[root@lemonade asterisk]# asterisk -r Asterisk 1.8.17.0, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.8.17.0 currently running on lemonade (pid = 5429) Verbosity is at least 3 lemonade*CLI> core set debug 9 Core debug was 0 and is now 9 lemonade*CLI> sip set debug on SIP Debugging enabled lemonade*CLI> core set verbose 9 Verbosity was 3 and is now 9 == ISDN1#30: Incoming call '3456' -> '5998' -- ISDN1#30: received Calling Party Name 83 'Ant Nikrooz' -- Executing [5998@isdn-in:1] Goto("CAPI/ISDN1#30/5998-0", "from-pstn,8885998,1") in new stack -- Goto (from-pstn,8885998,1) -- Executing [8885998@from-pstn:1] Set("CAPI/ISDN1#30/5998-0", "__FROM_DID=8885998") in new stack -- Executing [8885998@from-pstn:2] Gosub("CAPI/ISDN1#30/5998-0", "app-blacklist-check,s,1()") in new stack -- Executing [s@app-blacklist-check:1] GotoIf("CAPI/ISDN1#30/5998-0", "0?blacklisted") in new stack -- Executing [s@app-blacklist-check:2] Set("CAPI/ISDN1#30/5998-0", "CALLED_BLACKLIST=1") in new stack -- Executing [s@app-blacklist-check:3] Return("CAPI/ISDN1#30/5998-0", "") in new stack -- Executing [8885998@from-pstn:3] Set("CAPI/ISDN1#30/5998-0", "CDR(did)=8885998") in new stack -- Executing [8885998@from-pstn:4] ExecIf("CAPI/ISDN1#30/5998-0", "0 ?Set(CALLERID(name)=3456)") in new stack [2013-05-10 09:24:45] WARNING[5536]: func_callerid.c:817 callerpres_read: CALLERPRES is deprecated. Use CALLERID(name-pres) or CALLERID(num-pres) instead. -- Executing [8885998@from-pstn:5] Set("CAPI/ISDN1#30/5998-0", "__CALLINGPRES_SV=allowed_passed_screen") in new stack -- Executing [8885998@from-pstn:6] Set("CAPI/ISDN1#30/5998-0", "CALLERPRES()=allowed_not_screened") in new stack -- Executing [8885998@from-pstn:7] Goto("CAPI/ISDN1#30/5998-0", "ext-trunk,1,1") in new stack -- Goto (ext-trunk,1,1) -- Executing [1@ext-trunk:1] Set("CAPI/ISDN1#30/5998-0", "TDIAL_STRING=SIP/Lync") in new stack -- Executing [1@ext-trunk:2] Set("CAPI/ISDN1#30/5998-0", "DIAL_TRUNK=1") in new stack -- Executing [1@ext-trunk:3] Goto("CAPI/ISDN1#30/5998-0", "ext-trunk,tdial,1") in new stack -- Goto (ext-trunk,tdial,1) -- Executing [tdial@ext-trunk:1] Set("CAPI/ISDN1#30/5998-0", "OUTBOUND_GROUP=OUT_1") in new stack -- Executing [tdial@ext-trunk:2] GotoIf("CAPI/ISDN1#30/5998-0", "1?nomax") in new stack -- Goto (ext-trunk,tdial,4) -- Executing [tdial@ext-trunk:4] ExecIf("CAPI/ISDN1#30/5998-0", "1?Set(CALLERPRES()=allowed_passed_screen)") in new stack -- Executing [tdial@ext-trunk:5] Set("CAPI/ISDN1#30/5998-0", "DIAL_NUMBER=8885998") in new stack -- Executing [tdial@ext-trunk:6] GosubIf("CAPI/ISDN1#30/5998-0", "1?sub-flp-1,s,1()") in new stack -- Executing [s@sub-flp-1:1] ExecIf("CAPI/ISDN1#30/5998-0", "0?Set(TARGET_FLP_1=5998)") in new stack -- Executing [s@sub-flp-1:2] GotoIf("CAPI/ISDN1#30/5998-0", "0?match") in new stack -- Executing [s@sub-flp-1:3] ExecIf("CAPI/ISDN1#30/5998-0", "0?Set(TARGET_FLP_1=5998)") in new stack -- Executing [s@sub-flp-1:4] GotoIf("CAPI/ISDN1#30/5998-0", "0?match") in new stack -- Executing [s@sub-flp-1:5] ExecIf("CAPI/ISDN1#30/5998-0", "1?Set(TARGET_FLP_1=5998)") in new stack -- Executing [s@sub-flp-1:6] GotoIf("CAPI/ISDN1#30/5998-0", "1?match") in new stack -- Goto (sub-flp-1,s,8) -- Executing [s@sub-flp-1:8] Set("CAPI/ISDN1#30/5998-0", "DIAL_NUMBER=5998") in new stack -- Executing [s@sub-flp-1:9] Return("CAPI/ISDN1#30/5998-0", "") in new stack -- Executing [tdial@ext-trunk:7] Set("CAPI/ISDN1#30/5998-0", "OUTNUM=5998") in new stack -- Executing [tdial@ext-trunk:8] Dial("CAPI/ISDN1#30/5998-0", "SIP/Lync/5998,300,") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 18652 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 100.100.100.19:5068: INVITE sip:5998@100.100.100.19:5068 SIP/2.0 Via: SIP/2.0/TCP 192.168.66.67:5060;branch=z9hG4bK340b5616;rport Max-Forwards: 70 From: "Ant Nikrooz" ;tag=as6553c3e9 To: Contact: Call-ID: 06b2493b76bd5d81779981400fdc87a9@192.168.66.67:5060 CSeq: 102 INVITE User-Agent: FPBX-2.10.0(1.8.17.0) Date: Fri, 10 May 2013 08:24:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "Ant Nikrooz" ;party=calling;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 288 v=0 o=root 2028465138 2028465138 IN IP4 192.168.66.67 s=Asterisk PBX 1.8.17.0 c=IN IP4 192.168.66.67 t=0 0 m=audio 18652 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called SIP/Lync/5998 <--- SIP read from TCP:100.100.100.19:5068 ---> SIP/2.0 100 Trying FROM: "Ant Nikrooz";tag=as6553c3e9 TO: CSEQ: 102 INVITE CALL-ID: 06b2493b76bd5d81779981400fdc87a9@192.168.66.67:5060 VIA: SIP/2.0/TCP 192.168.66.67:5060;branch=z9hG4bK340b5616;rport CONTENT-LENGTH: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from TCP:100.100.100.19:5068 ---> SIP/2.0 183 Session Progress FROM: "Ant Nikrooz";tag=as6553c3e9 TO: ;tag=cac0f86c9f;epid=781A35F9DD CSEQ: 102 INVITE CALL-ID: 06b2493b76bd5d81779981400fdc87a9@192.168.66.67:5060 VIA: SIP/2.0/TCP 192.168.66.67:5060;branch=z9hG4bK340b5616;rport CONTACT: CONTENT-LENGTH: 0 ALLOW: CANCEL ALLOW: BYE ALLOW: UPDATE ALLOW: PRACK SERVER: RTCC/5.0.0.0 MediationServer <-------------> --- (13 headers 0 lines) --- list_route: hop: -- SIP/Lync-00000000 is ringing <--- SIP read from TCP:100.100.100.19:5068 ---> SIP/2.0 180 Ringing FROM: "Ant Nikrooz";tag=as6553c3e9 TO: ;tag=cac0f86c9f;epid=781A35F9DD CSEQ: 102 INVITE CALL-ID: 06b2493b76bd5d81779981400fdc87a9@192.168.66.67:5060 VIA: SIP/2.0/TCP 192.168.66.67:5060;branch=z9hG4bK340b5616;rport CONTACT: CONTENT-LENGTH: 0 ALLOW: CANCEL ALLOW: BYE ALLOW: UPDATE ALLOW: PRACK SERVER: RTCC/5.0.0.0 MediationServer <-------------> --- (13 headers 0 lines) --- list_route: hop: -- SIP/Lync-00000000 is ringing <--- SIP read from TCP:100.100.100.19:5068 ---> SIP/2.0 200 OK FROM: "Ant Nikrooz";tag=as6553c3e9 TO: ;tag=cac0f86c9f;epid=781A35F9DD CSEQ: 102 INVITE CALL-ID: 06b2493b76bd5d81779981400fdc87a9@192.168.66.67:5060 VIA: SIP/2.0/TCP 192.168.66.67:5060;branch=z9hG4bK340b5616;rport CONTACT: CONTENT-LENGTH: 255 SUPPORTED: 100rel CONTENT-TYPE: application/sdp ALLOW: ACK SERVER: RTCC/5.0.0.0 MediationServer Allow: CANCEL,BYE,INVITE,PRACK,UPDATE v=0 o=- 131 1 IN IP4 100.100.100.19 s=session c=IN IP4 100.100.100.19 b=CT:1000 t=0 0 m=audio 56834 RTP/AVP 8 101 c=IN IP4 100.100.100.19 a=rtcp:56835 a=label:Audio a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (13 headers 14 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 100.100.100.19:56834 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 100.100.100.19:5068 Transmitting (NAT) to 100.100.100.19:5068: ACK sip:bacardi.bolton.ac.uk:5068;transport=Tcp;maddr=100.100.100.19 SIP/2.0 Via: SIP/2.0/TCP 192.168.66.67:5060;branch=z9hG4bK089c544f;rport Max-Forwards: 70 From: "Ant Nikrooz" ;tag=as6553c3e9 To: ;tag=cac0f86c9f Contact: Call-ID: 06b2493b76bd5d81779981400fdc87a9@192.168.66.67:5060 CSeq: 102 ACK User-Agent: FPBX-2.10.0(1.8.17.0) Content-Length: 0 --- -- SIP/Lync-00000000 answered CAPI/ISDN1#30/5998-0 == ISDN1#30: Answering for 5998 <--- SIP read from TCP:100.100.100.19:5068 ---> INVITE sip:3456@192.168.66.67:5060;transport=TCP SIP/2.0 FROM: ;epid=781A35F9DD;tag=cac0f86c9f TO: ;tag=as6553c3e9 CSEQ: 1 INVITE CALL-ID: 06b2493b76bd5d81779981400fdc87a9@192.168.66.67:5060 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 100.100.100.19:5068;branch=z9hG4bK9f691ad3 CONTACT: CONTENT-LENGTH: 0 SUPPORTED: 100rel USER-AGENT: RTCC/5.0.0.0 MediationServer CONTENT-TYPE: application/sdp <-------------> --- (12 headers 0 lines) --- Sending to 100.100.100.19:5068 (NAT) <--- Transmitting (NAT) to 100.100.100.19:5068 ---> SIP/2.0 100 Trying Via: SIP/2.0/TCP 100.100.100.19:5068;branch=z9hG4bK9f691ad3;received=100.100.100.19;rport=5068 From: ;epid=781A35F9DD;tag=cac0f86c9f To: ;tag=as6553c3e9 Call-ID: 06b2493b76bd5d81779981400fdc87a9@192.168.66.67:5060 CSeq: 1 INVITE Server: FPBX-2.10.0(1.8.17.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 18652 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 100.100.100.19:5068 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 100.100.100.19:5068;branch=z9hG4bK9f691ad3;received=100.100.100.19;rport=5068 From: ;epid=781A35F9DD;tag=cac0f86c9f To: ;tag=as6553c3e9 Call-ID: 06b2493b76bd5d81779981400fdc87a9@192.168.66.67:5060 CSeq: 1 INVITE Server: FPBX-2.10.0(1.8.17.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 288 v=0 o=root 2028465138 2028465139 IN IP4 192.168.66.67 s=Asterisk PBX 1.8.17.0 c=IN IP4 192.168.66.67 t=0 0 m=audio 18652 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from TCP:100.100.100.19:49333 ---> INVITE sip:3436@192.168.66.67;user=phone SIP/2.0 FROM: "Ant Nikrooz";epid=781A35F9DD;tag=e73cdd2af9 TO: CSEQ: 12499 INVITE CALL-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 100.100.100.19:49333;branch=z9hG4bK387a5a9d CONTACT: CONTENT-LENGTH: 313 REFERRED-BY: SUPPORTED: 100rel USER-AGENT: RTCC/5.0.0.0 MediationServer CONTENT-TYPE: application/sdp ALLOW: ACK P-ASSERTED-IDENTITY: "Ant Nikrooz" Privacy: id Allow: CANCEL,BYE,INVITE,PRACK,UPDATE v=0 o=root 2028465138 2028465139 IN IP4 192.168.66.67 s=session c=IN IP4 192.168.66.67 t=0 0 m=audio 18652 RTP/AVP 8 0 3 101 c=IN IP4 192.168.66.67 a=rtcp:18653 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (17 headers 15 lines) --- Sending to 100.100.100.19:49333 (NAT) Using INVITE request as basis request - 6cd8d158-78b2-49ac-b9dc-40e7406875dd Found peer 'from-Lync' for '3456;phone-context=enterprise' from 100.100.100.19:49333 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.66.67:18652 Looking for 3436 in from-lync (domain 192.168.66.67) list_route: hop: <--- Transmitting (NAT) to 100.100.100.19:49333 ---> SIP/2.0 100 Trying Via: SIP/2.0/TCP 100.100.100.19:49333;branch=z9hG4bK387a5a9d;received=100.100.100.19;rport=49333 From: "Ant Nikrooz";epid=781A35F9DD;tag=e73cdd2af9 To: Call-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd CSeq: 12499 INVITE Server: FPBX-2.10.0(1.8.17.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [3436@from-lync:1] Goto("SIP/from-Lync-00000001", "from-pstn,7773436,1") in new stack -- Goto (from-pstn,7773436,1) -- Executing [7773436@from-pstn:1] Set("SIP/from-Lync-00000001", "__FROM_DID=7773436") in new stack -- Executing [7773436@from-pstn:2] Gosub("SIP/from-Lync-00000001", "app-blacklist-check,s,1()") in new stack -- Executing [s@app-blacklist-check:1] GotoIf("SIP/from-Lync-00000001", "0?blacklisted") in new stack -- Executing [s@app-blacklist-check:2] Set("SIP/from-Lync-00000001", "CALLED_BLACKLIST=1") in new stack -- Executing [s@app-blacklist-check:3] Return("SIP/from-Lync-00000001", "") in new stack -- Executing [7773436@from-pstn:3] Set("SIP/from-Lync-00000001", "CDR(did)=7773436") in new stack -- Executing [7773436@from-pstn:4] ExecIf("SIP/from-Lync-00000001", "0 ?Set(CALLERID(name)=3456)") in new stack -- Executing [7773436@from-pstn:5] Set("SIP/from-Lync-00000001", "__CALLINGPRES_SV=allowed_not_screened") in new stack -- Executing [7773436@from-pstn:6] Set("SIP/from-Lync-00000001", "CALLERPRES()=allowed_not_screened") in new stack -- Executing [7773436@from-pstn:7] Goto("SIP/from-Lync-00000001", "ext-trunk,2,1") in new stack -- Goto (ext-trunk,2,1) -- Executing [2@ext-trunk:1] Set("SIP/from-Lync-00000001", "SS=$") in new stack -- Executing [2@ext-trunk:2] Set("SIP/from-Lync-00000001", "TDIAL_STRING=CAPI/ISDN1/${OUTNUM}") in new stack -- Executing [2@ext-trunk:3] Set("SIP/from-Lync-00000001", "DIAL_TRUNK=2") in new stack -- Executing [2@ext-trunk:4] Goto("SIP/from-Lync-00000001", "ext-trunk,tcustom,1") in new stack -- Goto (ext-trunk,tcustom,1) -- Executing [tcustom@ext-trunk:1] Set("SIP/from-Lync-00000001", "OUTBOUND_GROUP=OUT_2") in new stack -- Executing [tcustom@ext-trunk:2] GotoIf("SIP/from-Lync-00000001", "1?nomax") in new stack -- Goto (ext-trunk,tcustom,4) -- Executing [tcustom@ext-trunk:4] ExecIf("SIP/from-Lync-00000001", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack -- Executing [tcustom@ext-trunk:5] Set("SIP/from-Lync-00000001", "DIAL_NUMBER=7773436") in new stack -- Executing [tcustom@ext-trunk:6] GosubIf("SIP/from-Lync-00000001", "1?sub-flp-2,s,1()") in new stack -- Executing [s@sub-flp-2:1] ExecIf("SIP/from-Lync-00000001", "1?Set(TARGET_FLP_2=3436)") in new stack -- Executing [s@sub-flp-2:2] GotoIf("SIP/from-Lync-00000001", "1?match") in new stack -- Goto (sub-flp-2,s,10) -- Executing [s@sub-flp-2:10] Set("SIP/from-Lync-00000001", "DIAL_NUMBER=3436") in new stack -- Executing [s@sub-flp-2:11] Return("SIP/from-Lync-00000001", "") in new stack -- Executing [tcustom@ext-trunk:7] Set("SIP/from-Lync-00000001", "OUTNUM=3436") in new stack -- Executing [tcustom@ext-trunk:8] Set("SIP/from-Lync-00000001", "CALLERID(number)=3456") in new stack -- Executing [tcustom@ext-trunk:9] Set("SIP/from-Lync-00000001", "CALLERID(name)=Ant Nikrooz") in new stack -- Executing [tcustom@ext-trunk:10] Dial("SIP/from-Lync-00000001", "CAPI/ISDN1/3436,300,") in new stack -- ISDN1#29: * Sending CALLED/CONNECTED NAME 80 'Ant Nikrooz' -- Called CAPI/ISDN1/3436 -- CAPI/ISDN1#29/3436-1 is proceeding passing it to SIP/from-Lync-00000001 <--- Transmitting (NAT) to 100.100.100.19:49333 ---> SIP/2.0 100 Trying Via: SIP/2.0/TCP 100.100.100.19:49333;branch=z9hG4bK387a5a9d;received=100.100.100.19;rport=49333 From: "Ant Nikrooz";epid=781A35F9DD;tag=e73cdd2af9 To: Call-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd CSeq: 12499 INVITE Server: FPBX-2.10.0(1.8.17.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- ISDN1#29: received Called Party Name 83 'Ant DECT' -- ast_channel_queue_connected_line_update( aG, °Q·, ) -- CAPI/ISDN1#29/3436-1 is making progress passing it to SIP/from-Lync-00000001 Audio is at 15914 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 100.100.100.19:49333 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/TCP 100.100.100.19:49333;branch=z9hG4bK387a5a9d;received=100.100.100.19;rport=49333 From: "Ant Nikrooz";epid=781A35F9DD;tag=e73cdd2af9 To: ;tag=as1397fb66 Call-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd CSeq: 12499 INVITE Server: FPBX-2.10.0(1.8.17.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 286 v=0 o=root 576859798 576859798 IN IP4 192.168.66.67 s=Asterisk PBX 1.8.17.0 c=IN IP4 192.168.66.67 t=0 0 m=audio 15914 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=ptime:20 a=sendrecv <------------> -- CAPI/ISDN1#29/3436-1 is ringing <--- Transmitting (NAT) to 100.100.100.19:49333 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TCP 100.100.100.19:49333;branch=z9hG4bK387a5a9d;received=100.100.100.19;rport=49333 From: "Ant Nikrooz";epid=781A35F9DD;tag=e73cdd2af9 To: ;tag=as1397fb66 Call-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd CSeq: 12499 INVITE Server: FPBX-2.10.0(1.8.17.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '06b2493b76bd5d81779981400fdc87a9@192.168.66.67:5060' in 6400 ms (Method: INVITE) == Spawn extension (ext-trunk, tdial, 8) exited non-zero on 'CAPI/ISDN1#30/5998-0' == ISDN1#30: CAPI Hangingup for PLCI=0x201 in state 2 > ISDN1#30: CAPI INFO 0x3490: Normal call clearing -- ISDN1#29: received Connected Party Name 83 'Ant DECT' -- ast_channel_queue_connected_line_update( aG, °Q·, ) -- CAPI/ISDN1#29/3436-1 answered SIP/from-Lync-00000001 Audio is at 15914 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 100.100.100.19:49333 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 100.100.100.19:49333;branch=z9hG4bK387a5a9d;received=100.100.100.19;rport=49333 From: "Ant Nikrooz";epid=781A35F9DD;tag=e73cdd2af9 To: ;tag=as1397fb66 Call-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd CSeq: 12499 INVITE Server: FPBX-2.10.0(1.8.17.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 286 v=0 o=root 576859798 576859799 IN IP4 192.168.66.67 s=Asterisk PBX 1.8.17.0 c=IN IP4 192.168.66.67 t=0 0 m=audio 15914 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=ptime:20 a=sendrecv <------------> <--- SIP read from TCP:100.100.100.19:49333 ---> ACK sip:3436@192.168.66.67:5060;transport=TCP SIP/2.0 FROM: ;epid=781A35F9DD;tag=e73cdd2af9 TO: ;tag=as1397fb66 CSEQ: 12499 ACK CALL-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 100.100.100.19:49333;branch=z9hG4bK609aee71 CONTENT-LENGTH: 0 USER-AGENT: RTCC/5.0.0.0 MediationServer <-------------> --- (9 headers 0 lines) --- <--- SIP read from TCP:100.100.100.19:5068 ---> ACK sip:3456@192.168.66.67:5060;transport=TCP SIP/2.0 FROM: ;epid=781A35F9DD;tag=cac0f86c9f TO: ;tag=as6553c3e9 CSEQ: 1 ACK CALL-ID: 06b2493b76bd5d81779981400fdc87a9@192.168.66.67:5060 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 100.100.100.19:5068;branch=z9hG4bK5e942b70 CONTENT-LENGTH: 311 USER-AGENT: RTCC/5.0.0.0 MediationServer CONTENT-TYPE: application/sdp v=0 o=root 576859798 576859799 IN IP4 192.168.66.67 s=session c=IN IP4 192.168.66.67 t=0 0 m=audio 15914 RTP/AVP 0 8 3 101 c=IN IP4 192.168.66.67 a=rtcp:15915 a=sendrecv 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=ptime:20 <-------------> --- (10 headers 15 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 - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.66.67:15914 [2013-05-10 09:24:53] ERROR[5502]: astobj2.c:115 INTERNAL_OBJ: user_data is NULL lemonade*CLI> Disconnected from Asterisk server [root@lemonade asterisk]# /usr/sbin/safe_asterisk: line 145: 5485 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY} Asterisk ended with exit status 139 Asterisk exited on signal 11. Automatically restarting Asterisk.