1100203@192.168.125.116:5062 SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK49df6bf4;rport Max-Forwards: 70 From: "asterisk" ;tag=as280bfbd1 To: Contact: Call-ID: 3b373c780ff9f2f922da18e74cd18091@172.17.9.75:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Date: Wed, 06 Jun 2012 15:54:26 GMT Session-Expires: 180 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2012-06-06 10:54:27] VERBOSE[18315] chan_sip.c: set_destination: Parsing for address/port to send to [2012-06-06 10:54:27] VERBOSE[18315] chan_sip.c: set_destination: set destination to 192.168.125.116:5062 [2012-06-06 10:54:27] VERBOSE[18315] chan_sip.c: Audio is at 18178 [2012-06-06 10:54:27] VERBOSE[18315] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2012-06-06 10:54:27] VERBOSE[18315] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2012-06-06 10:54:27] VERBOSE[18315] chan_sip.c: Reliably Transmitting (NAT) to 192.168.125.116:5062: INVITE sip:1100203@192.168.125.116:5062 SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK7da7cff0;rport Max-Forwards: 70 From: ;tag=as52322d33 To: "1100203" ;tag=693ca13dce Contact: Call-ID: a317c3d69eca9877 CSeq: 102 INVITE User-Agent: Asterisk Session-Expires: 180;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 222 v=0 o=root 1471857749 1471857750 IN IP4 172.17.9.75 s=Asterisk c=IN IP4 172.17.9.75 t=0 0 m=audio 18178 RTP/AVP 0 125 a=rtpmap:0 PCMU/8000 a=rtpmap:125 telephone-event/8000 a=fmtp:125 0-16 a=ptime:20 a=sendrecv --- [2012-06-06 10:54:30] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog '3b373c780ff9f2f922da18e74cd18091@172.17.9.75:5060' Method: OPTIONS [2012-06-06 10:54:31] VERBOSE[18315] chan_sip.c: set_destination: Parsing for address/port to send to [2012-06-06 10:54:31] VERBOSE[18315] chan_sip.c: set_destination: set destination to 192.168.250.107:55163 [2012-06-06 10:54:31] VERBOSE[18315] chan_sip.c: Audio is at 18786 [2012-06-06 10:54:31] VERBOSE[18315] chan_sip.c: Adding codec 0x2 (gsm) to SDP [2012-06-06 10:54:31] VERBOSE[18315] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2012-06-06 10:54:31] VERBOSE[18315] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.250.107:55163: INVITE sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK4c4cae19 Max-Forwards: 70 From: "James Smith" ;tag=as7e97a879 To: ;tag=872b0261 Contact: Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 103 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 219 v=0 o=root 115440865 115440865 IN IP4 172.17.9.75 s=Asterisk c=IN IP4 172.17.9.75 t=0 0 m=audio 18786 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2012-06-06 10:54:32] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK4c4cae19 Contact: To: ;tag=872b0261 From: "James Smith";tag=as7e97a879 Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 6.0.20943.0 Content-Length: 214 v=0 o=3cxVCE 403730880 272306011 IN IP4 192.168.250.107 s=3cxVCE Audio Call c=IN IP4 192.168.250.107 t=0 0 m=audio 40014 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [2012-06-06 10:54:32] VERBOSE[18344] chan_sip.c: --- (12 headers 9 lines) --- [2012-06-06 10:54:32] VERBOSE[18344] chan_sip.c: Found RTP audio format 3 [2012-06-06 10:54:32] VERBOSE[18344] chan_sip.c: Found RTP audio format 101 [2012-06-06 10:54:32] VERBOSE[18344] chan_sip.c: Found audio description format GSM for ID 3 [2012-06-06 10:54:32] VERBOSE[18344] chan_sip.c: Found audio description format telephone-event for ID 101 [2012-06-06 10:54:32] VERBOSE[18344] chan_sip.c: Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) [2012-06-06 10:54:32] VERBOSE[18344] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2012-06-06 10:54:32] VERBOSE[18344] chan_sip.c: Peer audio RTP is at port 192.168.250.107:40014 [2012-06-06 10:54:32] VERBOSE[18344] chan_sip.c: set_destination: Parsing for address/port to send to [2012-06-06 10:54:32] VERBOSE[18344] chan_sip.c: set_destination: set destination to 192.168.250.107:55163 [2012-06-06 10:54:32] VERBOSE[18344] chan_sip.c: Transmitting (no NAT) to 192.168.250.107:55163: ACK sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK6f476bf8 Max-Forwards: 70 From: "James Smith" ;tag=as7e97a879 To: ;tag=872b0261 Contact: Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 103 ACK User-Agent: Asterisk Content-Length: 0 --- [2012-06-06 10:54:35] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog 'YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM.' Method: REGISTER [2012-06-06 10:54:40] VERBOSE[18315] chan_sip.c: Reliably Transmitting (NAT) to 192.168.125.116:5062: OPTIONS sip:1100203@192.168.125.116:5062 SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK0cf20961;rport Max-Forwards: 70 From: "asterisk" ;tag=as1324cf47 To: Contact: Call-ID: 5872244a606c6df94488a29b16c9ffbe@172.17.9.75:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Date: Wed, 06 Jun 2012 15:54:40 GMT Session-Expires: 180 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2012-06-06 10:54:44] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog '5872244a606c6df94488a29b16c9ffbe@172.17.9.75:5060' Method: OPTIONS [2012-06-06 10:54:54] VERBOSE[18315] chan_sip.c: Reliably Transmitting (NAT) to 192.168.125.116:5062: OPTIONS sip:1100203@192.168.125.116:5062 SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK7d7333aa;rport Max-Forwards: 70 From: "asterisk" ;tag=as0b4cfae2 To: Contact: Call-ID: 661b434a635b1c383e069e67376ed168@172.17.9.75:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Date: Wed, 06 Jun 2012 15:54:54 GMT Session-Expires: 180 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2012-06-06 10:54:54] WARNING[18315] chan_sip.c: sip_xmit of 0xa175bb8 (len 592) to 192.168.125.116:5062 returned -2: Interrupted system call [2012-06-06 10:54:58] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog '661b434a635b1c383e069e67376ed168@172.17.9.75:5060' Method: OPTIONS [2012-06-06 10:55:03] VERBOSE[18315] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.250.107:55163: OPTIONS sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK440ae658 Max-Forwards: 70 From: "asterisk" ;tag=as4bc81756 To: Contact: Call-ID: 18536f211b6c4135033715bd3c0b7c5c@172.17.9.75:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Date: Wed, 06 Jun 2012 15:55:03 GMT Session-Expires: 180 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2012-06-06 10:55:03] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK440ae658 Contact: To: ;tag=cc76fe49 From: "asterisk";tag=as4bc81756 Call-ID: 18536f211b6c4135033715bd3c0b7c5c@172.17.9.75:5060 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 <-------------> [2012-06-06 10:55:03] VERBOSE[18344] chan_sip.c: --- (13 headers 0 lines) --- [2012-06-06 10:55:04] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog '18536f211b6c4135033715bd3c0b7c5c@172.17.9.75:5060' Method: OPTIONS [2012-06-06 10:55:14] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> <-------------> [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> REGISTER sip:172.17.9.75:5060 SIP/2.0 Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-da2a082b2d6d993d-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Test Account" From: "Test Account";tag=ae076621 Call-ID: YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM. CSeq: 5 REGISTER Expires: 120 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Supported: replaces User-Agent: 3CXPhone 6.0.20943.0 Authorization: Digest username="10081",realm="asterisk",nonce="1671504c",uri="sip:172.17.9.75:5060",response="5a466c4a8eb867a9306f3c61ac60e777",algorithm=MD5 Content-Length: 0 <-------------> [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: --- (14 headers 0 lines) --- [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: Sending to 192.168.250.107:55163 (NAT) [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: <--- Transmitting (no NAT) to 192.168.250.107:55163 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-da2a082b2d6d993d-1---d8754z-;received=192.168.250.107;rport=55163 From: "Test Account";tag=ae076621 To: "Test Account";tag=as4218f5e0 Call-ID: YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM. CSeq: 5 REGISTER Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7cd45225" Content-Length: 0 <------------> [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: Scheduling destruction of SIP dialog 'YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM.' in 32000 ms (Method: REGISTER) [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> REGISTER sip:172.17.9.75:5060 SIP/2.0 Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-6c3aac0bfc5aa619-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Test Account" From: "Test Account";tag=ae076621 Call-ID: YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM. CSeq: 6 REGISTER Expires: 120 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Supported: replaces User-Agent: 3CXPhone 6.0.20943.0 Authorization: Digest username="10081",realm="asterisk",nonce="7cd45225",uri="sip:172.17.9.75:5060",response="da9a5293481567935126a4cda010e356",algorithm=MD5 Content-Length: 0 <-------------> [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: --- (14 headers 0 lines) --- [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: Sending to 192.168.250.107:55163 (no NAT) [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.250.107:55163: OPTIONS sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK105832c6 Max-Forwards: 70 From: "asterisk" ;tag=as7601ea1b To: Contact: Call-ID: 7841057a3fa8170870a91d7a2efb2d0f@172.17.9.75:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Date: Wed, 06 Jun 2012 15:55:51 GMT Session-Expires: 180 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: <--- Transmitting (no NAT) to 192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-6c3aac0bfc5aa619-1---d8754z-;received=192.168.250.107;rport=55163 From: "Test Account";tag=ae076621 To: "Test Account";tag=as4218f5e0 Call-ID: YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM. CSeq: 6 REGISTER Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Wed, 06 Jun 2012 15:55:51 GMT Content-Length: 0 <------------> [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: Scheduling destruction of SIP dialog 'YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM.' in 32000 ms (Method: REGISTER) [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK105832c6 Contact: To: ;tag=6b5c822e From: "asterisk";tag=as7601ea1b Call-ID: 7841057a3fa8170870a91d7a2efb2d0f@172.17.9.75:5060 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 <-------------> [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: --- (13 headers 0 lines) --- [2012-06-06 10:55:51] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog '7841057a3fa8170870a91d7a2efb2d0f@172.17.9.75:5060' Method: OPTIONS [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SUBSCRIBE sip:asterisk@172.17.9.75:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-e024896dca1e4574-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Test Account";tag=as6ce6c6b8 From: "Test Account";tag=e4488a14 Call-ID: YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA. CSeq: 6 SUBSCRIBE Expires: 120 User-Agent: 3CXPhone 6.0.20943.0 Authorization: Digest username="10081",realm="asterisk",nonce="55f424db",uri="sip:asterisk@172.17.9.75:5060;transport=TCP",response="2d0ab7cdf64143675acaae4bb2be29ce",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: --- (13 headers 0 lines) --- [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: Found peer '10081' for '10081' from 192.168.250.107:55163 [2012-06-06 10:55:51] NOTICE[18344] chan_sip.c: Correct auth, but based on stale nonce received from '"Test Account";tag=e4488a14' [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: <--- Transmitting (no NAT) to 192.168.250.107:55163 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-e024896dca1e4574-1---d8754z-;received=192.168.250.107;rport=55163 From: "Test Account";tag=e4488a14 To: "Test Account";tag=as6ce6c6b8 Call-ID: YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA. CSeq: 6 SUBSCRIBE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="365c9bc3", stale=true Content-Length: 0 <------------> [2012-06-06 10:55:51] VERBOSE[18344] chan_sip.c: Scheduling destruction of SIP dialog 'YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA.' in 15936 ms (Method: SUBSCRIBE) [2012-06-06 10:55:52] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SUBSCRIBE sip:asterisk@172.17.9.75:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-f2264f2a12312952-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Test Account";tag=as6ce6c6b8 From: "Test Account";tag=e4488a14 Call-ID: YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA. CSeq: 7 SUBSCRIBE Expires: 120 User-Agent: 3CXPhone 6.0.20943.0 Authorization: Digest username="10081",realm="asterisk",nonce="365c9bc3",uri="sip:asterisk@172.17.9.75:5060;transport=TCP",response="33674b1c6dc4312d31a793de76998070",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> [2012-06-06 10:55:52] VERBOSE[18344] chan_sip.c: --- (13 headers 0 lines) --- [2012-06-06 10:55:52] VERBOSE[18344] chan_sip.c: Found peer '10081' for '10081' from 192.168.250.107:55163 [2012-06-06 10:55:52] VERBOSE[18344] chan_sip.c: Scheduling destruction of SIP dialog 'YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA.' in 130000 ms (Method: SUBSCRIBE) [2012-06-06 10:55:52] VERBOSE[18344] chan_sip.c: <--- Transmitting (no NAT) to 192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.250.107:63492;branch=z9hG4bK-d8754z-f2264f2a12312952-1---d8754z-;received=192.168.250.107;rport=55163 From: "Test Account";tag=e4488a14 To: "Test Account";tag=as6ce6c6b8 Call-ID: YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA. CSeq: 7 SUBSCRIBE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Content-Length: 0 <------------> [2012-06-06 10:55:52] VERBOSE[18344] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.250.107:63492: NOTIFY sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK71bfceca;rport Max-Forwards: 70 Route: From: "asterisk" ;tag=as6ce6c6b8 To: ;tag=e4488a14 Contact: Call-ID: YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA. CSeq: 104 NOTIFY User-Agent: Asterisk Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 105 Messages-Waiting: no Message-Account: sip:asterisk@172.17.9.75;transport=TCP Voice-Message: 0/0 (0/0) --- [2012-06-06 10:55:52] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK71bfceca;rport=5060 Contact: To: ;tag=e4488a14 From: "asterisk";tag=as6ce6c6b8 Call-ID: YzkxNTNkYzE2YTFiMTY1NDdhOTg4ZTVhMzg5Y2QxYTA. CSeq: 104 NOTIFY User-Agent: 3CXPhone 6.0.20943.0 Content-Length: 0 <-------------> [2012-06-06 10:55:52] VERBOSE[18344] chan_sip.c: --- (9 headers 0 lines) --- [2012-06-06 10:55:57] VERBOSE[18315] chan_sip.c: set_destination: Parsing for address/port to send to [2012-06-06 10:55:57] VERBOSE[18315] chan_sip.c: set_destination: set destination to 192.168.125.116:5062 [2012-06-06 10:55:57] VERBOSE[18315] chan_sip.c: Audio is at 18178 [2012-06-06 10:55:57] VERBOSE[18315] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2012-06-06 10:55:57] VERBOSE[18315] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2012-06-06 10:55:57] VERBOSE[18315] chan_sip.c: Reliably Transmitting (NAT) to 192.168.125.116:5062: INVITE sip:1100203@192.168.125.116:5062 SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK2a994e4f;rport Max-Forwards: 70 From: ;tag=as52322d33 To: "1100203" ;tag=693ca13dce Contact: Call-ID: a317c3d69eca9877 CSeq: 103 INVITE User-Agent: Asterisk Session-Expires: 180;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 222 v=0 o=root 1471857749 1471857750 IN IP4 172.17.9.75 s=Asterisk c=IN IP4 172.17.9.75 t=0 0 m=audio 18178 RTP/AVP 0 125 a=rtpmap:0 PCMU/8000 a=rtpmap:125 telephone-event/8000 a=fmtp:125 0-16 a=ptime:20 a=sendrecv --- [2012-06-06 10:55:57] WARNING[18315] chan_sip.c: sip_xmit of 0xa2d9d90 (len 842) to 192.168.125.116:5062 returned -2: Interrupted system call [2012-06-06 10:56:01] VERBOSE[18315] chan_sip.c: set_destination: Parsing for address/port to send to [2012-06-06 10:56:01] VERBOSE[18315] chan_sip.c: set_destination: set destination to 192.168.250.107:55163 [2012-06-06 10:56:01] VERBOSE[18315] chan_sip.c: Audio is at 18786 [2012-06-06 10:56:01] VERBOSE[18315] chan_sip.c: Adding codec 0x2 (gsm) to SDP [2012-06-06 10:56:01] VERBOSE[18315] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2012-06-06 10:56:01] VERBOSE[18315] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.250.107:55163: INVITE sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK1085d2b9 Max-Forwards: 70 From: "James Smith" ;tag=as7e97a879 To: ;tag=872b0261 Contact: Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 104 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 219 v=0 o=root 115440865 115440865 IN IP4 172.17.9.75 s=Asterisk c=IN IP4 172.17.9.75 t=0 0 m=audio 18786 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2012-06-06 10:56:02] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK1085d2b9 Contact: To: ;tag=872b0261 From: "James Smith";tag=as7e97a879 Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 6.0.20943.0 Content-Length: 214 v=0 o=3cxVCE 403730880 272306012 IN IP4 192.168.250.107 s=3cxVCE Audio Call c=IN IP4 192.168.250.107 t=0 0 m=audio 40014 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [2012-06-06 10:56:02] VERBOSE[18344] chan_sip.c: --- (12 headers 9 lines) --- [2012-06-06 10:56:02] VERBOSE[18344] chan_sip.c: Found RTP audio format 3 [2012-06-06 10:56:02] VERBOSE[18344] chan_sip.c: Found RTP audio format 101 [2012-06-06 10:56:02] VERBOSE[18344] chan_sip.c: Found audio description format GSM for ID 3 [2012-06-06 10:56:02] VERBOSE[18344] chan_sip.c: Found audio description format telephone-event for ID 101 [2012-06-06 10:56:02] VERBOSE[18344] chan_sip.c: Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) [2012-06-06 10:56:02] VERBOSE[18344] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2012-06-06 10:56:02] VERBOSE[18344] chan_sip.c: Peer audio RTP is at port 192.168.250.107:40014 [2012-06-06 10:56:02] VERBOSE[18344] chan_sip.c: set_destination: Parsing for address/port to send to [2012-06-06 10:56:02] VERBOSE[18344] chan_sip.c: set_destination: set destination to 192.168.250.107:55163 [2012-06-06 10:56:02] VERBOSE[18344] chan_sip.c: Transmitting (no NAT) to 192.168.250.107:55163: ACK sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK1a22caab Max-Forwards: 70 From: "James Smith" ;tag=as7e97a879 To: ;tag=872b0261 Contact: Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 104 ACK User-Agent: Asterisk Content-Length: 0 --- [2012-06-06 10:56:23] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog 'YzVlNjgyYWY4NjMzODMyNTUzODhjNGEyOTJjYjIxYTM.' Method: REGISTER [2012-06-06 10:56:51] VERBOSE[18315] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.250.107:55163: OPTIONS sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK50942f15 Max-Forwards: 70 From: "asterisk" ;tag=as15185dcd To: Contact: Call-ID: 0dd8a7b76907fdd50c6695db057a44cc@172.17.9.75:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Date: Wed, 06 Jun 2012 15:56:51 GMT Session-Expires: 180 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2012-06-06 10:56:52] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK50942f15 Contact: To: ;tag=862c5c18 From: "asterisk";tag=as15185dcd Call-ID: 0dd8a7b76907fdd50c6695db057a44cc@172.17.9.75:5060 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 <-------------> [2012-06-06 10:56:52] VERBOSE[18344] chan_sip.c: --- (13 headers 0 lines) --- [2012-06-06 10:56:52] VERBOSE[18315] chan_sip.c: Really destroying SIP dialog '0dd8a7b76907fdd50c6695db057a44cc@172.17.9.75:5060' Method: OPTIONS [2012-06-06 10:57:27] VERBOSE[18315] chan_sip.c: set_destination: Parsing for address/port to send to [2012-06-06 10:57:27] VERBOSE[18315] chan_sip.c: set_destination: set destination to 192.168.125.116:5062 [2012-06-06 10:57:27] VERBOSE[18315] chan_sip.c: Audio is at 18178 [2012-06-06 10:57:27] VERBOSE[18315] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2012-06-06 10:57:27] VERBOSE[18315] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2012-06-06 10:57:27] VERBOSE[18315] chan_sip.c: Reliably Transmitting (NAT) to 192.168.125.116:5062: INVITE sip:1100203@192.168.125.116:5062 SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK5b914102;rport Max-Forwards: 70 From: ;tag=as52322d33 To: "1100203" ;tag=693ca13dce Contact: Call-ID: a317c3d69eca9877 CSeq: 104 INVITE User-Agent: Asterisk Session-Expires: 180;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 222 v=0 o=root 1471857749 1471857750 IN IP4 172.17.9.75 s=Asterisk c=IN IP4 172.17.9.75 t=0 0 m=audio 18178 RTP/AVP 0 125 a=rtpmap:0 PCMU/8000 a=rtpmap:125 telephone-event/8000 a=fmtp:125 0-16 a=ptime:20 a=sendrecv --- [2012-06-06 10:57:27] WARNING[18315] chan_sip.c: sip_xmit of 0xa2d9d90 (len 842) to 192.168.125.116:5062 returned -2: Interrupted system call [2012-06-06 10:57:29] VERBOSE[18337] asterisk.c: -- Remote UNIX connection disconnected [2012-06-06 10:57:31] VERBOSE[18315] chan_sip.c: set_destination: Parsing for address/port to send to [2012-06-06 10:57:31] VERBOSE[18315] chan_sip.c: set_destination: set destination to 192.168.250.107:55163 [2012-06-06 10:57:31] VERBOSE[18315] chan_sip.c: Audio is at 18786 [2012-06-06 10:57:31] VERBOSE[18315] chan_sip.c: Adding codec 0x2 (gsm) to SDP [2012-06-06 10:57:31] VERBOSE[18315] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2012-06-06 10:57:31] VERBOSE[18315] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.250.107:55163: INVITE sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK4d842d75 Max-Forwards: 70 From: "James Smith" ;tag=as7e97a879 To: ;tag=872b0261 Contact: Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 105 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 219 v=0 o=root 115440865 115440865 IN IP4 172.17.9.75 s=Asterisk c=IN IP4 172.17.9.75 t=0 0 m=audio 18786 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2012-06-06 10:57:32] VERBOSE[18344] chan_sip.c: <--- SIP read from TCP:192.168.250.107:55163 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK4d842d75 Contact: To: ;tag=872b0261 From: "James Smith";tag=as7e97a879 Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 105 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 6.0.20943.0 Content-Length: 214 v=0 o=3cxVCE 403730880 272306013 IN IP4 192.168.250.107 s=3cxVCE Audio Call c=IN IP4 192.168.250.107 t=0 0 m=audio 40014 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [2012-06-06 10:57:32] VERBOSE[18344] chan_sip.c: --- (12 headers 9 lines) --- [2012-06-06 10:57:32] VERBOSE[18344] chan_sip.c: Found RTP audio format 3 [2012-06-06 10:57:32] VERBOSE[18344] chan_sip.c: Found RTP audio format 101 [2012-06-06 10:57:32] VERBOSE[18344] chan_sip.c: Found audio description format GSM for ID 3 [2012-06-06 10:57:32] VERBOSE[18344] chan_sip.c: Found audio description format telephone-event for ID 101 [2012-06-06 10:57:32] VERBOSE[18344] chan_sip.c: Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) [2012-06-06 10:57:32] VERBOSE[18344] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2012-06-06 10:57:32] VERBOSE[18344] chan_sip.c: Peer audio RTP is at port 192.168.250.107:40014 [2012-06-06 10:57:32] VERBOSE[18344] chan_sip.c: set_destination: Parsing for address/port to send to [2012-06-06 10:57:32] VERBOSE[18344] chan_sip.c: set_destination: set destination to 192.168.250.107:55163 [2012-06-06 10:57:32] VERBOSE[18344] chan_sip.c: Transmitting (no NAT) to 192.168.250.107:55163: ACK sip:10081@192.168.250.107:55163;rinstance=1f7b70ac8ad7b018;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.17.9.75:5060;branch=z9hG4bK6ab32fe2 Max-Forwards: 70 From: "James Smith" ;tag=as7e97a879 To: ;tag=872b0261 Contact: Call-ID: 32b6042c540323832d2c28d47f76b18c@172.17.9.75:5060 CSeq: 105 ACK User-Agent: Asterisk Content-Length: 0 ---