<--- SIP read from UDP:10.11.2.47:5060 ---> INVITE sip:107@10.11.2.47 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK357.81bdcce5.0 Via: SIP/2.0/UDP 10.11.2.37:5060;rport=5060;branch=z9hG4bK4158 From: ;tag=29997 To: "107" Call-ID: 18342 CSeq: 21 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 69 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Subject: Phone call Content-Length: 349 v=0 o=106 976 976 IN IP4 10.11.2.37 s=Talk c=IN IP4 10.11.2.37 t=0 0 m=audio 7078 RTP/AVP 110 3 0 8 101 a=rtpmap:110 speex/8000 a=fmtp:110 vbr=on a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 9078 RTP/AVP 98 a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 <-------------> --- (15 headers 16 lines) --- Sending to 10.11.2.47:5060 (NAT) Using INVITE request as basis request - 18342 Found peer '106' for '106' from 10.11.2.47:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 110 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format speex for ID 110 Found audio description format GSM for ID 3 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 Found RTP video format 98 Found video description format H263-1998 for ID 98 Capabilities: us - (gsm|ulaw|alaw|h263|h263p|h264|testlaw), peer - audio=(gsm|ulaw|alaw|speex)/video=(h263p)/text=(nothing), combined - (gsm|ulaw|alaw|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.11.2.37:7078 Peer video RTP is at port 10.11.2.37:9078 Looking for 107 in from-sip (domain 10.11.2.47) list_route: hop: <--- Transmitting (no NAT) to 10.11.2.47:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK357.81bdcce5.0;received=10.11.2.47 Via: SIP/2.0/UDP 10.11.2.37:5060;rport=5060;branch=z9hG4bK4158 Record-Route: From: ;tag=29997 To: "107" Call-ID: 18342 CSeq: 21 INVITE Server: Asterisk PBX 10.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [107@from-sip:1] Dial("SIP/106-0000001d", "SIP/107,60") in new stack == Using SIP RTP CoS mark 5 Audio is at 15732 Video is at 10.11.2.47:16908 Adding codec 100003 (ulaw) to SDP Adding video codec 200002 (h263) to SDP Adding video codec 200003 (h263p) to SDP Adding video codec 200004 (h264) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100004 (alaw) to SDP Adding codec 100017 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.11.2.47:5060: INVITE sip:107@10.11.2.47:5060 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440 Max-Forwards: 70 From: "106" ;tag=as149287b0 To: Contact: Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 102 INVITE User-Agent: Asterisk PBX 10.7.0 Date: Wed, 08 May 2013 09:09:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 434 v=0 o=root 476348733 476348733 IN IP4 10.11.2.47 s=Asterisk PBX 10.7.0 c=IN IP4 10.11.2.47 b=CT:384 t=0 0 m=audio 15732 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 16908 RTP/AVP 34 98 99 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv --- <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440 From: "106" ;tag=as149287b0 To: Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 102 INVITE Server: kamailio (3.3.1 (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- Called SIP/107 <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 101 Dialog Establishement Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440 Record-Route: From: "106" ;tag=as149287b0 To: ;tag=1777248976 Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 102 INVITE Contact: User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440 Record-Route: From: "106" ;tag=as149287b0 To: ;tag=1777248976 Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 102 INVITE Contact: User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- list_route: hop: -- SIP/107-0000001e is ringing <--- Transmitting (no NAT) to 10.11.2.47:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK357.81bdcce5.0;received=10.11.2.47 Via: SIP/2.0/UDP 10.11.2.37:5060;rport=5060;branch=z9hG4bK4158 Record-Route: From: ;tag=29997 To: "107" ;tag=as6e6ec5ef Call-ID: 18342 CSeq: 21 INVITE Server: Asterisk PBX 10.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> <--- SIP read from UDP:10.11.2.47:5060 ---> REGISTER sip:10.11.2.47:5080 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK9f36.d8026017.0 To: sip:106@10.11.2.47 From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-972d CSeq: 10 REGISTER Call-ID: 5eefa7c2-4330@127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.3.1 (i386/linux)) Contact: Expires: 60 <-------------> --- (10 headers 0 lines) --- Sending to 10.11.2.47:5060 (NAT) <--- Transmitting (no NAT) to 10.11.2.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK9f36.d8026017.0;received=10.11.2.47 From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-972d To: sip:106@10.11.2.47;tag=as282227a3 Call-ID: 5eefa7c2-4330@127.0.0.1 CSeq: 10 REGISTER Server: Asterisk PBX 10.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Wed, 08 May 2013 09:09:22 GMT Content-Length: 0 <------------> [May 8 17:09:22] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1 Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't exist (100) [May 8 17:09:22] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk]... [May 8 17:09:22] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check: Connection is down attempting to reconnect... [May 8 17:09:27] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting asterisk [May 8 17:09:27] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk] [May 8 17:09:27] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1 Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't exist (100) [May 8 17:09:27] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk]... [May 8 17:09:27] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check: Connection is down attempting to reconnect... [May 8 17:09:32] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting asterisk [May 8 17:09:32] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk] [May 8 17:09:32] WARNING[4068]: app_voicemail.c:5142 inboxcount2: SQL Execute error! [SELECT COUNT(*) FROM voicemessages WHERE dir = '/var/spool/asterisk/voicemail/default/106/INBOX'] Scheduling destruction of SIP dialog '2b27053f3491a2371bef4b0c12ff7754@10.11.2.47' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 10.11.2.47:5060: NOTIFY sip:106@10.11.2.47:5060 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK5c0ba38d Max-Forwards: 70 From: "asterisk" ;tag=as063ef50c To: Contact: Call-ID: 2b27053f3491a2371bef4b0c12ff7754@10.11.2.47 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 10.7.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 90 Messages-Waiting: no Message-Account: sip:asterisk@10.11.2.47 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog '5eefa7c2-4330@127.0.0.1' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440 Record-Route: From: "106" ;tag=as149287b0 To: ;tag=1777248976 Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 342 v=0 o=107 3544 3544 IN IP4 10.11.2.50 s=Talk c=IN IP4 10.11.2.50 t=0 0 m=audio 7078 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 9078 RTP/AVP 98 99 a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014 <-------------> --- (11 headers 15 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 Found RTP video format 98 Found RTP video format 99 Found video description format H263-1998 for ID 98 Found video description format H264 for ID 99 Capabilities: us - (gsm|ulaw|alaw|h263|h263p|h264|testlaw), peer - audio=(ulaw|alaw)/video=(h263p|h264)/text=(nothing), combined - (ulaw|alaw|h263p|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.11.2.50:7078 Peer video RTP is at port 10.11.2.50:9078 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.11.2.47:5060 Transmitting (no NAT) to 10.11.2.47:5060: ACK sip:107@10.11.2.50 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK78a9184a Route: Max-Forwards: 70 From: "106" ;tag=as149287b0 To: ;tag=1777248976 Contact: Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 102 ACK User-Agent: Asterisk PBX 10.7.0 Content-Length: 0 --- <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440 Record-Route: From: "106" ;tag=as149287b0 To: ;tag=1777248976 Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 342 v=0 o=107 3544 3544 IN IP4 10.11.2.50 s=Talk c=IN IP4 10.11.2.50 t=0 0 m=audio 7078 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 9078 RTP/AVP 98 99 a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014 <-------------> -- SIP/107-0000001e answered SIP/106-0000001d Audio is at 16686 Video is at 10.11.2.47:16818 Adding video codec 200003 (h263p) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.11.2.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK357.81bdcce5.0;received=10.11.2.47 Via: SIP/2.0/UDP 10.11.2.37:5060;rport=5060;branch=z9hG4bK4158 Record-Route: From: ;tag=29997 To: "107" ;tag=as6e6ec5ef Call-ID: 18342 CSeq: 21 INVITE Server: Asterisk PBX 10.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 380 v=0 o=root 742615961 742615961 IN IP4 10.11.2.47 s=Asterisk PBX 10.7.0 c=IN IP4 10.11.2.47 b=CT:384 t=0 0 m=audio 16686 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 16818 RTP/AVP 98 a=rtpmap:98 h263-1998/90000 a=sendrecv <------------> -- Remotely bridging SIP/106-0000001d and SIP/107-0000001e set_destination: Parsing for address/port to send to set_destination: set destination to 10.11.2.47:5060 Audio is at 15732 --- (11 headers 15 lines) --- Video is at 10.11.2.37:9078 Adding codec 100003 (ulaw) to SDP Adding video codec 200003 (h263p) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.11.2.47:5060: INVITE sip:107@10.11.2.50 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK49de740f Route: Max-Forwards: 70 From: "106" ;tag=as149287b0 To: ;tag=1777248976 Contact: Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 103 INVITE User-Agent: Asterisk PBX 10.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 355 v=0 o=root 476348733 476348734 IN IP4 10.11.2.37 s=Asterisk PBX 10.7.0 c=IN IP4 10.11.2.37 b=CT:384 t=0 0 m=audio 7078 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 9078 RTP/AVP 98 a=rtpmap:98 h263-1998/90000 a=sendrecv --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.11.2.47:5060 Transmitting (no NAT) to 10.11.2.47:5060: ACK sip:107@10.11.2.50 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK2c4b9ccb Route: Max-Forwards: 70 From: "106" ;tag=as149287b0 To: ;tag=1777248976 Contact: Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 102 ACK User-Agent: Asterisk PBX 10.7.0 Content-Length: 0 --- <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440 Record-Route: From: "106" ;tag=as149287b0 To: ;tag=1777248976 Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 342 v=0 o=107 3544 3544 IN IP4 10.11.2.50 s=Talk c=IN IP4 10.11.2.50 t=0 0 m=audio 7078 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 9078 RTP/AVP 98 99 a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014 <-------------> --- (11 headers 15 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.11.2.47:5060 Transmitting (no NAT) to 10.11.2.47:5060: ACK sip:107@10.11.2.50 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK14e1df5b Route: Max-Forwards: 70 From: "106" ;tag=as149287b0 To: ;tag=1777248976 Contact: Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 102 ACK User-Agent: Asterisk PBX 10.7.0 Content-Length: 0 --- <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK49de740f From: "106" ;tag=as149287b0 To: ;tag=1777248976 Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 103 INVITE Server: kamailio (3.3.1 (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK49de740f From: "106" ;tag=as149287b0 To: ;tag=1777248976 Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 103 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 280 v=0 o=107 3544 3544 IN IP4 10.11.2.50 s=Talk c=IN IP4 10.11.2.50 t=0 0 m=audio 7078 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 9078 RTP/AVP 98 a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 <-------------> --- (10 headers 13 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.11.2.47:5060 Transmitting (no NAT) to 10.11.2.47:5060: ACK sip:107@10.11.2.50 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK0cee6d4d Route: Max-Forwards: 70 From: "106" ;tag=as149287b0 To: ;tag=1777248976 Contact: Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 103 ACK User-Agent: Asterisk PBX 10.7.0 Content-Length: 0 --- <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK5c0ba38d From: "asterisk" ;tag=as063ef50c To: ;tag=22663 Call-ID: 2b27053f3491a2371bef4b0c12ff7754@10.11.2.47 CSeq: 102 NOTIFY User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.11.2.47:5060 ---> ACK sip:107@10.11.2.47:5080 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 10.11.2.37:5060;rport=5060;branch=z9hG4bK31853 From: ;tag=29997 To: "107" ;tag=as6e6ec5ef Call-ID: 18342 CSeq: 21 ACK Contact: Proxy-Authorization: Digest username="106", realm="10.11.2.47", nonce="UYoXa1GKFj+vG/6Ro/04MgF5oIIWA37e", uri="sip:107@10.11.2.47", response="ab8737041d9609bddd05bc5956a6604c", algorithm=MD5 Max-Forwards: 69 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> --- (12 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.11.2.47:5060 Audio is at 16686 Video is at 10.11.2.50:9078 Adding video codec 200003 (h263p) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.11.2.47:5060: INVITE sip:106@(null) SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4 Route: Max-Forwards: 70 From: "107" ;tag=as6e6ec5ef To: ;tag=29997 Contact: Call-ID: 18342 CSeq: 102 INVITE User-Agent: Asterisk PBX 10.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 355 v=0 o=root 742615961 742615962 IN IP4 10.11.2.50 s=Asterisk PBX 10.7.0 c=IN IP4 10.11.2.50 b=CT:384 t=0 0 m=audio 7078 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 9078 RTP/AVP 98 a=rtpmap:98 h263-1998/90000 a=sendrecv --- <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4 From: "107" ;tag=as6e6ec5ef To: ;tag=29997 Call-ID: 18342 CSeq: 102 INVITE Server: kamailio (3.3.1 (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 478 Unresolvable destination (478/SL) Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4 From: "107" ;tag=as6e6ec5ef To: ;tag=29997 Call-ID: 18342 CSeq: 102 INVITE Server: kamailio (3.3.1 (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- Got SIP response 478 "Unresolvable destination (478/SL)" back from 10.11.2.47:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 10.11.2.47:5060 Transmitting (no NAT) to 10.11.2.47:5060: ACK sip:106@(null) SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4 Route: Max-Forwards: 70 From: "107" ;tag=as6e6ec5ef To: ;tag=29997 Contact: Call-ID: 18342 CSeq: 102 ACK User-Agent: Asterisk PBX 10.7.0 Content-Length: 0 --- <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 478 Unresolvable destination (478/TM) Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4 From: "107" ;tag=as6e6ec5ef To: ;tag=29997 Call-ID: 18342 CSeq: 102 INVITE Server: kamailio (3.3.1 (i386/linux)) Content-Length: 0 <-------------> set_destination: Parsing for address/port to send to set_destination: set destination to 10.11.2.47:5060 Audio is at 15732 Video is at 10.11.2.47:16908 Adding codec 100003 (ulaw) to SDP Adding video codec 200003 (h263p) to SDP Adding video codec 200004 (h264) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.11.2.47:5060: INVITE sip:107@10.11.2.50 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK6f188933 Route: Max-Forwards: 70 From: "106" ;tag=as149287b0 To: ;tag=1777248976 Contact: Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 104 INVITE User-Agent: Asterisk PBX 10.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 384 v=0 o=root 476348733 476348735 IN IP4 10.11.2.47 s=Asterisk PBX 10.7.0 c=IN IP4 10.11.2.47 b=CT:384 t=0 0 m=audio 15732 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 16908 RTP/AVP 98 99 a=rtpmap:98 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv --- Scheduling destruction of SIP dialog '08e663336578468a35cd30953f5ae115@10.11.2.47' in 32000 ms (Method: INVITE) == Spawn extension (from-sip, 107, 1) exited non-zero on 'SIP/106-0000001d' --- (8 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.11.2.47:5060 Transmitting (no NAT) to 10.11.2.47:5060: ACK sip:106@(null) SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4 Route: Max-Forwards: 70 From: "107" ;tag=as6e6ec5ef To: ;tag=29997 Contact: Call-ID: 18342 CSeq: 102 ACK User-Agent: Asterisk PBX 10.7.0 Content-Length: 0 --- Really destroying SIP dialog '18342' Method: ACK <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK6f188933 From: "106" ;tag=as149287b0 To: ;tag=1777248976 Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 104 INVITE Server: kamailio (3.3.1 (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK6f188933 From: "106" ;tag=as149287b0 To: ;tag=1777248976 Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 104 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 342 v=0 o=107 3544 3544 IN IP4 10.11.2.50 s=Talk c=IN IP4 10.11.2.50 t=0 0 m=audio 7078 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 9078 RTP/AVP 98 99 a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014 <-------------> --- (10 headers 15 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.11.2.47:5060 Transmitting (no NAT) to 10.11.2.47:5060: ACK sip:107@10.11.2.50 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4ac1db0a Route: Max-Forwards: 70 From: "106" ;tag=as149287b0 To: ;tag=1777248976 Contact: Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 104 ACK User-Agent: Asterisk PBX 10.7.0 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.11.2.47:5060 Reliably Transmitting (no NAT) to 10.11.2.47:5060: BYE sip:107@10.11.2.50 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK37e4c951 Route: Max-Forwards: 70 From: "106" ;tag=as149287b0 To: ;tag=1777248976 Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 105 BYE User-Agent: Asterisk PBX 10.7.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '08e663336578468a35cd30953f5ae115@10.11.2.47' in 32000 ms (Method: INVITE) <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK37e4c951 From: "106" ;tag=as149287b0 To: ;tag=1777248976 Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47 CSeq: 105 BYE User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '08e663336578468a35cd30953f5ae115@10.11.2.47' Method: INVITE Really destroying SIP dialog '2b27053f3491a2371bef4b0c12ff7754@10.11.2.47' Method: NOTIFY Really destroying SIP dialog '5eefa7c2-4330@127.0.0.1' Method: REGISTER <--- SIP read from UDP:10.11.2.47:5060 ---> REGISTER sip:10.11.2.47:5080 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bKbf09.b6677ad3.0 To: sip:106@10.11.2.47 From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-8e78 CSeq: 10 REGISTER Call-ID: 5eefa7c2-4331@127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.3.1 (i386/linux)) Contact: Expires: 60 <-------------> --- (10 headers 0 lines) --- Sending to 10.11.2.47:5060 (NAT) <--- Transmitting (no NAT) to 10.11.2.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bKbf09.b6677ad3.0;received=10.11.2.47 From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-8e78 To: sip:106@10.11.2.47;tag=as7930b677 Call-ID: 5eefa7c2-4331@127.0.0.1 CSeq: 10 REGISTER Server: Asterisk PBX 10.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Wed, 08 May 2013 09:10:17 GMT Content-Length: 0 <------------> [May 8 17:10:17] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1 Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't exist (100) [May 8 17:10:17] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk]... [May 8 17:10:17] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check: Connection is down attempting to reconnect... [May 8 17:10:22] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting asterisk [May 8 17:10:22] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk] [May 8 17:10:22] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1 Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't exist (100) [May 8 17:10:22] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk]... [May 8 17:10:22] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check: Connection is down attempting to reconnect... [May 8 17:10:27] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting asterisk [May 8 17:10:27] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk] [May 8 17:10:27] WARNING[4068]: app_voicemail.c:5142 inboxcount2: SQL Execute error! [SELECT COUNT(*) FROM voicemessages WHERE dir = '/var/spool/asterisk/voicemail/default/106/INBOX'] Scheduling destruction of SIP dialog '1e79945f12aff0b43b9a2ed03c072152@10.11.2.47' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 10.11.2.47:5060: NOTIFY sip:106@10.11.2.47:5060 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4c0421f3 Max-Forwards: 70 From: "asterisk" ;tag=as1b9afa13 To: Contact: Call-ID: 1e79945f12aff0b43b9a2ed03c072152@10.11.2.47 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 10.7.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 90 Messages-Waiting: no Message-Account: sip:asterisk@10.11.2.47 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog '5eefa7c2-4331@127.0.0.1' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4c0421f3 From: "asterisk" ;tag=as1b9afa13 To: ;tag=12759 Call-ID: 1e79945f12aff0b43b9a2ed03c072152@10.11.2.47 CSeq: 102 NOTIFY User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '1e79945f12aff0b43b9a2ed03c072152@10.11.2.47' Method: NOTIFY Really destroying SIP dialog '5eefa7c2-4331@127.0.0.1' Method: REGISTER <--- SIP read from UDP:10.11.2.47:5060 ---> REGISTER sip:10.11.2.47:5080 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK6f9.3171097.0 To: sip:106@10.11.2.47 From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-a6d6 CSeq: 10 REGISTER Call-ID: 5eefa7c0-4332@127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.3.1 (i386/linux)) Contact: Expires: 60 <-------------> --- (10 headers 0 lines) --- Sending to 10.11.2.47:5060 (NAT) <--- Transmitting (no NAT) to 10.11.2.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK6f9.3171097.0;received=10.11.2.47 From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-a6d6 To: sip:106@10.11.2.47;tag=as7b360b52 Call-ID: 5eefa7c0-4332@127.0.0.1 CSeq: 10 REGISTER Server: Asterisk PBX 10.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Wed, 08 May 2013 09:11:12 GMT Content-Length: 0 <------------> [May 8 17:11:12] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1 Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't exist (100) [May 8 17:11:12] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk]... [May 8 17:11:12] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check: Connection is down attempting to reconnect... [May 8 17:11:17] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting asterisk [May 8 17:11:17] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk] [May 8 17:11:17] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1 Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't exist (100) [May 8 17:11:17] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk]... [May 8 17:11:17] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check: Connection is down attempting to reconnect... [May 8 17:11:22] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting asterisk [May 8 17:11:22] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk] [May 8 17:11:22] WARNING[4068]: app_voicemail.c:5142 inboxcount2: SQL Execute error! [SELECT COUNT(*) FROM voicemessages WHERE dir = '/var/spool/asterisk/voicemail/default/106/INBOX'] Scheduling destruction of SIP dialog '2d5c4bc9265ecbe21050c3007ba34d4c@10.11.2.47' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 10.11.2.47:5060: NOTIFY sip:106@10.11.2.47:5060 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK3f9b4370 Max-Forwards: 70 From: "asterisk" ;tag=as776cae27 To: Contact: Call-ID: 2d5c4bc9265ecbe21050c3007ba34d4c@10.11.2.47 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 10.7.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 90 Messages-Waiting: no Message-Account: sip:asterisk@10.11.2.47 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog '5eefa7c0-4332@127.0.0.1' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK3f9b4370 From: "asterisk" ;tag=as776cae27 To: ;tag=19478 Call-ID: 2d5c4bc9265ecbe21050c3007ba34d4c@10.11.2.47 CSeq: 102 NOTIFY User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '2d5c4bc9265ecbe21050c3007ba34d4c@10.11.2.47' Method: NOTIFY Really destroying SIP dialog '5eefa7c0-4332@127.0.0.1' Method: REGISTER <--- SIP read from UDP:10.11.2.47:5060 ---> REGISTER sip:10.11.2.47:5080 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK7f9.1dc36914.0 To: sip:106@10.11.2.47 From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-a73e CSeq: 10 REGISTER Call-ID: 5eefa7c1-4332@127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.3.1 (i386/linux)) Contact: Expires: 60 <-------------> --- (10 headers 0 lines) --- Sending to 10.11.2.47:5060 (NAT) <--- Transmitting (no NAT) to 10.11.2.47:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK7f9.1dc36914.0;received=10.11.2.47 From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-a73e To: sip:106@10.11.2.47;tag=as78794adf Call-ID: 5eefa7c1-4332@127.0.0.1 CSeq: 10 REGISTER Server: Asterisk PBX 10.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Wed, 08 May 2013 09:12:07 GMT Content-Length: 0 <------------> [May 8 17:12:07] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1 Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't exist (100) [May 8 17:12:07] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk]... [May 8 17:12:07] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check: Connection is down attempting to reconnect... [May 8 17:12:12] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting asterisk [May 8 17:12:12] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk] [May 8 17:12:12] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1 Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't exist (100) [May 8 17:12:12] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk]... [May 8 17:12:12] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check: Connection is down attempting to reconnect... [May 8 17:12:17] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting asterisk [May 8 17:12:17] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk] [May 8 17:12:17] WARNING[4068]: app_voicemail.c:5142 inboxcount2: SQL Execute error! [SELECT COUNT(*) FROM voicemessages WHERE dir = '/var/spool/asterisk/voicemail/default/106/INBOX'] Scheduling destruction of SIP dialog '5aa0a13752a06608634fd1202102221a@10.11.2.47' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 10.11.2.47:5060: NOTIFY sip:106@10.11.2.47:5060 SIP/2.0 Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK54115b1d Max-Forwards: 70 From: "asterisk" ;tag=as58704a46 To: Contact: Call-ID: 5aa0a13752a06608634fd1202102221a@10.11.2.47 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 10.7.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 90 Messages-Waiting: no Message-Account: sip:asterisk@10.11.2.47 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog '5eefa7c1-4332@127.0.0.1' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.11.2.47:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK54115b1d From: "asterisk" ;tag=as58704a46 To: ;tag=12490 Call-ID: 5aa0a13752a06608634fd1202102221a@10.11.2.47 CSeq: 102 NOTIFY User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- WH-PC*CLI>