root@kamast: ~ $ asterisk -rvvv Asterisk 13.28.0, Copyright (C) 1999 - 2014, 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 13.28.0 currently running on kamast (pid = 1057) kamast*CLI> sip set debug on SIP Debugging re-enabled <--- SIP read from UDP:192.168.1.220:5060 ---> REGISTER sip:192.168.1.220:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bKf50d.f0e297f3000000000000000000000000.0 To: From: ;tag=b8aedf903b5aacaad04d669d0415f139-cb0e CSeq: 10 REGISTER Call-ID: 226cdfda2acb89c1-1362@192.168.1.220 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.2.4 (x86_64/linux)) Contact: Expires: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.1.220:5060 (NAT) Sending to 192.168.1.220:5060 (NAT) <--- Transmitting (NAT) to 192.168.1.220:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bKf50d.f0e297f3000000000000000000000000.0;received=192.168.1.220;rport=5060 From: ;tag=b8aedf903b5aacaad04d669d0415f139-cb0e To: ;tag=as43390e06 Call-ID: 226cdfda2acb89c1-1362@192.168.1.220 CSeq: 10 REGISTER Server: Asterisk PBX 13.28.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c21b584" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '226cdfda2acb89c1-1362@192.168.1.220' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.1.220:5060 ---> REGISTER sip:192.168.1.220:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK16a2.0de6b7d6000000000000000000000000.0 To: From: ;tag=b8aedf903b5aacaad04d669d0415f139-556c CSeq: 10 REGISTER Call-ID: 226cdfda2acb89c3-1360@192.168.1.220 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.2.4 (x86_64/linux)) Contact: Expires: 3600 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.1.220:5060 (NAT) Sending to 192.168.1.220:5060 (NAT) <--- Transmitting (NAT) to 192.168.1.220:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK16a2.0de6b7d6000000000000000000000000.0;received=192.168.1.220;rport=5060 From: ;tag=b8aedf903b5aacaad04d669d0415f139-556c To: ;tag=as21ba153d Call-ID: 226cdfda2acb89c3-1360@192.168.1.220 CSeq: 10 REGISTER Server: Asterisk PBX 13.28.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="639e90d5" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '226cdfda2acb89c3-1360@192.168.1.220' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.1.220:5060 ---> REGISTER sip:192.168.1.220:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK5544.e96aade7000000000000000000000000.0 To: From: ;tag=b8aedf903b5aacaad04d669d0415f139-ee1a CSeq: 10 REGISTER Call-ID: 226cdfda2acb89c2-1366@192.168.1.220 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.2.4 (x86_64/linux)) Contact: Expires: 3600 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.1.220:5060 (NAT) Sending to 192.168.1.220:5060 (NAT) <--- Transmitting (NAT) to 192.168.1.220:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK5544.e96aade7000000000000000000000000.0;received=192.168.1.220;rport=5060 From: ;tag=b8aedf903b5aacaad04d669d0415f139-ee1a To: ;tag=as6368208e Call-ID: 226cdfda2acb89c2-1366@192.168.1.220 CSeq: 10 REGISTER Server: Asterisk PBX 13.28.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="553196a9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '226cdfda2acb89c2-1366@192.168.1.220' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.1.220:5060 ---> INVITE sip:2200@192.168.1.220 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bKea02.587987c6ebf7679b11e43b0381d9cf5f.0 Via: SIP/2.0/UDP 192.168.1.124:53134;received=192.168.1.124;branch=z9hG4bK881078656;rport=53134 From: ;tag=921537115 To: Call-ID: 1686350530-53134-2@BJC.BGI.B.BCE CSeq: 11 INVITE Contact: Max-Forwards: 69 User-Agent: Grandstream Wave 1.0.3.29 Privacy: none P-Preferred-Identity: Supported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 273 v=0 o=2201 8000 8000 IN IP4 192.168.1.124 s=SIP Call c=IN IP4 192.168.1.124 t=0 0 m=audio 34322 RTP/AVP 0 8 101 a=sendrecv a=rtcp:34323 IN IP4 192.168.1.124 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (18 headers 13 lines) --- Sending to 192.168.1.220:5060 (NAT) Sending to 192.168.1.220:5060 (NAT) Using INVITE request as basis request - 1686350530-53134-2@BJC.BGI.B.BCE Found peer '2201' for '2201' from 192.168.1.220:5060 <--- Reliably Transmitting (NAT) to 192.168.1.220:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bKea02.587987c6ebf7679b11e43b0381d9cf5f.0;received=192.168.1.220;rport=5060 Via: SIP/2.0/UDP 192.168.1.124:53134;received=192.168.1.124;branch=z9hG4bK881078656;rport=53134 From: ;tag=921537115 To: ;tag=as6d84c401 Call-ID: 1686350530-53134-2@BJC.BGI.B.BCE CSeq: 11 INVITE Server: Asterisk PBX 13.28.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5751ce2d" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1686350530-53134-2@BJC.BGI.B.BCE' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.1.220:5060 ---> ACK sip:2200@192.168.1.220 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bKea02.587987c6ebf7679b11e43b0381d9cf5f.0 From: ;tag=921537115 To: ;tag=as6d84c401 Call-ID: 1686350530-53134-2@BJC.BGI.B.BCE CSeq: 11 ACK Max-Forwards: 69 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '226cdfda2acb89c2-1360@192.168.1.220' Method: REGISTER Really destroying SIP dialog '226cdfda2acb89c1-1364@192.168.1.220' Method: REGISTER <--- SIP read from UDP:192.168.1.220:5060 ---> INVITE sip:2201@192.168.1.220:5060;transport=UDP SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK65f4.6d66a28122068dc7b1d0e78ebbb5b6f1.0 Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---bf7b312adf2d2639 Max-Forwards: 69 Contact: To: From: ;tag=dc6a6e43 Call-ID: lS-TaaAPk4bdBOlm9Md8Sw.. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 237 v=0 o=Z 0 0 IN IP4 192.168.1.7 s=Z c=IN IP4 192.168.1.7 t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (14 headers 12 lines) --- Sending to 192.168.1.220:5060 (NAT) Sending to 192.168.1.220:5060 (NAT) Using INVITE request as basis request - lS-TaaAPk4bdBOlm9Md8Sw.. Found peer '2200' for '2200' from 192.168.1.220:5060 <--- Reliably Transmitting (NAT) to 192.168.1.220:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK65f4.6d66a28122068dc7b1d0e78ebbb5b6f1.0;received=192.168.1.220;rport=5060 Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---bf7b312adf2d2639 From: ;tag=dc6a6e43 To: ;tag=as3abe9d45 Call-ID: lS-TaaAPk4bdBOlm9Md8Sw.. CSeq: 2 INVITE Server: Asterisk PBX 13.28.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="53e88fda" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'lS-TaaAPk4bdBOlm9Md8Sw..' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.1.220:5060 ---> ACK sip:2201@192.168.1.220:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK65f4.6d66a28122068dc7b1d0e78ebbb5b6f1.0 Max-Forwards: 69 To: ;tag=as3abe9d45 From: ;tag=dc6a6e43 Call-ID: lS-TaaAPk4bdBOlm9Md8Sw.. CSeq: 2 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.1.220:5060 ---> INVITE sip:2201@192.168.1.220:5060;transport=UDP SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK75f4.0f17c1d7306f5916f3c57249b76d8875.0 Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---3807847c5e55fd16 Max-Forwards: 69 Contact: To: From: ;tag=dc6a6e43 Call-ID: lS-TaaAPk4bdBOlm9Md8Sw.. CSeq: 3 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Authorization: Digest username="2200",realm="asterisk",nonce="53e88fda",uri="sip:2201@192.168.1.220:5060;transport=UDP",response="4ef78b02387758ae5180d8dcbd789550",algorithm=MD5 Allow-Events: presence, kpml, talk Content-Length: 237 v=0 o=Z 0 0 IN IP4 192.168.1.7 s=Z c=IN IP4 192.168.1.7 t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.1.220:5060 (NAT) Using INVITE request as basis request - lS-TaaAPk4bdBOlm9Md8Sw.. Found peer '2200' for '2200' from 192.168.1.220:5060 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 3 Found RTP audio format 110 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 97 Found RTP audio format 101 Found audio description format speex for ID 110 Found audio description format iLBC for ID 97 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|g729|gsm), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (alaw|gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.1.7:8000 Looking for 2201 in DialIn (domain 192.168.1.220) sip_route_dump: route/path hop: <--- Transmitting (NAT) to 192.168.1.220:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK75f4.0f17c1d7306f5916f3c57249b76d8875.0;received=192.168.1.220;rport=5060 Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---3807847c5e55fd16 Record-Route: From: ;tag=dc6a6e43 To: Call-ID: lS-TaaAPk4bdBOlm9Md8Sw.. CSeq: 3 INVITE Server: Asterisk PBX 13.28.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [2201@DialIn:1] Dial("SIP/2200-00000000", "SIP/2201") in new stack [Sep 3 15:05:08] WARNING[1432][C-00000002]: app_dial.c:2591 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [2201@DialIn:2] VoiceMail("SIP/2200-00000000", "2201,u") in new stack Audio is at 16900 Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.1.220:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK75f4.0f17c1d7306f5916f3c57249b76d8875.0;received=192.168.1.220;rport=5060 Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---3807847c5e55fd16 Record-Route: From: ;tag=dc6a6e43 To: ;tag=as4c7b3f98 Call-ID: lS-TaaAPk4bdBOlm9Md8Sw.. CSeq: 3 INVITE Server: Asterisk PBX 13.28.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 1924428461 1924428461 IN IP4 192.168.1.220 s=Asterisk PBX 13.28.0 c=IN IP4 192.168.1.220 t=0 0 m=audio 16900 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> Really destroying SIP dialog '7c582d3252da826f5aaa393a7e07751c@192.168.1.220:5080' Method: INVITE <--- SIP read from UDP:192.168.1.220:5060 ---> ACK sip:2201@192.168.1.220:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK75f4.210aaf84a7a9186a6db02526407a9020.0 Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---a05573ff8a9b9267 Max-Forwards: 69 Contact: To: ;tag=as4c7b3f98 From: ;tag=dc6a6e43 Call-ID: lS-TaaAPk4bdBOlm9Md8Sw.. CSeq: 3 ACK User-Agent: Z 3.15.40006 rv2.8.20 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Sep 3 15:05:09] WARNING[1432][C-00000002]: app_voicemail.c:6726 leave_voicemail: No entry in voicemail config file for '2201' -- Executing [2201@DialIn:3] Hangup("SIP/2200-00000000", "") in new stack == Spawn extension (DialIn, 2201, 3) exited non-zero on 'SIP/2200-00000000' Scheduling destruction of SIP dialog 'lS-TaaAPk4bdBOlm9Md8Sw..' in 32000 ms (Method: ACK) Reliably Transmitting (NAT) to 192.168.1.220:5060: BYE sip:2200@192.168.1.7:45108;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.220:5080;branch=z9hG4bK51b88b44;rport Route: Max-Forwards: 70 From: ;tag=as4c7b3f98 To: ;tag=dc6a6e43 Call-ID: lS-TaaAPk4bdBOlm9Md8Sw.. CSeq: 102 BYE User-Agent: Asterisk PBX 13.28.0 Proxy-Authorization: Digest username="2200", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.220", nonce="53e88fda", response="79068713ebb60b832e1b11128e7b30b3" X-Asterisk-HangupCause: Subscriber absent X-Asterisk-HangupCauseCode: 20 Content-Length: 0 --- [Sep 3 15:05:09] WARNING[1225]: res_odbc.c:1075 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Sep 3 15:05:09] ERROR[1225]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle. CDR failed. <--- SIP read from UDP:192.168.1.220:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.220:5080;received=192.168.1.220;branch=z9hG4bK51b88b44;rport=5080 Contact: To: ;tag=dc6a6e43 From: ;tag=as4c7b3f98 Call-ID: lS-TaaAPk4bdBOlm9Md8Sw.. CSeq: 102 BYE User-Agent: Z 3.15.40006 rv2.8.20 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'lS-TaaAPk4bdBOlm9Md8Sw..' Method: ACK Really destroying SIP dialog '226cdfda2acb89c1-1362@192.168.1.220' Method: REGISTER Really destroying SIP dialog '226cdfda2acb89c3-1360@192.168.1.220' Method: REGISTER Really destroying SIP dialog '226cdfda2acb89c2-1366@192.168.1.220' Method: REGISTER <--- SIP read from UDP:192.168.1.220:5060 ---> INVITE sip:2201@192.168.1.220:5060;transport=UDP SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK0dae.243a824c763a788b3062fcc9ee025dcc.0 Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---6b58a840ab70676f Max-Forwards: 69 Contact: To: From: ;tag=ea2c8d78 Call-ID: 1od4Xzsm69VX6V2k73jWEw.. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 237 v=0 o=Z 0 0 IN IP4 192.168.1.7 s=Z c=IN IP4 192.168.1.7 t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (14 headers 12 lines) --- Sending to 192.168.1.220:5060 (NAT) Sending to 192.168.1.220:5060 (NAT) Using INVITE request as basis request - 1od4Xzsm69VX6V2k73jWEw.. Found peer '2200' for '2200' from 192.168.1.220:5060 <--- Reliably Transmitting (NAT) to 192.168.1.220:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK0dae.243a824c763a788b3062fcc9ee025dcc.0;received=192.168.1.220;rport=5060 Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---6b58a840ab70676f From: ;tag=ea2c8d78 To: ;tag=as2dd9dd23 Call-ID: 1od4Xzsm69VX6V2k73jWEw.. CSeq: 2 INVITE Server: Asterisk PBX 13.28.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5da84077" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1od4Xzsm69VX6V2k73jWEw..' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.1.220:5060 ---> ACK sip:2201@192.168.1.220:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK0dae.243a824c763a788b3062fcc9ee025dcc.0 Max-Forwards: 69 To: ;tag=as2dd9dd23 From: ;tag=ea2c8d78 Call-ID: 1od4Xzsm69VX6V2k73jWEw.. CSeq: 2 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.1.220:5060 ---> INVITE sip:2201@192.168.1.220:5060;transport=UDP SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK1dae.8048dfe2e2bb5b27777e466752e27231.0 Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---7efa4b7e878e8087 Max-Forwards: 69 Contact: To: From: ;tag=ea2c8d78 Call-ID: 1od4Xzsm69VX6V2k73jWEw.. CSeq: 3 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Authorization: Digest username="2200",realm="asterisk",nonce="5da84077",uri="sip:2201@192.168.1.220:5060;transport=UDP",response="71781c02a2d439bc80981d7662b2d87a",algorithm=MD5 Allow-Events: presence, kpml, talk Content-Length: 237 v=0 o=Z 0 0 IN IP4 192.168.1.7 s=Z c=IN IP4 192.168.1.7 t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.1.220:5060 (NAT) Using INVITE request as basis request - 1od4Xzsm69VX6V2k73jWEw.. Found peer '2200' for '2200' from 192.168.1.220:5060 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 3 Found RTP audio format 110 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 97 Found RTP audio format 101 Found audio description format speex for ID 110 Found audio description format iLBC for ID 97 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|g729|gsm), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (alaw|gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.1.7:8000 Looking for 2201 in DialIn (domain 192.168.1.220) sip_route_dump: route/path hop: <--- Transmitting (NAT) to 192.168.1.220:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK1dae.8048dfe2e2bb5b27777e466752e27231.0;received=192.168.1.220;rport=5060 Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---7efa4b7e878e8087 Record-Route: From: ;tag=ea2c8d78 To: Call-ID: 1od4Xzsm69VX6V2k73jWEw.. CSeq: 3 INVITE Server: Asterisk PBX 13.28.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [2201@DialIn:1] Dial("SIP/2200-00000001", "SIP/2201") in new stack [Sep 3 15:05:23] WARNING[1435][C-00000003]: app_dial.c:2591 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [2201@DialIn:2] VoiceMail("SIP/2200-00000001", "2201,u") in new stack Audio is at 19634 Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.1.220:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK1dae.8048dfe2e2bb5b27777e466752e27231.0;received=192.168.1.220;rport=5060 Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---7efa4b7e878e8087 Record-Route: From: ;tag=ea2c8d78 To: ;tag=as564764f2 Call-ID: 1od4Xzsm69VX6V2k73jWEw.. CSeq: 3 INVITE Server: Asterisk PBX 13.28.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 104362391 104362391 IN IP4 192.168.1.220 s=Asterisk PBX 13.28.0 c=IN IP4 192.168.1.220 t=0 0 m=audio 19634 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> Really destroying SIP dialog '47b487740987fcc6581b748f605b07ad@192.168.1.220:5080' Method: INVITE <--- SIP read from UDP:192.168.1.220:5060 ---> ACK sip:2201@192.168.1.220:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK1dae.210333542ab05a25de4f08108f0ba08a.0 Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---1fdacb4bc80c606c Max-Forwards: 69 Contact: To: ;tag=as564764f2 From: ;tag=ea2c8d78 Call-ID: 1od4Xzsm69VX6V2k73jWEw.. CSeq: 3 ACK User-Agent: Z 3.15.40006 rv2.8.20 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Sep 3 15:05:24] WARNING[1435][C-00000003]: app_voicemail.c:6726 leave_voicemail: No entry in voicemail config file for '2201' -- Executing [2201@DialIn:3] Hangup("SIP/2200-00000001", "") in new stack == Spawn extension (DialIn, 2201, 3) exited non-zero on 'SIP/2200-00000001' Scheduling destruction of SIP dialog '1od4Xzsm69VX6V2k73jWEw..' in 32000 ms (Method: ACK) Reliably Transmitting (NAT) to 192.168.1.220:5060: BYE sip:2200@192.168.1.7:45108;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.220:5080;branch=z9hG4bK58ff235d;rport Route: Max-Forwards: 70 From: ;tag=as564764f2 To: ;tag=ea2c8d78 Call-ID: 1od4Xzsm69VX6V2k73jWEw.. CSeq: 102 BYE User-Agent: Asterisk PBX 13.28.0 Proxy-Authorization: Digest username="2200", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.220", nonce="5da84077", response="925eb75d16d876118d4e1b6b3680dc03" X-Asterisk-HangupCause: Subscriber absent X-Asterisk-HangupCauseCode: 20 Content-Length: 0 --- [Sep 3 15:05:24] WARNING[1225]: res_odbc.c:1075 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Sep 3 15:05:24] ERROR[1225]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle. CDR failed. <--- SIP read from UDP:192.168.1.220:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.220:5080;received=192.168.1.220;branch=z9hG4bK58ff235d;rport=5080 Contact: To: ;tag=ea2c8d78 From: ;tag=as564764f2 Call-ID: 1od4Xzsm69VX6V2k73jWEw.. CSeq: 102 BYE User-Agent: Z 3.15.40006 rv2.8.20 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '1od4Xzsm69VX6V2k73jWEw..' Method: ACK kamast*CLI> sip set debug off SIP Debugging Disabled kamast*CLI>