Asterisk 12.1.0, Copyright (C) 1999 - 2013 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. ========================================================================= Running as user 'asteriskpbx' Running under group 'asteriskpbx' Connected to Asterisk 12.1.0 currently running on SNCA00-PH00 (pid = 4639) SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK18995b85;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21768 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI> Really destroying SIP dialog '644d1bd7027daa0576e12ff149af9f4e@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK36f4c065;rport From: "asterisk" ;tag=as45c24d2a To: Contact: Call-ID: 5adee4f056681f33277ccbed4c91875e@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK36f4c065;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as45c24d2a To: ;tag=as709a568c Call-ID: 5adee4f056681f33277ccbed4c91875e@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '5adee4f056681f33277ccbed4c91875e@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK32808f98;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21806 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK32808f98;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21806 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b4ebbb3", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI> Really destroying SIP dialog '043acd9b7b1bfd8c44cb9cda57436a86@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI> Really destroying SIP dialog '5ec98666786fab237c51b5e37e377b8f@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI> Really destroying SIP dialog '1168f181604d8c2230ef4a5905cbd62c@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK18995b85;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21768 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI> Really destroying SIP dialog '29de4bae21edde9f747f9de1654bc6b8@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK32808f98;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21806 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK32808f98;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21806 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7e3f2e5d", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK567c33fa;rport From: "asterisk" ;tag=as0fbfae81 To: Contact: Call-ID: 5e01ea17210c62654835c47b62b70ae7@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK567c33fa;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as0fbfae81 To: ;tag=as28ec1cee Call-ID: 5e01ea17210c62654835c47b62b70ae7@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '5e01ea17210c62654835c47b62b70ae7@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK18995b85;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21768 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK32808f98;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21806 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK69f6cdbc;rport From: "asterisk" ;tag=as68ddfc39 To: Contact: Call-ID: 531f0d354d50c48044381db747792f48@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK69f6cdbc;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as68ddfc39 To: ;tag=as41b095ea Call-ID: 531f0d354d50c48044381db747792f48@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '531f0d354d50c48044381db747792f48@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK32808f98;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21806 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75a376e7", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK3d0aca67;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21769 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c17c042", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK49099e6e;rport From: "asterisk" ;tag=as29867d7b To: Contact: Call-ID: 13c4d2ca25bd78d84e9922bf5ce4da01@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK49099e6e;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as29867d7b To: ;tag=as6874a20d Call-ID: 13c4d2ca25bd78d84e9922bf5ce4da01@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '13c4d2ca25bd78d84e9922bf5ce4da01@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK3d0aca67;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21769 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI> Really destroying SIP dialog '736d8bf50b50fe565cabed695c5a005c@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40673072", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK42eb6d3d;rport From: "asterisk" ;tag=as682e2878 To: Contact: Call-ID: 2f852c705df7c39e171395997db885d7@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK42eb6d3d;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as682e2878 To: ;tag=as28a85923 Call-ID: 2f852c705df7c39e171395997db885d7@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2f852c705df7c39e171395997db885d7@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> INVITE sip:17147240410@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK888233890;rport Route: From: ;tag=34876174 To: Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 170 INVITE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Privacy: none P-Preferred-Identity: Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 339 v=0 o=000b825e3aa5 8000 8000 IN IP4 10.42.3.101 s=SIP Call c=IN IP4 10.42.3.101 t=0 0 m=audio 5004 RTP/AVP 0 8 18 9 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (18 headers 16 lines) --- SNCA00-PH00*CLI> Sending to 10.42.3.101:5060 (no NAT) SNCA00-PH00*CLI> Sending to 10.42.3.101:5060 (no NAT) Using INVITE request as basis request - 208247511-5060-17310@BA.EC.D.BAB SNCA00-PH00*CLI> Found peer '000b825e3aa5' for '000b825e3aa5' from 10.42.3.101:5060 SNCA00-PH00*CLI>  <--- Reliably Transmitting (no NAT) to 10.42.3.101:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK888233890;received=10.42.3.101;rport=5060 From: ;tag=34876174 To: ;tag=as5d5f4945 Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 170 INVITE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="500e8dad" Content-Length: 0 <------------> SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '208247511-5060-17310@BA.EC.D.BAB' in 32000 ms (Method: INVITE) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> ACK sip:17147240410@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK888233890;rport Route: From: ;tag=34876174 To: ;tag=as5d5f4945 Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 170 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> INVITE sip:17147240410@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK598328373;rport Route: From: ;tag=34876174 To: Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 171 INVITE Contact: Authorization: Digest username="000b825e3aa5", realm="asterisk", nonce="500e8dad", uri="sip:17147240410@10.42.0.20", response="f5be6e8d2fcb196d736e14c98bd6e0f1", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Privacy: none P-Preferred-Identity: Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 339 v=0 o=000b825e3aa5 8000 8000 IN IP4 10.42.3.101 s=SIP Call c=IN IP4 10.42.3.101 t=0 0 m=audio 5004 RTP/AVP 0 8 18 9 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (19 headers 16 lines) --- SNCA00-PH00*CLI> Sending to 10.42.3.101:5060 (no NAT) Using INVITE request as basis request - 208247511-5060-17310@BA.EC.D.BAB SNCA00-PH00*CLI> Found peer '000b825e3aa5' for '000b825e3aa5' from 10.42.3.101:5060 SNCA00-PH00*CLI>  == Using SIP RTP CoS mark 5 SNCA00-PH00*CLI> Found RTP audio format 0 Found RTP audio format 8 SNCA00-PH00*CLI> Found RTP audio format 18 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 101 SNCA00-PH00*CLI> Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format G722 for ID 9 Found audio description format G726-32 for ID 2 SNCA00-PH00*CLI> Found audio description format telephone-event for ID 101 SNCA00-PH00*CLI> Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.42.3.101:5004 SNCA00-PH00*CLI> Looking for 17147240410 in LocalSets (domain 10.42.0.20) SNCA00-PH00*CLI> list_route: route/path hop: SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.101:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK598328373;received=10.42.3.101;rport=5060 From: ;tag=34876174 To: Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 171 INVITE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> SNCA00-PH00*CLI>  -- Executing [17147240410@LocalSets:1] Progress("SIP/000b825e3aa5-00000002", "") in new stack SNCA00-PH00*CLI> Audio is at 26398 Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP SNCA00-PH00*CLI> Adding non-codec 0x1 (telephone-event) to SDP SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.101:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK598328373;received=10.42.3.101;rport=5060 From: ;tag=34876174 To: ;tag=as62690954 Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 171 INVITE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 269 v=0 o=root 421206821 421206821 IN IP4 10.42.0.20 s=Asterisk PBX 12.1.0 c=IN IP4 10.42.0.20 t=0 0 m=audio 26398 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=ptime:20 a=maxptime:150 a=sendrecv <------------> SNCA00-PH00*CLI>  -- Executing [17147240410@LocalSets:2] SayDigits("SIP/000b825e3aa5-00000002", "7240410") in new stack SNCA00-PH00*CLI>  -- Playing 'digits/7.ulaw' (language 'en') SNCA00-PH00*CLI>  > 0x7ffce4083800 -- Probation passed - setting RTP source address to 10.42.3.101:5004 SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK3d0aca67;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21769 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI>  -- Playing 'digits/2.ulaw' (language 'en') SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56fa3557", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK3a8cedf2;rport From: "asterisk" ;tag=as4aa46cd5 To: Contact: Call-ID: 47afad2f28b398a830fc09356d7fa26b@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK3a8cedf2;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as4aa46cd5 To: ;tag=as5a2f487c Call-ID: 47afad2f28b398a830fc09356d7fa26b@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '47afad2f28b398a830fc09356d7fa26b@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  -- Playing 'digits/4.ulaw' (language 'en') SNCA00-PH00*CLI>  -- Playing 'digits/0.ulaw' (language 'en') SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK3d0aca67;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21769 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI> Really destroying SIP dialog '35044e296bda1a3e76c80a9417a50f42@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI>  -- Playing 'digits/4.ulaw' (language 'en') SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d27a15a", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK183714fc;rport From: "asterisk" ;tag=as28c2d0b3 To: Contact: Call-ID: 4b81c5d355f76acd15310ef1035db7e3@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK183714fc;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as28c2d0b3 To: ;tag=as32070509 Call-ID: 4b81c5d355f76acd15310ef1035db7e3@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '4b81c5d355f76acd15310ef1035db7e3@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.100:5060 ---> SUBSCRIBE sip:000b825f735e@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.100:5060;branch=z9hG4bK919709273;rport Route: From: ;tag=1205332359 To: Call-ID: 52053088-5060-17292@BA.EC.D.BAA CSeq: 192900 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (17 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 10.42.3.100:5060 (no NAT) SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.100:5060 (no NAT) list_route: route/path hop: SNCA00-PH00*CLI> Found peer '000b825f735e' for '000b825f735e' from 10.42.3.100:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.100:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.3.100:5060;branch=z9hG4bK919709273;received=10.42.3.100;rport=5060 From: ;tag=1205332359 To: ;tag=as01b94ca6 Call-ID: 52053088-5060-17292@BA.EC.D.BAA CSeq: 192900 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f571c07" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '52053088-5060-17292@BA.EC.D.BAA' in 32000 ms (Method: SUBSCRIBE) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.100:5060 ---> SUBSCRIBE sip:000b825f735e@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.100:5060;branch=z9hG4bK501067292;rport Route: From: ;tag=1205332359 To: Call-ID: 52053088-5060-17292@BA.EC.D.BAA CSeq: 192901 SUBSCRIBE Contact: Authorization: Digest username="000b825f735e", realm="asterisk", nonce="2f571c07", uri="sip:000b825f735e@10.42.0.20", response="ed5a0d70d8edbe6a49ac88aa38e58308", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (18 headers 0 lines) --- SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.100:5060 (no NAT) Found peer '000b825f735e' for '000b825f735e' from 10.42.3.100:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.100:5060 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 10.42.3.100:5060;branch=z9hG4bK501067292;received=10.42.3.100;rport=5060 From: ;tag=1205332359 To: ;tag=as01b94ca6 Call-ID: 52053088-5060-17292@BA.EC.D.BAA CSeq: 192901 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> SNCA00-PH00*CLI> Really destroying SIP dialog '52053088-5060-17292@BA.EC.D.BAA' Method: SUBSCRIBE SNCA00-PH00*CLI>  -- Playing 'digits/1.ulaw' (language 'en') SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.103:5060 ---> SUBSCRIBE sip:000b825f735f@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.103:5060;branch=z9hG4bK1050586096;rport Route: From: ;tag=1819741874 To: Call-ID: 280931976-5060-17294@BA.EC.D.BAD CSeq: 192920 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (17 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 10.42.3.103:5060 (no NAT) SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.103:5060 (no NAT) list_route: route/path hop: Found peer '000b825f735f' for '000b825f735f' from 10.42.3.103:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.103:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.3.103:5060;branch=z9hG4bK1050586096;received=10.42.3.103;rport=5060 From: ;tag=1819741874 To: ;tag=as5e119f41 Call-ID: 280931976-5060-17294@BA.EC.D.BAD CSeq: 192920 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0d773899" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '280931976-5060-17294@BA.EC.D.BAD' in 32000 ms (Method: SUBSCRIBE) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.103:5060 ---> SUBSCRIBE sip:000b825f735f@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.103:5060;branch=z9hG4bK1905629962;rport Route: From: ;tag=1819741874 To: Call-ID: 280931976-5060-17294@BA.EC.D.BAD CSeq: 192921 SUBSCRIBE Contact: Authorization: Digest username="000b825f735f", realm="asterisk", nonce="0d773899", uri="sip:000b825f735f@10.42.0.20", response="74316a8d5a631eb0a3404f0b360a1985", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (18 headers 0 lines) --- Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.103:5060 (no NAT) Found peer '000b825f735f' for '000b825f735f' from 10.42.3.103:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.103:5060 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 10.42.3.103:5060;branch=z9hG4bK1905629962;received=10.42.3.103;rport=5060 From: ;tag=1819741874 To: ;tag=as5e119f41 Call-ID: 280931976-5060-17294@BA.EC.D.BAD CSeq: 192921 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> SNCA00-PH00*CLI> Really destroying SIP dialog '280931976-5060-17294@BA.EC.D.BAD' Method: SUBSCRIBE SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.100:5060 ---> SUBSCRIBE sip:000b825f735e@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.100:5060;branch=z9hG4bK1597533272;rport Route: From: ;tag=1654327206 To: Call-ID: 653167201-5060-17293@BA.EC.D.BAA CSeq: 192910 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (17 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 10.42.3.100:5060 (no NAT) SNCA00-PH00*CLI> Creating new subscription Sending to 10.42.3.100:5060 (no NAT) list_route: route/path hop: SNCA00-PH00*CLI> Found peer '000b825f735e' for '000b825f735e' from 10.42.3.100:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.100:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.3.100:5060;branch=z9hG4bK1597533272;received=10.42.3.100;rport=5060 From: ;tag=1654327206 To: ;tag=as6c393d2d Call-ID: 653167201-5060-17293@BA.EC.D.BAA CSeq: 192910 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3076e25f" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '653167201-5060-17293@BA.EC.D.BAA' in 32000 ms (Method: SUBSCRIBE) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.100:5060 ---> SUBSCRIBE sip:000b825f735e@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.100:5060;branch=z9hG4bK26510839;rport Route: From: ;tag=1654327206 To: Call-ID: 653167201-5060-17293@BA.EC.D.BAA CSeq: 192911 SUBSCRIBE Contact: Authorization: Digest username="000b825f735e", realm="asterisk", nonce="3076e25f", uri="sip:000b825f735e@10.42.0.20", response="49a99f78d510f3cd2b075c71a49bc043", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (18 headers 0 lines) --- SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.100:5060 (no NAT) Found peer '000b825f735e' for '000b825f735e' from 10.42.3.100:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.100:5060 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 10.42.3.100:5060;branch=z9hG4bK26510839;received=10.42.3.100;rport=5060 From: ;tag=1654327206 To: ;tag=as6c393d2d Call-ID: 653167201-5060-17293@BA.EC.D.BAA CSeq: 192911 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> SNCA00-PH00*CLI> Really destroying SIP dialog '653167201-5060-17293@BA.EC.D.BAA' Method: SUBSCRIBE SNCA00-PH00*CLI>  -- Playing 'digits/0.ulaw' (language 'en') SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.104:5060 ---> SUBSCRIBE sip:000b825c68cf@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.104:5060;branch=z9hG4bK254409278;rport Route: From: ;tag=1624850939 To: Call-ID: 311972129-5060-10474@BA.EC.D.BAE CSeq: 124720 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (17 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 10.42.3.104:5060 (no NAT) SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.104:5060 (no NAT) list_route: route/path hop: Found peer '000b825c68cf' for '000b825c68cf' from 10.42.3.104:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.104:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.3.104:5060;branch=z9hG4bK254409278;received=10.42.3.104;rport=5060 From: ;tag=1624850939 To: ;tag=as3c98a75b Call-ID: 311972129-5060-10474@BA.EC.D.BAE CSeq: 124720 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7e392405" Content-Length: 0 <------------> SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '311972129-5060-10474@BA.EC.D.BAE' in 32000 ms (Method: SUBSCRIBE) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.104:5060 ---> SUBSCRIBE sip:000b825c68cf@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.104:5060;branch=z9hG4bK630141490;rport Route: From: ;tag=1624850939 To: Call-ID: 311972129-5060-10474@BA.EC.D.BAE CSeq: 124721 SUBSCRIBE Contact: Authorization: Digest username="000b825c68cf", realm="asterisk", nonce="7e392405", uri="sip:000b825c68cf@10.42.0.20", response="ac578d9e2708b7e4654febca5836a7da", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (18 headers 0 lines) --- SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.104:5060 (no NAT) Found peer '000b825c68cf' for '000b825c68cf' from 10.42.3.104:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.104:5060 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 10.42.3.104:5060;branch=z9hG4bK630141490;received=10.42.3.104;rport=5060 From: ;tag=1624850939 To: ;tag=as3c98a75b Call-ID: 311972129-5060-10474@BA.EC.D.BAE CSeq: 124721 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog '311972129-5060-10474@BA.EC.D.BAE' Method: SUBSCRIBE SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.103:5060 ---> SUBSCRIBE sip:000b825f735f@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.103:5060;branch=z9hG4bK1163620559;rport Route: From: ;tag=1499347744 To: Call-ID: 2085551669-5060-17295@BA.EC.D.BAD CSeq: 192930 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (17 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 10.42.3.103:5060 (no NAT) Creating new subscription Sending to 10.42.3.103:5060 (no NAT) list_route: route/path hop: Found peer '000b825f735f' for '000b825f735f' from 10.42.3.103:5060 <--- Transmitting (no NAT) to 10.42.3.103:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.3.103:5060;branch=z9hG4bK1163620559;received=10.42.3.103;rport=5060 From: ;tag=1499347744 To: ;tag=as25e475f0 Call-ID: 2085551669-5060-17295@BA.EC.D.BAD CSeq: 192930 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="159201ba" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2085551669-5060-17295@BA.EC.D.BAD' in 32000 ms (Method: SUBSCRIBE) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.103:5060 ---> SUBSCRIBE sip:000b825f735f@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.103:5060;branch=z9hG4bK304297920;rport Route: From: ;tag=1499347744 To: Call-ID: 2085551669-5060-17295@BA.EC.D.BAD CSeq: 192931 SUBSCRIBE Contact: Authorization: Digest username="000b825f735f", realm="asterisk", nonce="159201ba", uri="sip:000b825f735f@10.42.0.20", response="7fe38f58d192e7417affd97ca830d203", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (18 headers 0 lines) --- SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.103:5060 (no NAT) Found peer '000b825f735f' for '000b825f735f' from 10.42.3.103:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.103:5060 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 10.42.3.103:5060;branch=z9hG4bK304297920;received=10.42.3.103;rport=5060 From: ;tag=1499347744 To: ;tag=as25e475f0 Call-ID: 2085551669-5060-17295@BA.EC.D.BAD CSeq: 192931 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog '2085551669-5060-17295@BA.EC.D.BAD' Method: SUBSCRIBE SNCA00-PH00*CLI>  -- Executing [17147240410@LocalSets:3] Set("SIP/000b825e3aa5-00000002", "CALLERID(num)=7144348010") in new stack SNCA00-PH00*CLI>  -- Executing [17147240410@LocalSets:4] Dial("SIP/000b825e3aa5-00000002", "SIP/wilogic-outbound/17147240410") in new stack SNCA00-PH00*CLI>  == Using SIP RTP CoS mark 5 SNCA00-PH00*CLI> Audio is at 20752 Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP SNCA00-PH00*CLI> Adding codec 100004 (alaw) to SDP Adding codec 100017 (testlaw) to SDP SNCA00-PH00*CLI> Reliably Transmitting (no NAT) to 38.96.36.14:5060: INVITE sip:17147240410@sws0.wlphone.net SIP/2.0 Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK3719d311 Max-Forwards: 70 From: ;tag=as6bda6c5c To: Contact: Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.1.0 Date: Wed, 26 Mar 2014 18:06:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "7144348010" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 240 v=0 o=root 474820536 474820536 IN IP4 38.96.43.209 s=Asterisk PBX 12.1.0 c=IN IP4 38.96.43.209 t=0 0 m=audio 20752 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:150 a=sendrecv --- SNCA00-PH00*CLI>  -- Called SIP/wilogic-outbound/17147240410 SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK7f95a59b;rport From: "asterisk" ;tag=as5c0e6f27 To: Contact: Call-ID: 784ddec05109eb6158d5b7d530526e30@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK7f95a59b;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as5c0e6f27 To: ;tag=as40f84be6 Call-ID: 784ddec05109eb6158d5b7d530526e30@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '784ddec05109eb6158d5b7d530526e30@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK3719d311;received=10.149.248.91 From: ;tag=as6bda6c5c To: ;tag=as33750136 Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75a5b52b" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI> Transmitting (no NAT) to 38.96.36.14:5060: ACK sip:17147240410@sws0.wlphone.net SIP/2.0 Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK3719d311 Max-Forwards: 70 From: ;tag=as6bda6c5c To: ;tag=as33750136 Contact: Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 12.1.0 Content-Length: 0 --- SNCA00-PH00*CLI> Audio is at 20752 Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP SNCA00-PH00*CLI> Adding codec 100004 (alaw) to SDP Adding codec 100017 (testlaw) to SDP SNCA00-PH00*CLI> Reliably Transmitting (no NAT) to 38.96.36.14:5060: INVITE sip:17147240410@sws0.wlphone.net SIP/2.0 Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK688f7252 Max-Forwards: 70 From: ;tag=as6bda6c5c To: Contact: Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 12.1.0 Proxy-Authorization: Digest username="32487144778734", realm="asterisk", algorithm=MD5, uri="sip:17147240410@sws0.wlphone.net", nonce="75a5b52b", response="7e43a1c0699d5a5bf364e4c4d9d5c333" Date: Wed, 26 Mar 2014 18:06:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "7144348010" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 240 v=0 o=root 474820536 474820537 IN IP4 38.96.43.209 s=Asterisk PBX 12.1.0 c=IN IP4 38.96.43.209 t=0 0 m=audio 20752 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:150 a=sendrecv --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK688f7252;received=10.149.248.91 From: ;tag=as6bda6c5c To: Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK48d43e44;rport From: "asterisk" ;tag=as28188664 To: Contact: Call-ID: 2b361bde5bba67ec1ac448370730aa2d@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK48d43e44;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as28188664 To: ;tag=as7c2bc3f2 Call-ID: 2b361bde5bba67ec1ac448370730aa2d@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2b361bde5bba67ec1ac448370730aa2d@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK1adabd51;rport From: "asterisk" ;tag=as3e7f44ec To: Contact: Call-ID: 1426e9945bd4843f73ee2277668a29ac@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> SNCA00-PH00*CLI> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK1adabd51;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as3e7f44ec To: ;tag=as033c10a3 Call-ID: 1426e9945bd4843f73ee2277668a29ac@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1426e9945bd4843f73ee2277668a29ac@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK688f7252;received=10.149.248.91 From: ;tag=as6bda6c5c To: ;tag=as63c0ce63 Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 181 v=0 o=root 24484 24484 IN IP4 38.96.36.14 s=session c=IN IP4 38.96.36.14 t=0 0 m=audio 38354 RTP/AVP 3 0 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - <-------------> --- (11 headers 9 lines) --- list_route: route/path hop: SNCA00-PH00*CLI> Found RTP audio format 3 SNCA00-PH00*CLI> Found RTP audio format 0 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 SNCA00-PH00*CLI> Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(gsm|ulaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw) SNCA00-PH00*CLI> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) SNCA00-PH00*CLI> Peer audio RTP is at port 38.96.36.14:38354 SNCA00-PH00*CLI>  -- SIP/wilogic-outbound-00000003 is making progress passing it to SIP/000b825e3aa5-00000002 SNCA00-PH00*CLI>  > 0x7ffcf800b5c0 -- Probation passed - setting RTP source address to 38.96.36.14:38354 SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.104:5060 ---> SUBSCRIBE sip:000b825c68cf@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.104:5060;branch=z9hG4bK1338022613;rport Route: From: ;tag=1436299137 To: Call-ID: 291814141-5060-10475@BA.EC.D.BAE CSeq: 124730 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (17 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 10.42.3.104:5060 (no NAT) Creating new subscription Sending to 10.42.3.104:5060 (no NAT) list_route: route/path hop: Found peer '000b825c68cf' for '000b825c68cf' from 10.42.3.104:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.104:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.3.104:5060;branch=z9hG4bK1338022613;received=10.42.3.104;rport=5060 From: ;tag=1436299137 To: ;tag=as0e4260bc Call-ID: 291814141-5060-10475@BA.EC.D.BAE CSeq: 124730 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="364c2fe0" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '291814141-5060-10475@BA.EC.D.BAE' in 32000 ms (Method: SUBSCRIBE) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.104:5060 ---> SUBSCRIBE sip:000b825c68cf@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.104:5060;branch=z9hG4bK85027586;rport Route: From: ;tag=1436299137 To: Call-ID: 291814141-5060-10475@BA.EC.D.BAE CSeq: 124731 SUBSCRIBE Contact: Authorization: Digest username="000b825c68cf", realm="asterisk", nonce="364c2fe0", uri="sip:000b825c68cf@10.42.0.20", response="2e5be82ceafa380f12adac4060522e65", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (18 headers 0 lines) --- SNCA00-PH00*CLI> Creating new subscription Sending to 10.42.3.104:5060 (no NAT) Found peer '000b825c68cf' for '000b825c68cf' from 10.42.3.104:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.104:5060 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 10.42.3.104:5060;branch=z9hG4bK85027586;received=10.42.3.104;rport=5060 From: ;tag=1436299137 To: ;tag=as0e4260bc Call-ID: 291814141-5060-10475@BA.EC.D.BAE CSeq: 124731 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog '291814141-5060-10475@BA.EC.D.BAE' Method: SUBSCRIBE SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.102:5060 ---> SUBSCRIBE sip:000b825f7360@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.102:5060;branch=z9hG4bK2030510468;rport Route: From: ;tag=1709270649 To: Call-ID: 1712259444-5060-17295@BA.EC.D.BAC CSeq: 192920 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (17 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 10.42.3.102:5060 (no NAT) SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.102:5060 (no NAT) list_route: route/path hop: Found peer '000b825f7360' for '000b825f7360' from 10.42.3.102:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.102:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.3.102:5060;branch=z9hG4bK2030510468;received=10.42.3.102;rport=5060 From: ;tag=1709270649 To: ;tag=as2946eefa Call-ID: 1712259444-5060-17295@BA.EC.D.BAC CSeq: 192920 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="22b942f8" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1712259444-5060-17295@BA.EC.D.BAC' in 32000 ms (Method: SUBSCRIBE) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.102:5060 ---> SUBSCRIBE sip:000b825f7360@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.102:5060;branch=z9hG4bK944698517;rport Route: From: ;tag=1709270649 To: Call-ID: 1712259444-5060-17295@BA.EC.D.BAC CSeq: 192921 SUBSCRIBE Contact: Authorization: Digest username="000b825f7360", realm="asterisk", nonce="22b942f8", uri="sip:000b825f7360@10.42.0.20", response="e2d19f67371a4367871787457ff8268f", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (18 headers 0 lines) --- SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.102:5060 (no NAT) Found peer '000b825f7360' for '000b825f7360' from 10.42.3.102:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.102:5060 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 10.42.3.102:5060;branch=z9hG4bK944698517;received=10.42.3.102;rport=5060 From: ;tag=1709270649 To: ;tag=as2946eefa Call-ID: 1712259444-5060-17295@BA.EC.D.BAC CSeq: 192921 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog '1712259444-5060-17295@BA.EC.D.BAC' Method: SUBSCRIBE SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> SUBSCRIBE sip:000b825e3aa5@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK2142548886;rport Route: From: ;tag=742476816 To: Call-ID: 230730301-5060-17311@BA.EC.D.BAB CSeq: 192920 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (17 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 10.42.3.101:5060 (no NAT) SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.101:5060 (no NAT) list_route: route/path hop: SNCA00-PH00*CLI> Found peer '000b825e3aa5' for '000b825e3aa5' from 10.42.3.101:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.101:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK2142548886;received=10.42.3.101;rport=5060 From: ;tag=742476816 To: ;tag=as32f788ef Call-ID: 230730301-5060-17311@BA.EC.D.BAB CSeq: 192920 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="346d8cae" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '230730301-5060-17311@BA.EC.D.BAB' in 32000 ms (Method: SUBSCRIBE) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> SUBSCRIBE sip:000b825e3aa5@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK1512500340;rport Route: From: ;tag=742476816 To: Call-ID: 230730301-5060-17311@BA.EC.D.BAB CSeq: 192921 SUBSCRIBE Contact: Authorization: Digest username="000b825e3aa5", realm="asterisk", nonce="346d8cae", uri="sip:000b825e3aa5@10.42.0.20", response="2c09490e5c1aecb014ef369871aad204", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (18 headers 0 lines) --- SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.101:5060 (no NAT) SNCA00-PH00*CLI> Found peer '000b825e3aa5' for '000b825e3aa5' from 10.42.3.101:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.101:5060 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK1512500340;received=10.42.3.101;rport=5060 From: ;tag=742476816 To: ;tag=as32f788ef Call-ID: 230730301-5060-17311@BA.EC.D.BAB CSeq: 192921 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog '230730301-5060-17311@BA.EC.D.BAB' Method: SUBSCRIBE SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK3d0aca67;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21769 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI> Really destroying SIP dialog '75028ae317b73e9d7d75efa17005113a@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.102:5060 ---> SUBSCRIBE sip:000b825f7360@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.102:5060;branch=z9hG4bK1905985761;rport Route: From: ;tag=1928422066 To: Call-ID: 197275413-5060-17296@BA.EC.D.BAC CSeq: 192930 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (17 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 10.42.3.102:5060 (no NAT) SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.102:5060 (no NAT) list_route: route/path hop: Found peer '000b825f7360' for '000b825f7360' from 10.42.3.102:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.102:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.3.102:5060;branch=z9hG4bK1905985761;received=10.42.3.102;rport=5060 From: ;tag=1928422066 To: ;tag=as6fa2ca3f Call-ID: 197275413-5060-17296@BA.EC.D.BAC CSeq: 192930 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="55f6edb9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '197275413-5060-17296@BA.EC.D.BAC' in 32000 ms (Method: SUBSCRIBE) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.102:5060 ---> SUBSCRIBE sip:000b825f7360@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.102:5060;branch=z9hG4bK1628921668;rport Route: From: ;tag=1928422066 To: Call-ID: 197275413-5060-17296@BA.EC.D.BAC CSeq: 192931 SUBSCRIBE Contact: Authorization: Digest username="000b825f7360", realm="asterisk", nonce="55f6edb9", uri="sip:000b825f7360@10.42.0.20", response="fcb28886820e3594d980ce57281063c6", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (18 headers 0 lines) --- SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.102:5060 (no NAT) Found peer '000b825f7360' for '000b825f7360' from 10.42.3.102:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.102:5060 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 10.42.3.102:5060;branch=z9hG4bK1628921668;received=10.42.3.102;rport=5060 From: ;tag=1928422066 To: ;tag=as6fa2ca3f Call-ID: 197275413-5060-17296@BA.EC.D.BAC CSeq: 192931 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog '197275413-5060-17296@BA.EC.D.BAC' Method: SUBSCRIBE SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ed4d37d", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK79d07b3d;rport From: "asterisk" ;tag=as3b2d8419 To: Contact: Call-ID: 3da5e62169dcdd527c2aaddf3dab42b7@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK79d07b3d;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as3b2d8419 To: ;tag=as74cda85e Call-ID: 3da5e62169dcdd527c2aaddf3dab42b7@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '3da5e62169dcdd527c2aaddf3dab42b7@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> SUBSCRIBE sip:000b825e3aa5@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK598649329;rport Route: From: ;tag=143143027 To: Call-ID: 1896684172-5060-17312@BA.EC.D.BAB CSeq: 192930 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (17 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 10.42.3.101:5060 (no NAT) SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.101:5060 (no NAT) list_route: route/path hop: Found peer '000b825e3aa5' for '000b825e3aa5' from 10.42.3.101:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.101:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK598649329;received=10.42.3.101;rport=5060 From: ;tag=143143027 To: ;tag=as756c2de4 Call-ID: 1896684172-5060-17312@BA.EC.D.BAB CSeq: 192930 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="53930705" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1896684172-5060-17312@BA.EC.D.BAB' in 32000 ms (Method: SUBSCRIBE) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> SUBSCRIBE sip:000b825e3aa5@10.42.0.20 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK557409960;rport Route: From: ;tag=143143027 To: Call-ID: 1896684172-5060-17312@BA.EC.D.BAB CSeq: 192931 SUBSCRIBE Contact: Authorization: Digest username="000b825e3aa5", realm="asterisk", nonce="53930705", uri="sip:000b825e3aa5@10.42.0.20", response="86e58f5a9c8cb16ffcd5b32b56bf5b49", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.1.14 Expires: 3600 Supported: replaces, path, timer Event: message-summary Accept: application/simple-message-summary Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (18 headers 0 lines) --- SNCA00-PH00*CLI> Creating new subscription SNCA00-PH00*CLI> Sending to 10.42.3.101:5060 (no NAT) Found peer '000b825e3aa5' for '000b825e3aa5' from 10.42.3.101:5060 SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.101:5060 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK557409960;received=10.42.3.101;rport=5060 From: ;tag=143143027 To: ;tag=as756c2de4 Call-ID: 1896684172-5060-17312@BA.EC.D.BAB CSeq: 192931 SUBSCRIBE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog '1896684172-5060-17312@BA.EC.D.BAB' Method: SUBSCRIBE SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK688f7252;received=10.149.248.91 From: ;tag=as6bda6c5c To: ;tag=as63c0ce63 Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI> list_route: route/path hop: SNCA00-PH00*CLI>  -- SIP/wilogic-outbound-00000003 is ringing SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 10.42.3.101:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK598328373;received=10.42.3.101;rport=5060 From: ;tag=34876174 To: ;tag=as62690954 Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 171 INVITE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK3d0aca67;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21769 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI> Really destroying SIP dialog '4ee01d8975bd36863518eba726abb99a@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="101730f7", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK66ecde88;rport From: "asterisk" ;tag=as52e7f359 To: Contact: Call-ID: 264f43f77012df6735198a5b6d522097@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK66ecde88;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as52e7f359 To: ;tag=as4a555b4a Call-ID: 264f43f77012df6735198a5b6d522097@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '264f43f77012df6735198a5b6d522097@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI> Really destroying SIP dialog '74603a025639735b041df32549d6f053@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI> Really destroying SIP dialog '7e302dce2d7ec14118d7236e6b6c71ea@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI> Really destroying SIP dialog '65b27db9031732dd1f734c483babdc79@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK3d0aca67;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21769 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI> Really destroying SIP dialog '48ab6c6b20509874527a6bd6661fd985@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="69f00296", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK0e586880;rport From: "asterisk" ;tag=as4c3a6a35 To: Contact: Call-ID: 23ceeab262a7358c2747593e3723ffce@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK0e586880;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as4c3a6a35 To: ;tag=as0fe2eb8f Call-ID: 23ceeab262a7358c2747593e3723ffce@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '23ceeab262a7358c2747593e3723ffce@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK1a19ccc6;rport From: "asterisk" ;tag=as6ff37a22 To: Contact: Call-ID: 47df7e5749e78b8650527abc152f5bba@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK1a19ccc6;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as6ff37a22 To: ;tag=as6625d626 Call-ID: 47df7e5749e78b8650527abc152f5bba@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '47df7e5749e78b8650527abc152f5bba@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK688f7252;received=10.149.248.91 From: ;tag=as6bda6c5c To: ;tag=as63c0ce63 Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 181 v=0 o=root 24484 24485 IN IP4 38.96.36.14 s=session c=IN IP4 38.96.36.14 t=0 0 m=audio 38354 RTP/AVP 3 0 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - <-------------> --- (11 headers 9 lines) --- SNCA00-PH00*CLI> Found RTP audio format 3 Found RTP audio format 0 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 SNCA00-PH00*CLI> Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(gsm|ulaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 38.96.36.14:38354 SNCA00-PH00*CLI> list_route: route/path hop: SNCA00-PH00*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 38.96.36.14:5060 SNCA00-PH00*CLI> Transmitting (no NAT) to 38.96.36.14:5060: ACK sip:17147240410@38.96.36.14 SIP/2.0 Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK68ee422c Max-Forwards: 70 From: ;tag=as6bda6c5c To: ;tag=as63c0ce63 Contact: Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 12.1.0 Content-Length: 0 --- SNCA00-PH00*CLI>  -- SIP/wilogic-outbound-00000003 answered SIP/000b825e3aa5-00000002 SNCA00-PH00*CLI> Audio is at 26398 Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP SNCA00-PH00*CLI>  <--- Reliably Transmitting (no NAT) to 10.42.3.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK598328373;received=10.42.3.101;rport=5060 From: ;tag=34876174 To: ;tag=as62690954 Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 171 INVITE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 269 v=0 o=root 421206821 421206821 IN IP4 10.42.0.20 s=Asterisk PBX 12.1.0 c=IN IP4 10.42.0.20 t=0 0 m=audio 26398 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=ptime:20 a=maxptime:150 a=sendrecv <------------> SNCA00-PH00*CLI>  -- Channel SIP/000b825e3aa5-00000002 joined 'simple_bridge' basic-bridge <9bbb23f1-a4dd-4a4e-8dd3-7a7511b55426> SNCA00-PH00*CLI>  -- Channel SIP/wilogic-outbound-00000003 joined 'simple_bridge' basic-bridge <9bbb23f1-a4dd-4a4e-8dd3-7a7511b55426> > Bridge 9bbb23f1-a4dd-4a4e-8dd3-7a7511b55426: switching from simple_bridge technology to native_rtp set_destination: Parsing for address/port to send to set_destination: set destination to 38.96.36.14:5060 Audio is at 20752 Adding codec 100003 (ulaw) to SDP Reliably Transmitting (no NAT) to 38.96.36.14:5060: INVITE sip:17147240410@38.96.36.14 SIP/2.0 Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK0d5f910f Max-Forwards: 70 From: ;tag=as6bda6c5c To: ;tag=as63c0ce63 Contact: Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "7144348010" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 190 v=0 o=root 474820536 474820538 IN IP4 10.42.3.101 s=Asterisk PBX 12.1.0 c=IN IP4 10.42.3.101 t=0 0 m=audio 5004 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:150 a=sendrecv --- SNCA00-PH00*CLI>  > 0x7ffcf800b5c0 -- Probation passed - setting RTP source address to 38.96.36.14:38354 SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK0d5f910f;received=10.149.248.91 From: ;tag=as6bda6c5c To: ;tag=as63c0ce63 Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 158 v=0 o=root 24484 24486 IN IP4 38.96.36.14 s=session c=IN IP4 38.96.36.14 t=0 0 m=audio 38354 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - <-------------> --- (11 headers 8 lines) --- SNCA00-PH00*CLI> Found RTP audio format 0 Found audio description format PCMU for ID 0 SNCA00-PH00*CLI> Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 38.96.36.14:38354 SNCA00-PH00*CLI> set_destination: Parsing for address/port to send to SNCA00-PH00*CLI> set_destination: set destination to 38.96.36.14:5060 Transmitting (no NAT) to 38.96.36.14:5060: ACK sip:17147240410@38.96.36.14 SIP/2.0 Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK34e2a191 Max-Forwards: 70 From: ;tag=as6bda6c5c To: ;tag=as63c0ce63 Contact: Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 12.1.0 Content-Length: 0 --- SNCA00-PH00*CLI>  > 0x7ffcf800b5c0 -- Probation passed - setting RTP source address to 38.96.36.14:38354 SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> ACK sip:17147240410@10.42.0.20:5060 SIP/2.0 Via: SIP/2.0/UDP 10.42.3.101:5060;branch=z9hG4bK2107748084;rport From: ;tag=34876174 To: ;tag=as62690954 Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 171 ACK Contact: X-Grandstream-PBX: true Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXP2160 1.0.1.14 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (13 headers 0 lines) --- set_destination: Parsing for address/port to send to SNCA00-PH00*CLI> set_destination: set destination to 10.42.3.101:5060 SNCA00-PH00*CLI> Audio is at 26398 Adding codec 100003 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP SNCA00-PH00*CLI> Reliably Transmitting (no NAT) to 10.42.3.101:5060: INVITE sip:000b825e3aa5@10.42.3.101:5060 SIP/2.0 Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK25f3245a;rport Max-Forwards: 70 From: ;tag=as62690954 To: ;tag=34876174 Contact: Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 102 INVITE User-Agent: Asterisk PBX 12.1.0 Session-Expires: 1800;refresher=uac Min-SE: 90 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: 247 v=0 o=root 421206821 421206822 IN IP4 38.96.36.14 s=Asterisk PBX 12.1.0 c=IN IP4 38.96.36.14 t=0 0 m=audio 38354 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK25f3245a;rport=5060 From: ;tag=as62690954 To: ;tag=34876174 Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 102 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP2160 1.0.1.14 Require: timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK25f3245a;rport=5060 From: ;tag=as62690954 To: ;tag=34876174 Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 102 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP2160 1.0.1.14 Session-Expires: 1800;refresher=uac Require: timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 217 v=0 o=000b825e3aa5 8000 8001 IN IP4 10.42.3.101 s=SIP Call c=IN IP4 10.42.3.101 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (14 headers 11 lines) --- SNCA00-PH00*CLI> Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 SNCA00-PH00*CLI> Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.42.3.101:5004 SNCA00-PH00*CLI> set_destination: Parsing for address/port to send to SNCA00-PH00*CLI> set_destination: set destination to 10.42.3.101:5060 Transmitting (no NAT) to 10.42.3.101:5060: ACK sip:000b825e3aa5@10.42.3.101:5060 SIP/2.0 Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK1569885b;rport Max-Forwards: 70 From: ;tag=as62690954 To: ;tag=34876174 Contact: Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 102 ACK User-Agent: Asterisk PBX 12.1.0 Content-Length: 0 --- SNCA00-PH00*CLI> set_destination: Parsing for address/port to send to SNCA00-PH00*CLI> set_destination: set destination to 38.96.36.14:5060 Audio is at 20752 Adding codec 100003 (ulaw) to SDP SNCA00-PH00*CLI> Reliably Transmitting (no NAT) to 38.96.36.14:5060: INVITE sip:17147240410@38.96.36.14 SIP/2.0 Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK7ad56c57 Max-Forwards: 70 From: ;tag=as6bda6c5c To: ;tag=as63c0ce63 Contact: Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 105 INVITE User-Agent: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "7144348010" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 190 v=0 o=root 474820536 474820539 IN IP4 10.42.3.101 s=Asterisk PBX 12.1.0 c=IN IP4 10.42.3.101 t=0 0 m=audio 5004 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:150 a=sendrecv --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK7ad56c57;received=10.149.248.91 From: ;tag=as6bda6c5c To: ;tag=as63c0ce63 Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 158 v=0 o=root 24484 24487 IN IP4 38.96.36.14 s=session c=IN IP4 38.96.36.14 t=0 0 m=audio 38354 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - <-------------> --- (11 headers 8 lines) --- SNCA00-PH00*CLI> Found RTP audio format 0 Found audio description format PCMU for ID 0 SNCA00-PH00*CLI> Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 38.96.36.14:38354 SNCA00-PH00*CLI> set_destination: Parsing for address/port to send to SNCA00-PH00*CLI> set_destination: set destination to 38.96.36.14:5060 SNCA00-PH00*CLI> Transmitting (no NAT) to 38.96.36.14:5060: ACK sip:17147240410@38.96.36.14 SIP/2.0 Via: SIP/2.0/UDP 38.96.43.209:5060;branch=z9hG4bK7cfdff3b Max-Forwards: 70 From: ;tag=as6bda6c5c To: ;tag=as63c0ce63 Contact: Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 105 ACK User-Agent: Asterisk PBX 12.1.0 Content-Length: 0 --- SNCA00-PH00*CLI>  > 0x7ffce4083800 -- Probation passed - setting RTP source address to 10.42.3.101:5004 SNCA00-PH00*CLI>  > 0x7ffcf800b5c0 -- Probation passed - setting RTP source address to 38.96.36.14:38354 SNCA00-PH00*CLI> Really destroying SIP dialog '6257c61834d5f08a073a585d5daec733@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI> Really destroying SIP dialog '3ba34f5b0d2576cb556fbafe598898f1@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI> Really destroying SIP dialog '6fe0471e12d5fe39170b2cbb0fda997b@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK3d0aca67;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21769 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK6e975fe5;rport From: "asterisk" ;tag=as7f575704 To: Contact: Call-ID: 5d6d0b505c5e454f7e533eef5595f596@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK6e975fe5;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as7f575704 To: ;tag=as2b79db3c Call-ID: 5d6d0b505c5e454f7e533eef5595f596@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '5d6d0b505c5e454f7e533eef5595f596@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK480be0b2;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21807 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6915ab9e", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI> Really destroying SIP dialog '7b5ef72117026e2545245ba07a0c5d37@38.96.36.14' Method: OPTIONS SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK0addab1a;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21770 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK78a38b2e;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21808 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="14505c58", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK78a38b2e;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21808 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK2a6ed6a5;rport From: "asterisk" ;tag=as024cf99a To: Contact: Call-ID: 69aa3c3c73ea02850eeab5c719a9f2e1@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK2a6ed6a5;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as024cf99a To: ;tag=as5adc7886 Call-ID: 69aa3c3c73ea02850eeab5c719a9f2e1@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '69aa3c3c73ea02850eeab5c719a9f2e1@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK0addab1a;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21770 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK6c1ee4a6;rport From: "asterisk" ;tag=as70515f90 To: Contact: Call-ID: 37159e6273421da000d82094624e623a@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK6c1ee4a6;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as70515f90 To: ;tag=as09236535 Call-ID: 37159e6273421da000d82094624e623a@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '37159e6273421da000d82094624e623a@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK78a38b2e;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21808 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK78a38b2e;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21808 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="08fde370", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> BYE sip:7144348010@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK7bcee3da;rport From: ;tag=as63c0ce63 To: ;tag=as6bda6c5c Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '00ba967c5169adf739d3558541f87594@38.96.43.209:5060' in 32000 ms (Method: BYE) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK7bcee3da;received=38.96.36.14;rport=5060 From: ;tag=as63c0ce63 To: ;tag=as6bda6c5c Call-ID: 00ba967c5169adf739d3558541f87594@38.96.43.209:5060 CSeq: 102 BYE Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> SNCA00-PH00*CLI>  -- Channel SIP/wilogic-outbound-00000003 left 'native_rtp' basic-bridge <9bbb23f1-a4dd-4a4e-8dd3-7a7511b55426> SNCA00-PH00*CLI> set_destination: Parsing for address/port to send to SNCA00-PH00*CLI> set_destination: set destination to 10.42.3.101:5060 SNCA00-PH00*CLI> Audio is at 26398 Adding codec 100003 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP SNCA00-PH00*CLI> Reliably Transmitting (no NAT) to 10.42.3.101:5060: INVITE sip:000b825e3aa5@10.42.3.101:5060 SIP/2.0 Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK17e10a22;rport Max-Forwards: 70 From: ;tag=as62690954 To: ;tag=34876174 Contact: Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 103 INVITE User-Agent: Asterisk PBX 12.1.0 Session-Expires: 1800;refresher=uac Min-SE: 90 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: 245 v=0 o=root 421206821 421206823 IN IP4 10.42.0.20 s=Asterisk PBX 12.1.0 c=IN IP4 10.42.0.20 t=0 0 m=audio 26398 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- SNCA00-PH00*CLI>  -- Channel SIP/000b825e3aa5-00000002 left 'native_rtp' basic-bridge <9bbb23f1-a4dd-4a4e-8dd3-7a7511b55426> SNCA00-PH00*CLI>  == Spawn extension (LocalSets, 17147240410, 4) exited non-zero on 'SIP/000b825e3aa5-00000002' SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '208247511-5060-17310@BA.EC.D.BAB' in 32000 ms (Method: ACK) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK4ddda6c6;rport From: "asterisk" ;tag=as2702223c To: Contact: Call-ID: 7130fe0d56e3d5df5117abbb5364a555@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) SNCA00-PH00*CLI>  <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK4ddda6c6;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as2702223c To: ;tag=as639cf1ae Call-ID: 7130fe0d56e3d5df5117abbb5364a555@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '7130fe0d56e3d5df5117abbb5364a555@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK17e10a22;rport=5060 From: ;tag=as62690954 To: ;tag=34876174 Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 103 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP2160 1.0.1.14 Require: timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK17e10a22;rport=5060 From: ;tag=as62690954 To: ;tag=34876174 Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 103 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP2160 1.0.1.14 Session-Expires: 1800;refresher=uac Require: timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 217 v=0 o=000b825e3aa5 8000 8002 IN IP4 10.42.3.101 s=SIP Call c=IN IP4 10.42.3.101 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (14 headers 11 lines) --- SNCA00-PH00*CLI> Found RTP audio format 0 Found RTP audio format 101 SNCA00-PH00*CLI> Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 SNCA00-PH00*CLI> Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.42.3.101:5004 SNCA00-PH00*CLI> set_destination: Parsing for address/port to send to SNCA00-PH00*CLI> set_destination: set destination to 10.42.3.101:5060 SNCA00-PH00*CLI> Transmitting (no NAT) to 10.42.3.101:5060: ACK sip:000b825e3aa5@10.42.3.101:5060 SIP/2.0 Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK12bc2ef1;rport Max-Forwards: 70 From: ;tag=as62690954 To: ;tag=34876174 Contact: Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 103 ACK User-Agent: Asterisk PBX 12.1.0 Content-Length: 0 --- SNCA00-PH00*CLI> set_destination: Parsing for address/port to send to SNCA00-PH00*CLI> set_destination: set destination to 10.42.3.101:5060 SNCA00-PH00*CLI> Reliably Transmitting (no NAT) to 10.42.3.101:5060: BYE sip:000b825e3aa5@10.42.3.101:5060 SIP/2.0 Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK57927e48;rport Max-Forwards: 70 From: ;tag=as62690954 To: ;tag=34876174 Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 104 BYE User-Agent: Asterisk PBX 12.1.0 Proxy-Authorization: Digest username="000b825e3aa5", realm="asterisk", algorithm=MD5, uri="sip:10.42.0.20", nonce="500e8dad", response="c63d41e32349f3fd77b86175dd60cba1" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '208247511-5060-17310@BA.EC.D.BAB' in 32000 ms (Method: ACK) SNCA00-PH00*CLI>  <--- SIP read from UDP:10.42.3.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK57927e48;rport=5060 From: ;tag=as62690954 To: ;tag=34876174 Call-ID: 208247511-5060-17310@BA.EC.D.BAB CSeq: 104 BYE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP2160 1.0.1.14 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '208247511-5060-17310@BA.EC.D.BAB' Method: ACK SNCA00-PH00*CLI>  <--- SIP read from UDP:64.154.41.158:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK0addab1a;received=100.32.93.142;rport=5060 From: ;tag=as351fd883 To: ;tag=as66ccb078 Call-ID: 676c1c2f54e4c50403e4b93124c2d040@10.42.0.20 CSeq: 21770 REGISTER Server: FPBX-2.9.0(1.8.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4559b7b3", stale=true Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK78a38b2e;received=10.149.248.91 From: ;tag=as1e717ae7 To: ;tag=as0568e4f9 Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21808 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="073988b8", stale=true Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> OPTIONS sip:32847144778734@38.96.43.209:5060 SIP/2.0 Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK424e84e0;rport From: "asterisk" ;tag=as3b714637 To: Contact: Call-ID: 03e61c4054431fb968937c111e62a4ff@38.96.36.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Mar 2014 18:07:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SNCA00-PH00*CLI> Sending to 38.96.36.14:5060 (no NAT) SNCA00-PH00*CLI> Looking for 32847144778734 in unauthenticated (domain 38.96.43.209) <--- Transmitting (no NAT) to 38.96.36.14:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 38.96.36.14:5060;branch=z9hG4bK424e84e0;received=38.96.36.14;rport=5060 From: "asterisk" ;tag=as3b714637 To: ;tag=as399ff18c Call-ID: 03e61c4054431fb968937c111e62a4ff@38.96.36.14 CSeq: 102 OPTIONS Server: Asterisk PBX 12.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> SNCA00-PH00*CLI> Scheduling destruction of SIP dialog '03e61c4054431fb968937c111e62a4ff@38.96.36.14' in 32000 ms (Method: OPTIONS) SNCA00-PH00*CLI>  <--- SIP read from UDP:38.96.36.14:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.42.0.20:5060;branch=z9hG4bK78a38b2e;received=10.149.248.91 From: ;tag=as1e717ae7 To: Call-ID: 041f18b70317c7080822673b5d65b5ee@10.42.0.20 CSeq: 21808 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SNCA00-PH00*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups