Asterisk 13.0.0, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 13.0.0 currently running on antarespbx-camus (pid = 3750) antarespbx-camus*CLI>  <--- SIP read from UDP:192.168.130.104:64251 ---> <-------------> antarespbx-camus*CLI>  Really destroying SIP dialog '9eJv0MT8qnzvkM1KyJ9EIA..' Method: REGISTER antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> OPTIONS sip:s@192.168.130.68:5060 SIP/2.0 Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK2aa7963e;rport Max-Forwards: 70 From: "asterisk" ;tag=as34b6b85f To: Contact: Call-ID: 0af397bd405311833d9bea5c65b21f05@50.116.31.15:5060 CSeq: 102 OPTIONS User-Agent: VozOrgSIP1 Date: Fri, 16 Jan 2015 22:59:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 50.116.31.15:5060 (no NAT) Looking for s in incoming (domain 192.168.130.68) <--- Transmitting (no NAT) to 50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK2aa7963e;received=50.116.31.15;rport=5060 From: "asterisk" ;tag=as34b6b85f To: ;tag=as6bd10d2d Call-ID: 0af397bd405311833d9bea5c65b21f05@50.116.31.15:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0af397bd405311833d9bea5c65b21f05@50.116.31.15:5060' in 32000 ms (Method: OPTIONS) antarespbx-camus*CLI> Retransmitting #6 (no NAT) to 192.168.130.102:5060: INVITE sip:93147920634@192.168.130.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK087b29c4 Max-Forwards: 70 From: "Antares Soporte" ;tag=as39dc455f To: Contact: Call-ID: 750f5114598d73c90c0e77234fea8eb5@192.168.130.68:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 22:59:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 302 v=0 o=root 2057435009 2057435009 IN IP4 192.168.130.68 s=Asterisk PBX 13.0.0 c=IN IP4 192.168.130.68 t=0 0 m=audio 18882 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- antarespbx-camus*CLI> Scheduling destruction of SIP dialog '750f5114598d73c90c0e77234fea8eb5@192.168.130.68:5060' in 32000 ms (Method: INVITE) -- SIP/camus-00000015 is circuit-busy Scheduling destruction of SIP dialog '750f5114598d73c90c0e77234fea8eb5@192.168.130.68:5060' in 32000 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/1/0) antarespbx-camus*CLI> [Jan 16 17:59:58] WARNING[3823]: chan_sip.c:4042 retrans_pkt: Retransmission timeout reached on transmission 750f5114598d73c90c0e77234fea8eb5@192.168.130.68:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32001ms with no response Really destroying SIP dialog '750f5114598d73c90c0e77234fea8eb5@192.168.130.68:5060' Method: INVITE antarespbx-camus*CLI>  -- AGI Script Executing Application: (Goto) Options: (hangup-context,called,1) -- Goto (hangup-context,called,1) antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=camus&dialrule=34&delete_initial_digits=2&delete_final_digits=0&prefix=&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: 'DIALSTATUS ***CONGESTION' antarespbx-camus*CLI>  -- AGI Script Executing Application: (Playback) Options: (on-busy) Audio is at 14682 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBkflNtkae1HvT3cqRAheMEHopUBrtNZF;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=ZkoMuqJJTMyyBEUnxtoQ To: ;tag=as1dee0a9e Call-ID: cc5b3b92-f051-5e91-4936-0a9892ed4eb4 CSeq: 49913 INVITE Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 893 v=0 o=root 722670115 722670115 IN IP4 192.168.130.68 s=Asterisk PBX 13.0.0 c=IN IP4 192.168.130.68 t=0 0 m=audio 14682 RTP/SAVPF 0 8 3 126 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-16 a=ptime:20 a=maxptime:150 a=ice-ufrag:44ed69346047bfd767ef8f17134436b0 a=ice-pwd:1c73567a24f002982f02b7370c9bc007 a=candidate:Hc0a88244 1 UDP 2130706431 192.168.130.68 14682 typ host a=candidate:Sb53ef970 1 UDP 1694498815 181.62.249.112 14682 typ srflx raddr 192.168.130.68 rport 14682 a=candidate:Hc0a88244 2 UDP 2130706430 192.168.130.68 14683 typ host a=candidate:Sb53ef970 2 UDP 1694498814 181.62.249.112 14683 typ srflx raddr 192.168.130.68 rport 14683 a=connection:new a=setup:active a=fingerprint:SHA-256 12:1E:9F:B5:CB:6A:4A:BF:EB:2A:81:96:B5:EA:F4:80:FF:7D:27:65:75:BD:DB:86:98:AD:5E:7E:2F:16:3F:2C a=sendrecv <------------> antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> ACK sip:9893147920634@192.168.130.68:5060;transport=WS SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKkdkTvPzWgEKSbH5HSCGB;rport From: "Antares Soporte";tag=ZkoMuqJJTMyyBEUnxtoQ To: ;tag=as1dee0a9e Contact: "Antares Soporte";+g.oma.sip-im;+sip.ice;language="es,fr" Call-ID: cc5b3b92-f051-5e91-4936-0a9892ed4eb4 CSeq: 49913 ACK Content-Length: 0 Route: Max-Forwards: 70 Authorization: Digest username="er",realm="192.168.130.68",nonce="17d17d07",uri="sip:9893147920634@192.168.130.68:5060;transport=WS",response="bc85e3a13368071065dc7a2c79bf8e46",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: ER Technology LTDA <-------------> --- (13 headers 0 lines) --- antarespbx-camus*CLI>  -- Playing 'on-busy.slin' (language 'es') antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> REGISTER sip:192.168.130.68 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKfhTvxdHo2gTu1qvTQRy7MduOEiwyk1bf;rport From: "Antares Soporte";tag=aJEXkgub0eq4X2lwxkJc To: "Antares Soporte" Contact: "Antares Soporte";expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr" Call-ID: df282f12-a3ea-dc9d-8a07-08673dce3527 CSeq: 61172 REGISTER Content-Length: 0 Route: Max-Forwards: 70 Authorization: Digest username="er",realm="192.168.130.68",nonce="3d1cca90",uri="sip:192.168.130.68",response="e31f065daa8d8c612c1d8c7b28d78a0d",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: ER Technology LTDA <-------------> --- (13 headers 0 lines) --- <--- Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKfhTvxdHo2gTu1qvTQRy7MduOEiwyk1bf;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=aJEXkgub0eq4X2lwxkJc To: "Antares Soporte";tag=as25e16ac7 Call-ID: df282f12-a3ea-dc9d-8a07-08673dce3527 CSeq: 61172 REGISTER Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="192.168.130.68", nonce="08bac52d" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'df282f12-a3ea-dc9d-8a07-08673dce3527' in 32000 ms (Method: REGISTER) antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> REGISTER sip:192.168.130.68 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKr5GNuQDeG2k4UlGy1ExGUg5MDVs2563m;rport From: "Antares Soporte";tag=aJEXkgub0eq4X2lwxkJc To: "Antares Soporte" Contact: "Antares Soporte";expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr" Call-ID: df282f12-a3ea-dc9d-8a07-08673dce3527 CSeq: 61173 REGISTER Content-Length: 0 Route: Max-Forwards: 70 Authorization: Digest username="er",realm="192.168.130.68",nonce="08bac52d",uri="sip:192.168.130.68",response="0165b1c6baa5bdd722d46a2f6620f7a2",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: ER Technology LTDA <-------------> antarespbx-camus*CLI> --- (13 headers 0 lines) --- antarespbx-camus*CLI> Reliably Transmitting (NAT) to 192.168.130.102:57026: OPTIONS sip:er@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.130.68:5060;branch=z9hG4bK60623929;rport Max-Forwards: 70 From: "asterisk" ;tag=as74af9df3 To: Contact: Call-ID: 32b8d8ea3e99129d40f1869e4b0e570d@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 22:59:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKr5GNuQDeG2k4UlGy1ExGUg5MDVs2563m;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=aJEXkgub0eq4X2lwxkJc To: "Antares Soporte";tag=as25e16ac7 Call-ID: df282f12-a3ea-dc9d-8a07-08673dce3527 CSeq: 61173 REGISTER Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 200 Contact: ;expires=200 Date: Fri, 16 Jan 2015 22:59:59 GMT Content-Length: 0 <------------> antarespbx-camus*CLI> Scheduling destruction of SIP dialog 'df282f12-a3ea-dc9d-8a07-08673dce3527' in 32000 ms (Method: REGISTER) antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> SIP/2.0 405 Method Not Allowed Via: SIP/2.0/WS 192.168.130.68:5060;rport=5060;branch=z9hG4bK60623929 From: "asterisk";tag=as74af9df3 To: Call-ID: 32b8d8ea3e99129d40f1869e4b0e570d@192.168.130.68:5060 CSeq: 102 OPTIONS Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (7 headers 0 lines) --- antarespbx-camus*CLI> [Jan 16 17:59:59] ERROR[19896][C-0000000a]: utils.c:1371 ast_carefulwrite: write() returned error: Broken pipe antarespbx-camus*CLI>  -- AGI Script agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=camus&dialrule=34&delete_initial_digits=2&delete_final_digits=0&prefix=&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid= completed, returning 0 antarespbx-camus*CLI> [Jan 16 17:59:59] ERROR[19896][C-0000000a]: utils.c:1371 ast_carefulwrite: write() returned error: Broken pipe antarespbx-camus*CLI>  -- Executing [called@hangup-context:1] AGI("SIP/er-00000014", "agi://127.0.0.1:4573/?app=posthangup&recordid=None&duration=1") in new stack antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=posthangup&recordid=None&duration=1: u'UNIQUEID:1421449166.40' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=posthangup&recordid=None&duration=1: u'callerid:er' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=posthangup&recordid=None&duration=1: u'calleridname:Antares Soporte' antarespbx-camus*CLI>  -- AGI Script Executing Application: (Hangup) Options: () == Spawn extension (hangup-context, called, 1) exited non-zero on 'SIP/er-00000014' Scheduling destruction of SIP dialog 'cc5b3b92-f051-5e91-4936-0a9892ed4eb4' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 192.168.130.102:57026: BYE sip:er@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.130.68:5060;branch=z9hG4bK32d7367c;rport Max-Forwards: 70 From: ;tag=as1dee0a9e To: "Antares Soporte";tag=ZkoMuqJJTMyyBEUnxtoQ Call-ID: cc5b3b92-f051-5e91-4936-0a9892ed4eb4 CSeq: 102 BYE User-Agent: Asterisk PBX 13.0.0 Proxy-Authorization: Digest username="er", realm="192.168.130.68", algorithm=MD5, uri="sip:192.168.130.68", nonce="17d17d07", response="4304df48296ce0df90310d293b589a9c" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- antarespbx-camus*CLI>  == MixMonitor close filestream (mixed) == End MixMonitor Recording SIP/er-00000014 antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.130.68:5060;rport=5060;branch=z9hG4bK32d7367c From: ;tag=as1dee0a9e To: "Antares Soporte";tag=ZkoMuqJJTMyyBEUnxtoQ Contact: Call-ID: cc5b3b92-f051-5e91-4936-0a9892ed4eb4 CSeq: 102 BYE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived antarespbx-camus*CLI> Really destroying SIP dialog '32b8d8ea3e99129d40f1869e4b0e570d@192.168.130.68:5060' Method: OPTIONS Really destroying SIP dialog 'cc5b3b92-f051-5e91-4936-0a9892ed4eb4' Method: INVITE antarespbx-camus*CLI> Reliably Transmitting (no NAT) to 50.116.31.15:5060: OPTIONS sip:sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK01d2e396 Max-Forwards: 70 From: "asterisk" ;tag=as332311df To: Contact: Call-ID: 1e1f86437da655a162ee51ed180fb3a5@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:00:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI> Reliably Transmitting (no NAT) to 208.66.193.46:5060: OPTIONS sip:sip.ipsofactum.com SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK439c8926 Max-Forwards: 70 From: "asterisk" ;tag=as4fe1ec1e To: Contact: Call-ID: 13cb82811d73dcf100443a7264cdefb7@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:00:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK01d2e396;received=181.62.249.112;rport=2048 From: "asterisk" ;tag=as332311df To: ;tag=as64ea2537 Call-ID: 1e1f86437da655a162ee51ed180fb3a5@192.168.130.68:5060 CSeq: 102 OPTIONS Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '1e1f86437da655a162ee51ed180fb3a5@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI>  <--- SIP read from UDP:208.66.193.46:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK439c8926;received=181.62.249.112;rport=2048 From: "asterisk" ;tag=as4fe1ec1e To: ;tag=as37828d01 Call-ID: 13cb82811d73dcf100443a7264cdefb7@192.168.130.68:5060 CSeq: 102 OPTIONS Server: Ipsoswitch - VoIP Powered by IPSOFACTUM Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '13cb82811d73dcf100443a7264cdefb7@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Reliably Transmitting (NAT) to 173.192.95.227:5060: OPTIONS sip:173.192.95.227 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK350eaaa5;rport Max-Forwards: 70 From: "asterisk" ;tag=as3d23a66b To: Contact: Call-ID: 4f4e98275f83a98a5a07fe705757140b@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:00:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:173.192.95.227:5060 ---> SIP/2.0 200 OK CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK350eaaa5;rport From: "asterisk" ;tag=as3d23a66b Call-ID: 4f4e98275f83a98a5a07fe705757140b@192.168.130.68:5060 To: ;tag=160156151738 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (8 headers 0 lines) --- antarespbx-camus*CLI> Really destroying SIP dialog '4f4e98275f83a98a5a07fe705757140b@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Reliably Transmitting (NAT) to 173.255.199.156:5060: OPTIONS sip:173.255.199.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK25a3811b;rport Max-Forwards: 70 From: "asterisk" ;tag=as7ba94190 To: Contact: Call-ID: 3cdad381156d4a617a9b56fe53580dac@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:00:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:173.255.199.156:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK25a3811b;received=181.62.249.112;rport=2048 From: "asterisk" ;tag=as7ba94190 To: ;tag=as248bfcb6 Call-ID: 3cdad381156d4a617a9b56fe53580dac@192.168.130.68:5060 CSeq: 102 OPTIONS Server: VozOrgSIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (12 headers 0 lines) --- antarespbx-camus*CLI> Really destroying SIP dialog '3cdad381156d4a617a9b56fe53580dac@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Really destroying SIP dialog '0c0cec7354568a4a37393788208cacb0@50.116.31.15:5060' Method: OPTIONS antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> INVITE sip:93147920634@192.168.130.68 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKUl6kDxievzWqzxmKlds9Ufw5fcCBwPpO;rport From: "Antares Soporte";tag=1fzemhoA46PBe3wN06B1 To: Contact: "Antares Soporte";+g.oma.sip-im;+sip.ice;language="es,fr" Call-ID: 82a9aeff-279c-2414-673c-0882dfcd2600 CSeq: 61650 INVITE Content-Type: application/sdp Content-Length: 2766 Route: Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: ER Technology LTDA v=0 o=- 5725125286898753000 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS O7epInWmLV3OdAKa3KcZILmnNrFdPH4pnkf0 m=audio 35686 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 181.62.249.112 b=AS:64 a=rtcp:35686 IN IP4 181.62.249.112 a=candidate:252621849 1 udp 2122260223 192.168.130.102 35686 typ host generation 0 a=candidate:252621849 2 udp 2122260223 192.168.130.102 35686 typ host generation 0 a=candidate:4062413514 1 udp 2122194687 192.168.122.1 36454 typ host generation 0 a=candidate:4062413514 2 udp 2122194687 192.168.122.1 36454 typ host generation 0 a=candidate:498801655 1 udp 2122129151 192.168.130.101 42792 typ host generation 0 a=candidate:498801655 2 udp 2122129151 192.168.130.101 42792 typ host generation 0 a=candidate:1724185560 1 udp 1686052607 181.62.249.112 35686 typ srflx raddr 192.168.130.102 rport 35686 generation 0 a=candidate:1724185560 2 udp 1686052607 181.62.249.112 35686 typ srflx raddr 192.168.130.102 rport 35686 generation 0 a=candidate:1953584182 1 udp 1685921535 181.62.249.112 42792 typ srflx raddr 192.168.130.101 rport 42792 generation 0 a=candidate:1953584182 2 udp 1685921535 181.62.249.112 42792 typ srflx raddr 192.168.130.101 rport 42792 generation 0 a=candidate:1099778281 1 tcp 1518280447 192.168.130.102 0 typ host tcptype active generation 0 a=candidate:1099778281 2 tcp 1518280447 192.168.130.102 0 typ host tcptype active generation 0 a=candidate:3164634682 1 tcp 1518214911 192.168.122.1 0 typ host tcptype active generation 0 a=candidate:3164634682 2 tcp 1518214911 192.168.122.1 0 typ host tcptype active generation 0 a=candidate:1396238087 1 tcp 1518149375 192.168.130.101 0 typ host tcptype active generation 0 a=candidate:1396238087 2 tcp 1518149375 192.168.130.101 0 typ host tcptype active generation 0 a=ice-ufrag:nEyEH5BsjuT1rizJ a=ice-pwd:aZp+RM4Kv9VnHzcRJlEaEu+J a=ice-options:google-ice a=fingerprint:sha-256 B4:3C:60:B0:60:86:89:8D:0B:A9:49:08:E8:F9:98:D3:7F:E5:9F:FD:5B:A6:C1:BE:B9:3A:B9:1D:C3:4F:32:91 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:528907053 cname:6UKlGaci38ADo7cn a=ssrc:528907053 msid:O7epInWmLV3OdAKa3KcZILmnNrFdPH4pnkf0 c17b9cc6-063e-481b-9934-a13f74b89a93 a=ssrc:528907053 mslabel:O7epInWmLV3OdAKa3KcZILmnNrFdPH4pnkf0 a=ssrc:528907053 label:c17b9cc6-063e-481b-9934-a13f74b89a93 <-------------> antarespbx-camus*CLI> --- (13 headers 51 lines) --- antarespbx-camus*CLI> Using INVITE request as basis request - 82a9aeff-279c-2414-673c-0882dfcd2600 antarespbx-camus*CLI> Found peer 'er' for 'er' from 192.168.130.102:57026 antarespbx-camus*CLI>  <--- Reliably Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKUl6kDxievzWqzxmKlds9Ufw5fcCBwPpO;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=1fzemhoA46PBe3wN06B1 To: ;tag=as1996e832 Call-ID: 82a9aeff-279c-2414-673c-0882dfcd2600 CSeq: 61650 INVITE Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="192.168.130.68", nonce="519eed2c" Content-Length: 0 <------------> antarespbx-camus*CLI> Scheduling destruction of SIP dialog '82a9aeff-279c-2414-673c-0882dfcd2600' in 6400 ms (Method: INVITE) antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> ACK sip:93147920634@192.168.130.68 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKUl6kDxievzWqzxmKlds9Ufw5fcCBwPpO;rport From: "Antares Soporte";tag=1fzemhoA46PBe3wN06B1 To: ;tag=as1996e832 Call-ID: 82a9aeff-279c-2414-673c-0882dfcd2600 CSeq: 61650 ACK Content-Length: 0 Route: Max-Forwards: 70 <-------------> antarespbx-camus*CLI> --- (9 headers 0 lines) --- antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> INVITE sip:93147920634@192.168.130.68 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKedLmWmC0B0wLtVTIu5c0svaCA1RBFKzA;rport From: "Antares Soporte";tag=1fzemhoA46PBe3wN06B1 To: Contact: "Antares Soporte";+g.oma.sip-im;+sip.ice;language="es,fr" Call-ID: 82a9aeff-279c-2414-673c-0882dfcd2600 CSeq: 61651 INVITE Content-Type: application/sdp Content-Length: 2766 Route: Max-Forwards: 70 Authorization: Digest username="er",realm="192.168.130.68",nonce="519eed2c",uri="sip:93147920634@192.168.130.68",response="8a7526d73a263d9447d6d2504850fc15",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: ER Technology LTDA v=0 o=- 5725125286898753000 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS O7epInWmLV3OdAKa3KcZILmnNrFdPH4pnkf0 m=audio 35686 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 181.62.249.112 b=AS:64 a=rtcp:35686 IN IP4 181.62.249.112 a=candidate:252621849 1 udp 2122260223 192.168.130.102 35686 typ host generation 0 a=candidate:252621849 2 udp 2122260223 192.168.130.102 35686 typ host generation 0 a=candidate:4062413514 1 udp 2122194687 192.168.122.1 36454 typ host generation 0 a=candidate:4062413514 2 udp 2122194687 192.168.122.1 36454 typ host generation 0 a=candidate:498801655 1 udp 2122129151 192.168.130.101 42792 typ host generation 0 a=candidate:498801655 2 udp 2122129151 192.168.130.101 42792 typ host generation 0 a=candidate:1724185560 1 udp 1686052607 181.62.249.112 35686 typ srflx raddr 192.168.130.102 rport 35686 generation 0 a=candidate:1724185560 2 udp 1686052607 181.62.249.112 35686 typ srflx raddr 192.168.130.102 rport 35686 generation 0 a=candidate:1953584182 1 udp 1685921535 181.62.249.112 42792 typ srflx raddr 192.168.130.101 rport 42792 generation 0 a=candidate:1953584182 2 udp 1685921535 181.62.249.112 42792 typ srflx raddr 192.168.130.101 rport 42792 generation 0 a=candidate:1099778281 1 tcp 1518280447 192.168.130.102 0 typ host tcptype active generation 0 a=candidate:1099778281 2 tcp 1518280447 192.168.130.102 0 typ host tcptype active generation 0 a=candidate:3164634682 1 tcp 1518214911 192.168.122.1 0 typ host tcptype active generation 0 a=candidate:3164634682 2 tcp 1518214911 192.168.122.1 0 typ host tcptype active generation 0 a=candidate:1396238087 1 tcp 1518149375 192.168.130.101 0 typ host tcptype active generation 0 a=candidate:1396238087 2 tcp 1518149375 192.168.130.101 0 typ host tcptype active generation 0 a=ice-ufrag:nEyEH5BsjuT1rizJ a=ice-pwd:aZp+RM4Kv9VnHzcRJlEaEu+J a=ice-options:google-ice a=fingerprint:sha-256 B4:3C:60:B0:60:86:89:8D:0B:A9:49:08:E8:F9:98:D3:7F:E5:9F:FD:5B:A6:C1:BE:B9:3A:B9:1D:C3:4F:32:91 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:528907053 cname:6UKlGaci38ADo7cn a=ssrc:528907053 msid:O7epInWmLV3OdAKa3KcZILmnNrFdPH4pnkf0 c17b9cc6-063e-481b-9934-a13f74b89a93 a=ssrc:528907053 mslabel:O7epInWmLV3OdAKa3KcZILmnNrFdPH4pnkf0 a=ssrc:528907053 label:c17b9cc6-063e-481b-9934-a13f74b89a93 <-------------> antarespbx-camus*CLI> --- (14 headers 51 lines) --- antarespbx-camus*CLI> Using INVITE request as basis request - 82a9aeff-279c-2414-673c-0882dfcd2600 antarespbx-camus*CLI> Found peer 'er' for 'er' from 192.168.130.102:57026 antarespbx-camus*CLI>  == Using SIP RTP CoS mark 5 antarespbx-camus*CLI>  == Using UDPTL CoS mark 5 antarespbx-camus*CLI> Found RTP audio format 111 antarespbx-camus*CLI> Found RTP audio format 103 antarespbx-camus*CLI> Found RTP audio format 104 antarespbx-camus*CLI> Found RTP audio format 0 antarespbx-camus*CLI> Found RTP audio format 8 antarespbx-camus*CLI> Found RTP audio format 106 antarespbx-camus*CLI> Found RTP audio format 105 antarespbx-camus*CLI> Found RTP audio format 13 antarespbx-camus*CLI> Found RTP audio format 126 antarespbx-camus*CLI> Found audio description format opus for ID 111 antarespbx-camus*CLI> Found unknown media description format ISAC for ID 103 antarespbx-camus*CLI> Found unknown media description format ISAC for ID 104 antarespbx-camus*CLI> Found audio description format PCMU for ID 0 antarespbx-camus*CLI> Found audio description format PCMA for ID 8 antarespbx-camus*CLI> Found unknown media description format CN for ID 106 antarespbx-camus*CLI> Found unknown media description format CN for ID 105 antarespbx-camus*CLI> Found audio description format CN for ID 13 antarespbx-camus*CLI> Found audio description format telephone-event for ID 126 antarespbx-camus*CLI> Capabilities: us - (ulaw|alaw|gsm), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) antarespbx-camus*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|) antarespbx-camus*CLI> Peer audio RTP is at port 181.62.249.112:35686 antarespbx-camus*CLI> Looking for 93147920634 in internal (domain 192.168.130.68) antarespbx-camus*CLI> sip_route_dump: route/path hop: antarespbx-camus*CLI>  <--- Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 100 Trying Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKedLmWmC0B0wLtVTIu5c0svaCA1RBFKzA;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=1fzemhoA46PBe3wN06B1 To: Call-ID: 82a9aeff-279c-2414-673c-0882dfcd2600 CSeq: 61651 INVITE Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> antarespbx-camus*CLI>  -- Executing [93147920634@internal:1] AGI("SIP/er-00000016", "agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=") in new stack antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: u'er is not locked' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: 'Verify in Blacklist in rule 93147920634' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: 'CONTEXTO : outgoing' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: '[ agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: 'Permission antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: 'recordCall' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: u'USER: antarespbx-camus*CLI>  -- AGI Script Executing Application: (MixMonitor) Options: (/records/ertech/admin/1421449212.44_er_93147920634_20150116_180012__ADMIN_OUTBOUND_.wav,baW(2)) antarespbx-camus*CLI>  -- AGI Script Executing Application: (Set) Options: (AUDIOHOOK_INHERIT(MixMonitor)=yes) antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: None antarespbx-camus*CLI>  == Begin MixMonitor Recording SIP/er-00000016 antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: 'Call antarespbx-camus*CLI>  -- AGI Script Executing Application: (Dial) Options: (SIP/2062589054/573147920634,60,iCHgRrwWTtF(hangup-context^caller^1)) antarespbx-camus*CLI>  == Using SIP RTP CoS mark 5 We think we can do text Audio is at 12404 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding codec g723 to SDP Adding codec g726 to SDP Adding codec g726aal2 to SDP Adding codec adpcm to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec lpc10 to SDP Adding codec g729 to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec ilbc to SDP Adding codec g722 to SDP Adding codec siren7 to SDP Adding codec siren14 to SDP Adding codec testlaw to SDP Adding codec g719 to SDP Adding codec opus to SDP Adding codec none to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 50.116.31.15:5060: INVITE sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK1116636c Max-Forwards: 70 From: ;tag=as178fbcb1 To: Contact: Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:00:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "8912490" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 898 v=0 o=root 46856499 46856499 IN IP4 192.168.130.68 s=Asterisk PBX 13.0.0 c=IN IP4 192.168.130.68 t=0 0 m=audio 12404 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- antarespbx-camus*CLI>  -- Called SIP/2062589054/573147920634 antarespbx-camus*CLI>  <--- Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 180 Ringing Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKedLmWmC0B0wLtVTIu5c0svaCA1RBFKzA;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=1fzemhoA46PBe3wN06B1 To: ;tag=as76999731 Call-ID: 82a9aeff-279c-2414-673c-0882dfcd2600 CSeq: 61651 INVITE Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> antarespbx-camus*CLI> Retransmitting #1 (no NAT) to 50.116.31.15:5060: INVITE sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK1116636c Max-Forwards: 70 From: ;tag=as178fbcb1 To: Contact: Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:00:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "8912490" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 898 v=0 o=root 46856499 46856499 IN IP4 192.168.130.68 s=Asterisk PBX 13.0.0 c=IN IP4 192.168.130.68 t=0 0 m=audio 12404 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK1116636c;received=181.62.249.112;rport=2048 From: ;tag=as178fbcb1 To: ;tag=as0b390dbf Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 102 INVITE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a3cc6f5" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (no NAT) to 50.116.31.15:5060: ACK sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK1116636c Max-Forwards: 70 From: ;tag=as178fbcb1 To: ;tag=as0b390dbf Contact: Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 --- We think we can do text Audio is at 12404 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding codec g723 to SDP Adding codec g726 to SDP Adding codec g726aal2 to SDP Adding codec adpcm to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec lpc10 to SDP Adding codec g729 to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec ilbc to SDP Adding codec g722 to SDP Adding codec siren7 to SDP Adding codec siren14 to SDP Adding codec testlaw to SDP Adding codec g719 to SDP Adding codec opus to SDP Adding codec none to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 50.116.31.15:5060: INVITE sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK00abb683 Max-Forwards: 70 From: ;tag=as178fbcb1 To: Contact: Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="2062589054", realm="asterisk", algorithm=MD5, uri="sip:573147920634@sip1.voztovoice.org", nonce="1a3cc6f5", response="261bf1b02dcc3df9edd1b7787773cfa4" Date: Fri, 16 Jan 2015 23:00:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "8912490" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 898 v=0 o=root 46856499 46856500 IN IP4 192.168.130.68 s=Asterisk PBX 13.0.0 c=IN IP4 192.168.130.68 t=0 0 m=audio 12404 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK1116636c;received=181.62.249.112;rport=2048 From: ;tag=as178fbcb1 To: ;tag=as0b390dbf Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 102 INVITE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a3cc6f5" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (no NAT) to 50.116.31.15:5060: ACK sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK00abb683 Max-Forwards: 70 From: ;tag=as178fbcb1 To: Contact: Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 --- antarespbx-camus*CLI> Retransmitting #1 (no NAT) to 50.116.31.15:5060: INVITE sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK00abb683 Max-Forwards: 70 From: ;tag=as178fbcb1 To: Contact: Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="2062589054", realm="asterisk", algorithm=MD5, uri="sip:573147920634@sip1.voztovoice.org", nonce="1a3cc6f5", response="261bf1b02dcc3df9edd1b7787773cfa4" Date: Fri, 16 Jan 2015 23:00:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "8912490" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 898 v=0 o=root 46856499 46856500 IN IP4 192.168.130.68 s=Asterisk PBX 13.0.0 c=IN IP4 192.168.130.68 t=0 0 m=audio 12404 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK00abb683;received=181.62.249.112;rport=2048 From: ;tag=as178fbcb1 To: Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 103 INVITE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> --- (12 headers 0 lines) --- antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK00abb683;received=181.62.249.112;rport=2048 From: ;tag=as178fbcb1 To: Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 103 INVITE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (12 headers 0 lines) --- antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK00abb683;received=181.62.249.112;rport=2048 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 103 INVITE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 299 v=0 o=a2billing 678208857 678208857 IN IP4 50.116.31.15 s=A2Billing c=IN IP4 50.116.31.15 t=0 0 m=audio 17956 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (11 headers 14 lines) --- sip_route_dump: no route/path Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 failed to extend from 64 to 98 Capabilities: us - (ulaw|alaw|gsm|g723|g726|g726aal2|adpcm|slin|slin|slin|slin|), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 50.116.31.15:17956 antarespbx-camus*CLI>  -- SIP/2062589054-00000017 is making progress passing it to SIP/er-00000016 antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> OPTIONS sip:s@192.168.130.68:5060 SIP/2.0 Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK3846c583;rport Max-Forwards: 70 From: "asterisk" ;tag=as693d4ab7 To: Contact: Call-ID: 4b7d84935ffafe48587899424dca1c81@50.116.31.15:5060 CSeq: 102 OPTIONS User-Agent: VozOrgSIP1 Date: Fri, 16 Jan 2015 23:00:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 50.116.31.15:5060 (no NAT) Looking for s in incoming (domain 192.168.130.68) <--- Transmitting (no NAT) to 50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK3846c583;received=50.116.31.15;rport=5060 From: "asterisk" ;tag=as693d4ab7 To: ;tag=as75913613 Call-ID: 4b7d84935ffafe48587899424dca1c81@50.116.31.15:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '4b7d84935ffafe48587899424dca1c81@50.116.31.15:5060' in 32000 ms (Method: OPTIONS) antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK00abb683;received=181.62.249.112;rport=2048 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 103 INVITE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 299 v=0 o=a2billing 678208857 678208857 IN IP4 50.116.31.15 s=A2Billing c=IN IP4 50.116.31.15 t=0 0 m=audio 17956 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (11 headers 14 lines) --- sip_route_dump: no route/path [Jan 16 18:00:21] WARNING[3823][C-0000000b]: chan_sip.c:16033 __set_address_from_contact: Invalid contact uri (missing sip: or sips:), attempting to use anyway [Jan 16 18:00:21] WARNING[3823][C-0000000b]: chan_sip.c:16046 __set_address_from_contact: Invalid URI: parse_uri failed to acquire hostport Transmitting (no NAT) to 50.116.31.15:5060: ACK sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK4489d2d0 Max-Forwards: 70 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Contact: Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 --- Reliably Transmitting (no NAT) to 50.116.31.15:5060: BYE sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK6a366d42 Max-Forwards: 70 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="2062589054", realm="asterisk", algorithm=MD5, uri="sip:573147920634@sip1.voztovoice.org", nonce="1a3cc6f5", response="de37cc5062247eb856633311ce66a6fb" X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- Scheduling destruction of SIP dialog '5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060' in 6400 ms (Method: INVITE) antarespbx-camus*CLI>  -- SIP/2062589054-00000017 answered SIP/er-00000016 Audio is at 14546 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKedLmWmC0B0wLtVTIu5c0svaCA1RBFKzA;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=1fzemhoA46PBe3wN06B1 To: ;tag=as76999731 Call-ID: 82a9aeff-279c-2414-673c-0882dfcd2600 CSeq: 61651 INVITE Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 893 v=0 o=root 667188154 667188154 IN IP4 192.168.130.68 s=Asterisk PBX 13.0.0 c=IN IP4 192.168.130.68 t=0 0 m=audio 14546 RTP/SAVPF 0 8 3 126 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-16 a=ptime:20 a=maxptime:150 a=ice-ufrag:584aa5eb6b05b25a26484f4c75b7fabe a=ice-pwd:6301398d7d5edc2061ddc26b7f334d44 a=candidate:Hc0a88244 1 UDP 2130706431 192.168.130.68 14546 typ host a=candidate:Sb53ef970 1 UDP 1694498815 181.62.249.112 14546 typ srflx raddr 192.168.130.68 rport 14546 a=candidate:Hc0a88244 2 UDP 2130706430 192.168.130.68 14547 typ host a=candidate:Sb53ef970 2 UDP 1694498814 181.62.249.112 14547 typ srflx raddr 192.168.130.68 rport 14547 a=connection:new a=setup:active a=fingerprint:SHA-256 12:1E:9F:B5:CB:6A:4A:BF:EB:2A:81:96:B5:EA:F4:80:FF:7D:27:65:75:BD:DB:86:98:AD:5E:7E:2F:16:3F:2C a=sendrecv <------------> -- Channel SIP/er-00000016 joined 'simple_bridge' basic-bridge <5b75f698-9c8a-4950-ba7a-821158c1c882> antarespbx-camus*CLI>  -- Channel SIP/2062589054-00000017 joined 'simple_bridge' basic-bridge <5b75f698-9c8a-4950-ba7a-821158c1c882> antarespbx-camus*CLI>  -- Channel SIP/2062589054-00000017 left 'simple_bridge' basic-bridge <5b75f698-9c8a-4950-ba7a-821158c1c882> antarespbx-camus*CLI>  -- Channel SIP/er-00000016 left 'simple_bridge' basic-bridge <5b75f698-9c8a-4950-ba7a-821158c1c882> antarespbx-camus*CLI>  -- AGI Script Executing Application: (Goto) Options: (hangup-context,called,1) -- Goto (hangup-context,called,1) antarespbx-camus*CLI> Scheduling destruction of SIP dialog '5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060' in 6400 ms (Method: INVITE) antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: 'DIALSTATUS ***ANSWER' antarespbx-camus*CLI> Reliably Transmitting (no NAT) to 50.116.31.15:5060: BYE sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK59e0a273 Max-Forwards: 70 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 105 BYE User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="2062589054", realm="asterisk", algorithm=MD5, uri="sip:573147920634@sip1.voztovoice.org", nonce="1a3cc6f5", response="de37cc5062247eb856633311ce66a6fb" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- antarespbx-camus*CLI>  -- AGI Script agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid= completed, returning 0 antarespbx-camus*CLI>  -- Executing [called@hangup-context:1] AGI("SIP/er-00000016", "agi://127.0.0.1:4573/?app=posthangup&recordid=None&duration=8") in new stack antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> ACK sip:93147920634@192.168.130.68:5060;transport=WS SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKDR6n7LIRT5prq50aM3ng;rport From: "Antares Soporte";tag=1fzemhoA46PBe3wN06B1 To: ;tag=as76999731 Contact: "Antares Soporte";+g.oma.sip-im;+sip.ice;language="es,fr" Call-ID: 82a9aeff-279c-2414-673c-0882dfcd2600 CSeq: 61651 ACK Content-Length: 0 Route: Max-Forwards: 70 Authorization: Digest username="er",realm="192.168.130.68",nonce="519eed2c",uri="sip:93147920634@192.168.130.68:5060;transport=WS",response="97bf4c489ad05bf1d7b0e90b301fdce4",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: ER Technology LTDA <-------------> --- (13 headers 0 lines) --- antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=posthangup&recordid=None&duration=8: u'UNIQUEID:1421449212.44' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=posthangup&recordid=None&duration=8: u'callerid:8912490' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=posthangup&recordid=None&duration=8: u'calleridname:unknown' antarespbx-camus*CLI>  -- AGI Script Executing Application: (Hangup) Options: () == Spawn extension (hangup-context, called, 1) exited non-zero on 'SIP/er-00000016' Scheduling destruction of SIP dialog '82a9aeff-279c-2414-673c-0882dfcd2600' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 192.168.130.102:57026: BYE sip:er@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.130.68:5060;branch=z9hG4bK5cc8b398;rport Max-Forwards: 70 From: ;tag=as76999731 To: "Antares Soporte";tag=1fzemhoA46PBe3wN06B1 Call-ID: 82a9aeff-279c-2414-673c-0882dfcd2600 CSeq: 102 BYE User-Agent: Asterisk PBX 13.0.0 Proxy-Authorization: Digest username="er", realm="192.168.130.68", algorithm=MD5, uri="sip:192.168.130.68", nonce="519eed2c", response="869951176796f7e9ce14e34112a19c5b" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- antarespbx-camus*CLI>  == MixMonitor close filestream (mixed) == End MixMonitor Recording SIP/er-00000016 antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.130.68:5060;rport=5060;branch=z9hG4bK5cc8b398 From: ;tag=as76999731 To: "Antares Soporte";tag=1fzemhoA46PBe3wN06B1 Contact: Call-ID: 82a9aeff-279c-2414-673c-0882dfcd2600 CSeq: 102 BYE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK00abb683;received=181.62.249.112;rport=2048 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 103 INVITE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 299 v=0 o=a2billing 678208857 678208857 IN IP4 50.116.31.15 s=A2Billing c=IN IP4 50.116.31.15 t=0 0 m=audio 17956 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (11 headers 14 lines) --- Transmitting (no NAT) to 50.116.31.15:5060: ACK sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK57329681 Max-Forwards: 70 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Contact: Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 --- Really destroying SIP dialog '82a9aeff-279c-2414-673c-0882dfcd2600' Method: INVITE antarespbx-camus*CLI> Retransmitting #1 (no NAT) to 50.116.31.15:5060: BYE sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK6a366d42 Max-Forwards: 70 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="2062589054", realm="asterisk", algorithm=MD5, uri="sip:573147920634@sip1.voztovoice.org", nonce="1a3cc6f5", response="de37cc5062247eb856633311ce66a6fb" X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK6a366d42;received=181.62.249.112;rport=2048 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 104 BYE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- antarespbx-camus*CLI> Retransmitting #1 (no NAT) to 50.116.31.15:5060: BYE sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK59e0a273 Max-Forwards: 70 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 105 BYE User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="2062589054", realm="asterisk", algorithm=MD5, uri="sip:573147920634@sip1.voztovoice.org", nonce="1a3cc6f5", response="de37cc5062247eb856633311ce66a6fb" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK59e0a273;received=181.62.249.112;rport=2048 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 105 BYE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060' Method: INVITE antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 500 Server error Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK6a366d42;received=181.62.249.112;rport=2048 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 104 BYE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK59e0a273;received=181.62.249.112;rport=2048 From: ;tag=as178fbcb1 To: ;tag=as3e50bd53 Call-ID: 5039fa615a953a6c56fe382b569d84bd@192.168.130.68:5060 CSeq: 105 BYE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- antarespbx-camus*CLI>  <--- SIP read from UDP:192.168.130.104:64251 ---> REGISTER sip:192.168.130.68;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.130.104:64251;branch=z9hG4bK-524287-1---958ea88baa2eec0b;rport Max-Forwards: 70 Contact: To: From: ;tag=567d0976 Call-ID: 9eJv0MT8qnzvkM1KyJ9EIA.. CSeq: 199 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Zoiper r28827 Authorization: Digest username="oeramos",realm="192.168.130.68",nonce="71ecd28f",uri="sip:192.168.130.68;transport=UDP",response="1a883dcaff43859b6ca90bd2871ba1ed",algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to 192.168.130.104:64251 (no NAT) Sending to 192.168.130.104:64251 (no NAT) <--- Transmitting (no NAT) to 192.168.130.104:64251 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.130.104:64251;branch=z9hG4bK-524287-1---958ea88baa2eec0b;received=192.168.130.104;rport=64251 From: ;tag=567d0976 To: ;tag=as5cf46615 Call-ID: 9eJv0MT8qnzvkM1KyJ9EIA.. CSeq: 199 REGISTER Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="192.168.130.68", nonce="0d217ff6" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9eJv0MT8qnzvkM1KyJ9EIA..' in 32000 ms (Method: REGISTER) antarespbx-camus*CLI>  <--- SIP read from UDP:192.168.130.104:64251 ---> REGISTER sip:192.168.130.68;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.130.104:64251;branch=z9hG4bK-524287-1---706e4c3f1015ac6e;rport Max-Forwards: 70 Contact: To: From: ;tag=567d0976 Call-ID: 9eJv0MT8qnzvkM1KyJ9EIA.. CSeq: 200 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Zoiper r28827 Authorization: Digest username="oeramos",realm="192.168.130.68",nonce="0d217ff6",uri="sip:192.168.130.68;transport=UDP",response="32a101d5ad80877a3a79175add7c5757",algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to 192.168.130.104:64251 (no NAT) antarespbx-camus*CLI> Reliably Transmitting (no NAT) to 192.168.130.104:64251: OPTIONS sip:oeramos@192.168.130.104:64251;rinstance=f31723138230a356;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK009138fc Max-Forwards: 70 From: "asterisk" ;tag=as7ba83861 To: Contact: Call-ID: 7d58f1427311db8078ad900876eb0d97@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:00:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- Transmitting (no NAT) to 192.168.130.104:64251 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.104:64251;branch=z9hG4bK-524287-1---706e4c3f1015ac6e;received=192.168.130.104;rport=64251 From: ;tag=567d0976 To: ;tag=as5cf46615 Call-ID: 9eJv0MT8qnzvkM1KyJ9EIA.. CSeq: 200 REGISTER Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Fri, 16 Jan 2015 23:00:24 GMT Content-Length: 0 <------------> antarespbx-camus*CLI> Scheduling destruction of SIP dialog '9eJv0MT8qnzvkM1KyJ9EIA..' in 32000 ms (Method: REGISTER) antarespbx-camus*CLI>  <--- SIP read from UDP:192.168.130.104:64251 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK009138fc Contact: To: ;tag=4b80c113 From: "asterisk" ;tag=as7ba83861 Call-ID: 7d58f1427311db8078ad900876eb0d97@192.168.130.68:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Zoiper r28827 Allow-Events: presence, kpml Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (14 headers 0 lines) --- antarespbx-camus*CLI> Really destroying SIP dialog '7d58f1427311db8078ad900876eb0d97@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Really destroying SIP dialog '0af397bd405311833d9bea5c65b21f05@50.116.31.15:5060' Method: OPTIONS antarespbx-camus*CLI> Really destroying SIP dialog 'df282f12-a3ea-dc9d-8a07-08673dce3527' Method: REGISTER antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> OPTIONS sip:s@192.168.130.68:5060 SIP/2.0 Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK719251bb;rport Max-Forwards: 70 From: "asterisk" ;tag=as5b12fcdd To: Contact: Call-ID: 421acde24c04552d0b2d8edb314df482@50.116.31.15:5060 CSeq: 102 OPTIONS User-Agent: VozOrgSIP1 Date: Fri, 16 Jan 2015 23:00:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 50.116.31.15:5060 (no NAT) Looking for s in incoming (domain 192.168.130.68) <--- Transmitting (no NAT) to 50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK719251bb;received=50.116.31.15;rport=5060 From: "asterisk" ;tag=as5b12fcdd To: ;tag=as7151fb98 Call-ID: 421acde24c04552d0b2d8edb314df482@50.116.31.15:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '421acde24c04552d0b2d8edb314df482@50.116.31.15:5060' in 32000 ms (Method: OPTIONS) antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> INVITE sip:93147920634@192.168.130.68 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKNplIk0DlzUrGcFUbIXc17FpM0TulxCaK;rport From: "Antares Soporte";tag=1BwXMeSgrReJyIGlHlss To: Contact: "Antares Soporte";+g.oma.sip-im;+sip.ice;language="es,fr" Call-ID: 62b4e37a-f281-97d6-3c63-d469f75901dd CSeq: 42834 INVITE Content-Type: application/sdp Content-Length: 2770 Route: Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: ER Technology LTDA v=0 o=- 2980551421737276000 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS hXucrIyDXa4qMtqFhzc0A64P871UwxWYtycJ m=audio 52078 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 181.62.249.112 b=AS:64 a=rtcp:52078 IN IP4 181.62.249.112 a=candidate:252621849 1 udp 2122260223 192.168.130.102 52078 typ host generation 0 a=candidate:252621849 2 udp 2122260223 192.168.130.102 52078 typ host generation 0 a=candidate:4062413514 1 udp 2122194687 192.168.122.1 52029 typ host generation 0 a=candidate:4062413514 2 udp 2122194687 192.168.122.1 52029 typ host generation 0 a=candidate:498801655 1 udp 2122129151 192.168.130.101 42937 typ host generation 0 a=candidate:498801655 2 udp 2122129151 192.168.130.101 42937 typ host generation 0 a=candidate:1724185560 1 udp 1686052607 181.62.249.112 52078 typ srflx raddr 192.168.130.102 rport 52078 generation 0 a=candidate:1724185560 2 udp 1686052607 181.62.249.112 52078 typ srflx raddr 192.168.130.102 rport 52078 generation 0 a=candidate:1953584182 1 udp 1685921535 181.62.249.112 42937 typ srflx raddr 192.168.130.101 rport 42937 generation 0 a=candidate:1953584182 2 udp 1685921535 181.62.249.112 42937 typ srflx raddr 192.168.130.101 rport 42937 generation 0 a=candidate:1099778281 1 tcp 1518280447 192.168.130.102 0 typ host tcptype active generation 0 a=candidate:1099778281 2 tcp 1518280447 192.168.130.102 0 typ host tcptype active generation 0 a=candidate:3164634682 1 tcp 1518214911 192.168.122.1 0 typ host tcptype active generation 0 a=candidate:3164634682 2 tcp 1518214911 192.168.122.1 0 typ host tcptype active generation 0 a=candidate:1396238087 1 tcp 1518149375 192.168.130.101 0 typ host tcptype active generation 0 a=candidate:1396238087 2 tcp 1518149375 192.168.130.101 0 typ host tcptype active generation 0 a=ice-ufrag:vfnuNJJ+SSbcQ5TU a=ice-pwd:1E9W4yWFv2lPVNfHMWqiDxh3 a=ice-options:google-ice a=fingerprint:sha-256 B4:3C:60:B0:60:86:89:8D:0B:A9:49:08:E8:F9:98:D3:7F:E5:9F:FD:5B:A6:C1:BE:B9:3A:B9:1D:C3:4F:32:91 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2046228759 cname:M+ImMF9LEjeuV4Kb a=ssrc:2046228759 msid:hXucrIyDXa4qMtqFhzc0A64P871UwxWYtycJ b95001ed-ebf3-4914-be2a-6ebc006d068d a=ssrc:2046228759 mslabel:hXucrIyDXa4qMtqFhzc0A64P871UwxWYtycJ a=ssrc:2046228759 label:b95001ed-ebf3-4914-be2a-6ebc006d068d <-------------> antarespbx-camus*CLI> --- (13 headers 51 lines) --- antarespbx-camus*CLI> Using INVITE request as basis request - 62b4e37a-f281-97d6-3c63-d469f75901dd antarespbx-camus*CLI> Found peer 'er' for 'er' from 192.168.130.102:57026 antarespbx-camus*CLI>  <--- Reliably Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKNplIk0DlzUrGcFUbIXc17FpM0TulxCaK;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=1BwXMeSgrReJyIGlHlss To: ;tag=as02293b19 Call-ID: 62b4e37a-f281-97d6-3c63-d469f75901dd CSeq: 42834 INVITE Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="192.168.130.68", nonce="731840e7" Content-Length: 0 <------------> antarespbx-camus*CLI> Scheduling destruction of SIP dialog '62b4e37a-f281-97d6-3c63-d469f75901dd' in 6400 ms (Method: INVITE) antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> ACK sip:93147920634@192.168.130.68 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKNplIk0DlzUrGcFUbIXc17FpM0TulxCaK;rport From: "Antares Soporte";tag=1BwXMeSgrReJyIGlHlss To: ;tag=as02293b19 Call-ID: 62b4e37a-f281-97d6-3c63-d469f75901dd CSeq: 42834 ACK Content-Length: 0 Route: Max-Forwards: 70 <-------------> antarespbx-camus*CLI> --- (9 headers 0 lines) --- antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> INVITE sip:93147920634@192.168.130.68 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKByzOWzM3kM9bWfiSo06W3GhBBpXt7sIB;rport From: "Antares Soporte";tag=1BwXMeSgrReJyIGlHlss To: Contact: "Antares Soporte";+g.oma.sip-im;+sip.ice;language="es,fr" Call-ID: 62b4e37a-f281-97d6-3c63-d469f75901dd CSeq: 42835 INVITE Content-Type: application/sdp Content-Length: 2770 Route: Max-Forwards: 70 Authorization: Digest username="er",realm="192.168.130.68",nonce="731840e7",uri="sip:93147920634@192.168.130.68",response="7059cc629807f66fa2bf07c7c0925c33",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: ER Technology LTDA v=0 o=- 2980551421737276000 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS hXucrIyDXa4qMtqFhzc0A64P871UwxWYtycJ m=audio 52078 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 181.62.249.112 b=AS:64 a=rtcp:52078 IN IP4 181.62.249.112 a=candidate:252621849 1 udp 2122260223 192.168.130.102 52078 typ host generation 0 a=candidate:252621849 2 udp 2122260223 192.168.130.102 52078 typ host generation 0 a=candidate:4062413514 1 udp 2122194687 192.168.122.1 52029 typ host generation 0 a=candidate:4062413514 2 udp 2122194687 192.168.122.1 52029 typ host generation 0 a=candidate:498801655 1 udp 2122129151 192.168.130.101 42937 typ host generation 0 a=candidate:498801655 2 udp 2122129151 192.168.130.101 42937 typ host generation 0 a=candidate:1724185560 1 udp 1686052607 181.62.249.112 52078 typ srflx raddr 192.168.130.102 rport 52078 generation 0 a=candidate:1724185560 2 udp 1686052607 181.62.249.112 52078 typ srflx raddr 192.168.130.102 rport 52078 generation 0 a=candidate:1953584182 1 udp 1685921535 181.62.249.112 42937 typ srflx raddr 192.168.130.101 rport 42937 generation 0 a=candidate:1953584182 2 udp 1685921535 181.62.249.112 42937 typ srflx raddr 192.168.130.101 rport 42937 generation 0 a=candidate:1099778281 1 tcp 1518280447 192.168.130.102 0 typ host tcptype active generation 0 a=candidate:1099778281 2 tcp 1518280447 192.168.130.102 0 typ host tcptype active generation 0 a=candidate:3164634682 1 tcp 1518214911 192.168.122.1 0 typ host tcptype active generation 0 a=candidate:3164634682 2 tcp 1518214911 192.168.122.1 0 typ host tcptype active generation 0 a=candidate:1396238087 1 tcp 1518149375 192.168.130.101 0 typ host tcptype active generation 0 a=candidate:1396238087 2 tcp 1518149375 192.168.130.101 0 typ host tcptype active generation 0 a=ice-ufrag:vfnuNJJ+SSbcQ5TU a=ice-pwd:1E9W4yWFv2lPVNfHMWqiDxh3 a=ice-options:google-ice a=fingerprint:sha-256 B4:3C:60:B0:60:86:89:8D:0B:A9:49:08:E8:F9:98:D3:7F:E5:9F:FD:5B:A6:C1:BE:B9:3A:B9:1D:C3:4F:32:91 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2046228759 cname:M+ImMF9LEjeuV4Kb a=ssrc:2046228759 msid:hXucrIyDXa4qMtqFhzc0A64P871UwxWYtycJ b95001ed-ebf3-4914-be2a-6ebc006d068d a=ssrc:2046228759 mslabel:hXucrIyDXa4qMtqFhzc0A64P871UwxWYtycJ a=ssrc:2046228759 label:b95001ed-ebf3-4914-be2a-6ebc006d068d <-------------> antarespbx-camus*CLI> --- (14 headers 51 lines) --- antarespbx-camus*CLI> Using INVITE request as basis request - 62b4e37a-f281-97d6-3c63-d469f75901dd antarespbx-camus*CLI> Found peer 'er' for 'er' from 192.168.130.102:57026 antarespbx-camus*CLI>  == Using SIP RTP CoS mark 5 antarespbx-camus*CLI>  == Using UDPTL CoS mark 5 antarespbx-camus*CLI> Found RTP audio format 111 antarespbx-camus*CLI> Found RTP audio format 103 antarespbx-camus*CLI> Found RTP audio format 104 antarespbx-camus*CLI> Found RTP audio format 0 antarespbx-camus*CLI> Found RTP audio format 8 antarespbx-camus*CLI> Found RTP audio format 106 antarespbx-camus*CLI> Found RTP audio format 105 antarespbx-camus*CLI> Found RTP audio format 13 antarespbx-camus*CLI> Found RTP audio format 126 antarespbx-camus*CLI> Found audio description format opus for ID 111 antarespbx-camus*CLI> Found unknown media description format ISAC for ID 103 antarespbx-camus*CLI> Found unknown media description format ISAC for ID 104 antarespbx-camus*CLI> Found audio description format PCMU for ID 0 antarespbx-camus*CLI> Found audio description format PCMA for ID 8 antarespbx-camus*CLI> Found unknown media description format CN for ID 106 antarespbx-camus*CLI> Found unknown media description format CN for ID 105 antarespbx-camus*CLI> Found audio description format CN for ID 13 antarespbx-camus*CLI> Found audio description format telephone-event for ID 126 antarespbx-camus*CLI> Capabilities: us - (ulaw|alaw|gsm), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) antarespbx-camus*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|) antarespbx-camus*CLI> Peer audio RTP is at port 181.62.249.112:52078 antarespbx-camus*CLI> Looking for 93147920634 in internal (domain 192.168.130.68) antarespbx-camus*CLI> sip_route_dump: route/path hop: antarespbx-camus*CLI>  <--- Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 100 Trying Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKByzOWzM3kM9bWfiSo06W3GhBBpXt7sIB;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=1BwXMeSgrReJyIGlHlss To: Call-ID: 62b4e37a-f281-97d6-3c63-d469f75901dd CSeq: 42835 INVITE Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> antarespbx-camus*CLI>  -- Executing [93147920634@internal:1] AGI("SIP/er-00000018", "agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=") in new stack antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: u'er is not locked' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: 'Verify in Blacklist in rule 93147920634' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: 'CONTEXTO : outgoing' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: '[ agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: 'Permission antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: 'recordCall' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: u'USER: antarespbx-camus*CLI>  -- AGI Script Executing Application: (MixMonitor) Options: (/records/ertech/admin/1421449247.48_er_93147920634_20150116_180047__ADMIN_OUTBOUND_.wav,baW(2)) antarespbx-camus*CLI>  -- AGI Script Executing Application: (Set) Options: (AUDIOHOOK_INHERIT(MixMonitor)=yes) antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: None antarespbx-camus*CLI>  == Begin MixMonitor Recording SIP/er-00000018 antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid=: 'Call antarespbx-camus*CLI>  -- AGI Script Executing Application: (Dial) Options: (SIP/2062589054/573147920634,60,iCHgRrwWTtF(hangup-context^caller^1)) antarespbx-camus*CLI>  == Using SIP RTP CoS mark 5 We think we can do text Audio is at 14540 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding codec g723 to SDP Adding codec g726 to SDP Adding codec g726aal2 to SDP Adding codec adpcm to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec lpc10 to SDP Adding codec g729 to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec ilbc to SDP Adding codec g722 to SDP Adding codec siren7 to SDP Adding codec siren14 to SDP Adding codec testlaw to SDP Adding codec g719 to SDP Adding codec opus to SDP Adding codec none to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 50.116.31.15:5060: INVITE sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK461ebfc9 Max-Forwards: 70 From: ;tag=as3393cee8 To: Contact: Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:00:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "8912490" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 902 v=0 o=root 2145650748 2145650748 IN IP4 192.168.130.68 s=Asterisk PBX 13.0.0 c=IN IP4 192.168.130.68 t=0 0 m=audio 14540 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- antarespbx-camus*CLI>  -- Called SIP/2062589054/573147920634 antarespbx-camus*CLI>  <--- Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 180 Ringing Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKByzOWzM3kM9bWfiSo06W3GhBBpXt7sIB;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=1BwXMeSgrReJyIGlHlss To: ;tag=as2806ac81 Call-ID: 62b4e37a-f281-97d6-3c63-d469f75901dd CSeq: 42835 INVITE Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> antarespbx-camus*CLI> Retransmitting #1 (no NAT) to 50.116.31.15:5060: INVITE sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK461ebfc9 Max-Forwards: 70 From: ;tag=as3393cee8 To: Contact: Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:00:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "8912490" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 902 v=0 o=root 2145650748 2145650748 IN IP4 192.168.130.68 s=Asterisk PBX 13.0.0 c=IN IP4 192.168.130.68 t=0 0 m=audio 14540 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK461ebfc9;received=181.62.249.112;rport=2048 From: ;tag=as3393cee8 To: ;tag=as3c6e8420 Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 102 INVITE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d41b320" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (no NAT) to 50.116.31.15:5060: ACK sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK461ebfc9 Max-Forwards: 70 From: ;tag=as3393cee8 To: ;tag=as3c6e8420 Contact: Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 --- We think we can do text Audio is at 14540 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding codec g723 to SDP Adding codec g726 to SDP Adding codec g726aal2 to SDP Adding codec adpcm to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec slin to SDP Adding codec lpc10 to SDP Adding codec g729 to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec speex to SDP Adding codec ilbc to SDP Adding codec g722 to SDP Adding codec siren7 to SDP Adding codec siren14 to SDP Adding codec testlaw to SDP Adding codec g719 to SDP Adding codec opus to SDP Adding codec none to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 50.116.31.15:5060: INVITE sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK34039290 Max-Forwards: 70 From: ;tag=as3393cee8 To: Contact: Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="2062589054", realm="asterisk", algorithm=MD5, uri="sip:573147920634@sip1.voztovoice.org", nonce="3d41b320", response="755f0df86050a84f9f6c33b1393fbd21" Date: Fri, 16 Jan 2015 23:00:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "8912490" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 902 v=0 o=root 2145650748 2145650749 IN IP4 192.168.130.68 s=Asterisk PBX 13.0.0 c=IN IP4 192.168.130.68 t=0 0 m=audio 14540 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- antarespbx-camus*CLI> Retransmitting #1 (no NAT) to 50.116.31.15:5060: INVITE sip:573147920634@sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK34039290 Max-Forwards: 70 From: ;tag=as3393cee8 To: Contact: Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="2062589054", realm="asterisk", algorithm=MD5, uri="sip:573147920634@sip1.voztovoice.org", nonce="3d41b320", response="755f0df86050a84f9f6c33b1393fbd21" Date: Fri, 16 Jan 2015 23:00:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "8912490" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 902 v=0 o=root 2145650748 2145650749 IN IP4 192.168.130.68 s=Asterisk PBX 13.0.0 c=IN IP4 192.168.130.68 t=0 0 m=audio 14540 RTP/AVP 0 8 3 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK34039290;received=181.62.249.112;rport=2048 From: ;tag=as3393cee8 To: Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 103 INVITE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> --- (12 headers 0 lines) --- antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK34039290;received=181.62.249.112;rport=2048 From: ;tag=as3393cee8 To: Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 103 INVITE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (12 headers 0 lines) --- antarespbx-camus*CLI> Really destroying SIP dialog '4b7d84935ffafe48587899424dca1c81@50.116.31.15:5060' Method: OPTIONS antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK34039290;received=181.62.249.112;rport=2048 From: ;tag=as3393cee8 To: ;tag=as6b55c4d7 Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 103 INVITE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 299 v=0 o=a2billing 881057265 881057265 IN IP4 50.116.31.15 s=A2Billing c=IN IP4 50.116.31.15 t=0 0 m=audio 17762 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (14 headers 14 lines) --- sip_route_dump: route/path hop: Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 failed to extend from 64 to 98 Capabilities: us - (ulaw|alaw|gsm|g723|g726|g726aal2|adpcm|slin|slin|slin|slin|), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 50.116.31.15:17762 antarespbx-camus*CLI>  -- SIP/2062589054-00000019 is making progress passing it to SIP/er-00000018 antarespbx-camus*CLI>  Really destroying SIP dialog '9eJv0MT8qnzvkM1KyJ9EIA..' Method: REGISTER antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK34039290;received=181.62.249.112;rport=2048 From: ;tag=as3393cee8 To: ;tag=as6b55c4d7 Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 103 INVITE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 299 v=0 o=a2billing 881057265 881057265 IN IP4 50.116.31.15 s=A2Billing c=IN IP4 50.116.31.15 t=0 0 m=audio 17762 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (14 headers 14 lines) --- sip_route_dump: route/path hop: set_destination: Parsing for address/port to send to set_destination: set destination to 50.116.31.15:5060 Transmitting (no NAT) to 50.116.31.15:5060: ACK sip:573147920634@50.116.31.15:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK5967e050 Max-Forwards: 70 From: ;tag=as3393cee8 To: ;tag=as6b55c4d7 Contact: Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 --- antarespbx-camus*CLI>  -- SIP/2062589054-00000019 answered SIP/er-00000018 Audio is at 12840 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKByzOWzM3kM9bWfiSo06W3GhBBpXt7sIB;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=1BwXMeSgrReJyIGlHlss To: ;tag=as2806ac81 Call-ID: 62b4e37a-f281-97d6-3c63-d469f75901dd CSeq: 42835 INVITE Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 895 v=0 o=root 1491108327 1491108327 IN IP4 192.168.130.68 s=Asterisk PBX 13.0.0 c=IN IP4 192.168.130.68 t=0 0 m=audio 12840 RTP/SAVPF 0 8 3 126 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-16 a=ptime:20 a=maxptime:150 a=ice-ufrag:5ec6040a5c0bc93575577ba45a968d32 a=ice-pwd:356edbe21213351f6ac8083d11089a8c a=candidate:Hc0a88244 1 UDP 2130706431 192.168.130.68 12840 typ host a=candidate:Sb53ef970 1 UDP 1694498815 181.62.249.112 12840 typ srflx raddr 192.168.130.68 rport 12840 a=candidate:Hc0a88244 2 UDP 2130706430 192.168.130.68 12841 typ host a=candidate:Sb53ef970 2 UDP 1694498814 181.62.249.112 12841 typ srflx raddr 192.168.130.68 rport 12841 a=connection:new a=setup:active a=fingerprint:SHA-256 12:1E:9F:B5:CB:6A:4A:BF:EB:2A:81:96:B5:EA:F4:80:FF:7D:27:65:75:BD:DB:86:98:AD:5E:7E:2F:16:3F:2C a=sendrecv <------------> -- Channel SIP/er-00000018 joined 'simple_bridge' basic-bridge antarespbx-camus*CLI>  -- Channel SIP/2062589054-00000019 joined 'simple_bridge' basic-bridge antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> ACK sip:93147920634@192.168.130.68:5060;transport=WS SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKiy4HzAW2rpZqNgOn1tD8;rport From: "Antares Soporte";tag=1BwXMeSgrReJyIGlHlss To: ;tag=as2806ac81 Contact: "Antares Soporte";+g.oma.sip-im;+sip.ice;language="es,fr" Call-ID: 62b4e37a-f281-97d6-3c63-d469f75901dd CSeq: 42835 ACK Content-Length: 0 Route: Max-Forwards: 70 Authorization: Digest username="er",realm="192.168.130.68",nonce="731840e7",uri="sip:93147920634@192.168.130.68:5060;transport=WS",response="726bb4f8b50df30ba63794ef129a5b83",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: ER Technology LTDA <-------------> --- (13 headers 0 lines) --- antarespbx-camus*CLI> Reliably Transmitting (NAT) to 192.168.130.102:57026: OPTIONS sip:er@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.130.68:5060;branch=z9hG4bK29ea690b;rport Max-Forwards: 70 From: "asterisk" ;tag=as39aa44b7 To: Contact: Call-ID: 585b5f6a5af5416c76b6c6690cf3ec29@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:00:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> SIP/2.0 405 Method Not Allowed Via: SIP/2.0/WS 192.168.130.68:5060;rport=5060;branch=z9hG4bK29ea690b From: "asterisk";tag=as39aa44b7 To: Call-ID: 585b5f6a5af5416c76b6c6690cf3ec29@192.168.130.68:5060 CSeq: 102 OPTIONS Content-Length: 0 <-------------> --- (7 headers 0 lines) --- antarespbx-camus*CLI>  <--- SIP read from UDP:192.168.130.104:64251 ---> <-------------> Really destroying SIP dialog '585b5f6a5af5416c76b6c6690cf3ec29@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Reliably Transmitting (no NAT) to 50.116.31.15:5060: OPTIONS sip:sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK1d652ab5 Max-Forwards: 70 From: "asterisk" ;tag=as6bce1eae To: Contact: Call-ID: 6d65913a6bc980c9716f907d1cdc6238@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:01:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI> Reliably Transmitting (no NAT) to 208.66.193.46:5060: OPTIONS sip:sip.ipsofactum.com SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK4f08ddfd Max-Forwards: 70 From: "asterisk" ;tag=as57b02059 To: Contact: Call-ID: 0ef59f2e230128c03bba0e6a74089bd8@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:01:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:208.66.193.46:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK4f08ddfd;received=181.62.249.112;rport=2048 From: "asterisk" ;tag=as57b02059 To: ;tag=as2d8ca23f Call-ID: 0ef59f2e230128c03bba0e6a74089bd8@192.168.130.68:5060 CSeq: 102 OPTIONS Server: Ipsoswitch - VoIP Powered by IPSOFACTUM Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '0ef59f2e230128c03bba0e6a74089bd8@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK1d652ab5;received=181.62.249.112;rport=2048 From: "asterisk" ;tag=as6bce1eae To: ;tag=as7d82c342 Call-ID: 6d65913a6bc980c9716f907d1cdc6238@192.168.130.68:5060 CSeq: 102 OPTIONS Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '6d65913a6bc980c9716f907d1cdc6238@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Reliably Transmitting (NAT) to 173.192.95.227:5060: OPTIONS sip:173.192.95.227 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK309f98da;rport Max-Forwards: 70 From: "asterisk" ;tag=as1135c1b0 To: Contact: Call-ID: 0a846d8272a86cc30135385a3c895ed0@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:01:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:173.192.95.227:5060 ---> SIP/2.0 200 OK CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK309f98da;rport From: "asterisk" ;tag=as1135c1b0 Call-ID: 0a846d8272a86cc30135385a3c895ed0@192.168.130.68:5060 To: ;tag=160157151738 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (8 headers 0 lines) --- antarespbx-camus*CLI> Really destroying SIP dialog '0a846d8272a86cc30135385a3c895ed0@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Reliably Transmitting (NAT) to 173.255.199.156:5060: OPTIONS sip:173.255.199.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK5386bbed;rport Max-Forwards: 70 From: "asterisk" ;tag=as1e5ab708 To: Contact: Call-ID: 421864fc35eead860d88e7ba47eff619@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:01:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:173.255.199.156:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK5386bbed;received=181.62.249.112;rport=2048 From: "asterisk" ;tag=as1e5ab708 To: ;tag=as5f678902 Call-ID: 421864fc35eead860d88e7ba47eff619@192.168.130.68:5060 CSeq: 102 OPTIONS Server: VozOrgSIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (12 headers 0 lines) --- antarespbx-camus*CLI> Really destroying SIP dialog '421864fc35eead860d88e7ba47eff619@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Really destroying SIP dialog '421acde24c04552d0b2d8edb314df482@50.116.31.15:5060' Method: OPTIONS antarespbx-camus*CLI> [Jan 16 18:01:22] NOTICE[3823]: chan_sip.c:15178 sip_reregister: -- Re-registration for 15011501@sip.ipsofactum.com antarespbx-camus*CLI> REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 208.66.193.46:5060: REGISTER sip:sip.ipsofactum.com SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK4eb68e4a Max-Forwards: 70 From: ;tag=as0d434a34 To: Call-ID: 23d1b53013ad94930cd71e7057254c2d@127.0.0.2 CSeq: 122 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="15011501", realm="ipsofactum.com", algorithm=MD5, uri="sip:sip.ipsofactum.com", nonce="22b5b3fc", response="835bb1c9f5d0d2d301f23c2b0c161d97" Expires: 120 Contact: Content-Length: 0 --- [Jan 16 18:01:22] NOTICE[3823]: chan_sip.c:15178 sip_reregister: -- Re-registration for 2062589054@sip1.voztovoice.org antarespbx-camus*CLI> REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 50.116.31.15:5060: REGISTER sip:sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK699e685b Max-Forwards: 70 From: ;tag=as5f05c4b2 To: Call-ID: 0a8bf4c30530ae821813f278257eed34@127.0.0.2 CSeq: 122 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="2062589054", realm="asterisk", algorithm=MD5, uri="sip:sip1.voztovoice.org", nonce="60e89577", response="f465c4bcae151469597a8d976a07857e" Expires: 120 Contact: Content-Length: 0 --- <--- SIP read from UDP:208.66.193.46:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK4eb68e4a;received=181.62.249.112;rport=2048 From: ;tag=as0d434a34 To: ;tag=as3f02220c Call-ID: 23d1b53013ad94930cd71e7057254c2d@127.0.0.2 CSeq: 122 REGISTER Server: Ipsoswitch - VoIP Powered by IPSOFACTUM Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="ipsofactum.com", nonce="12fbbebb" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Responding to challenge, registration to domain/host name sip.ipsofactum.com antarespbx-camus*CLI> REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 208.66.193.46:5060: REGISTER sip:sip.ipsofactum.com SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK186a4c7a Max-Forwards: 70 From: ;tag=as0d434a34 To: Call-ID: 23d1b53013ad94930cd71e7057254c2d@127.0.0.2 CSeq: 123 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="15011501", realm="ipsofactum.com", algorithm=MD5, uri="sip:sip.ipsofactum.com", nonce="12fbbebb", response="ba1a2e1baaa2c8ce287ef63152fab36c" Expires: 120 Contact: Content-Length: 0 --- <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK699e685b;received=181.62.249.112;rport=2048 From: ;tag=as5f05c4b2 To: ;tag=as4f98d597 Call-ID: 0a8bf4c30530ae821813f278257eed34@127.0.0.2 CSeq: 122 REGISTER Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a60356f" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Responding to challenge, registration to domain/host name sip1.voztovoice.org antarespbx-camus*CLI> REGISTER 12 headers, 0 lines antarespbx-camus*CLI> Reliably Transmitting (no NAT) to 50.116.31.15:5060: REGISTER sip:sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK1ffd9ad5 Max-Forwards: 70 From: ;tag=as5f05c4b2 To: Call-ID: 0a8bf4c30530ae821813f278257eed34@127.0.0.2 CSeq: 123 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="2062589054", realm="asterisk", algorithm=MD5, uri="sip:sip1.voztovoice.org", nonce="0a60356f", response="4c5639a2f70808ffcdcd7932fa292fe0" Expires: 120 Contact: Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:208.66.193.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK186a4c7a;received=181.62.249.112;rport=2048 From: ;tag=as0d434a34 To: ;tag=as3f02220c Call-ID: 23d1b53013ad94930cd71e7057254c2d@127.0.0.2 CSeq: 123 REGISTER Server: Ipsoswitch - VoIP Powered by IPSOFACTUM Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 16 Jan 2015 23:01:21 GMT Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (13 headers 0 lines) --- antarespbx-camus*CLI> [Jan 16 18:01:22] NOTICE[3823]: chan_sip.c:23726 handle_response_register: Outbound Registration: Expiry for sip.ipsofactum.com is 120 sec (Scheduling reregistration in 105 s) antarespbx-camus*CLI> Really destroying SIP dialog '23d1b53013ad94930cd71e7057254c2d@127.0.0.2' Method: REGISTER antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> OPTIONS sip:s@192.168.130.68:5060 SIP/2.0 Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK54f82d7f;rport Max-Forwards: 70 From: "asterisk" ;tag=as2b32d977 To: Contact: Call-ID: 56280500777e6f8945c2cb5034e7f1eb@50.116.31.15:5060 CSeq: 102 OPTIONS User-Agent: VozOrgSIP1 Date: Fri, 16 Jan 2015 23:01:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (13 headers 0 lines) --- antarespbx-camus*CLI> Sending to 50.116.31.15:5060 (no NAT) antarespbx-camus*CLI> Looking for s in incoming (domain 192.168.130.68) antarespbx-camus*CLI>  <--- Transmitting (no NAT) to 50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK54f82d7f;received=50.116.31.15;rport=5060 From: "asterisk" ;tag=as2b32d977 To: ;tag=as211aa72f Call-ID: 56280500777e6f8945c2cb5034e7f1eb@50.116.31.15:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> antarespbx-camus*CLI> Scheduling destruction of SIP dialog '56280500777e6f8945c2cb5034e7f1eb@50.116.31.15:5060' in 32000 ms (Method: OPTIONS) antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK1ffd9ad5;received=181.62.249.112;rport=2048 From: ;tag=as5f05c4b2 To: ;tag=as4f98d597 Call-ID: 0a8bf4c30530ae821813f278257eed34@127.0.0.2 CSeq: 123 REGISTER Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Fri, 16 Jan 2015 23:01:22 GMT Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (13 headers 0 lines) --- antarespbx-camus*CLI> [Jan 16 18:01:22] NOTICE[3823]: chan_sip.c:23726 handle_response_register: Outbound Registration: Expiry for sip1.voztovoice.org is 120 sec (Scheduling reregistration in 105 s) antarespbx-camus*CLI> Really destroying SIP dialog '0a8bf4c30530ae821813f278257eed34@127.0.0.2' Method: REGISTER antarespbx-camus*CLI>  <--- SIP read from UDP:192.168.130.104:64251 ---> REGISTER sip:192.168.130.68;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.130.104:64251;branch=z9hG4bK-524287-1---e4c60f5fb131d816;rport Max-Forwards: 70 Contact: To: From: ;tag=567d0976 Call-ID: 9eJv0MT8qnzvkM1KyJ9EIA.. CSeq: 201 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Zoiper r28827 Authorization: Digest username="oeramos",realm="192.168.130.68",nonce="0d217ff6",uri="sip:192.168.130.68;transport=UDP",response="32a101d5ad80877a3a79175add7c5757",algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to 192.168.130.104:64251 (no NAT) Sending to 192.168.130.104:64251 (no NAT) <--- Transmitting (no NAT) to 192.168.130.104:64251 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.130.104:64251;branch=z9hG4bK-524287-1---e4c60f5fb131d816;received=192.168.130.104;rport=64251 From: ;tag=567d0976 To: ;tag=as637f5878 Call-ID: 9eJv0MT8qnzvkM1KyJ9EIA.. CSeq: 201 REGISTER Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="192.168.130.68", nonce="0622645e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9eJv0MT8qnzvkM1KyJ9EIA..' in 32000 ms (Method: REGISTER) antarespbx-camus*CLI>  <--- SIP read from UDP:192.168.130.104:64251 ---> REGISTER sip:192.168.130.68;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.130.104:64251;branch=z9hG4bK-524287-1---8885c6625fa3a315;rport Max-Forwards: 70 Contact: To: From: ;tag=567d0976 Call-ID: 9eJv0MT8qnzvkM1KyJ9EIA.. CSeq: 202 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Zoiper r28827 Authorization: Digest username="oeramos",realm="192.168.130.68",nonce="0622645e",uri="sip:192.168.130.68;transport=UDP",response="9094e23bbe9408100bbe19629ff8c34b",algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to 192.168.130.104:64251 (no NAT) antarespbx-camus*CLI> Reliably Transmitting (no NAT) to 192.168.130.104:64251: OPTIONS sip:oeramos@192.168.130.104:64251;rinstance=f31723138230a356;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK64115960 Max-Forwards: 70 From: "asterisk" ;tag=as5955ab70 To: Contact: Call-ID: 2377ab882ed77f1c45b086454b04422f@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:01:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- Transmitting (no NAT) to 192.168.130.104:64251 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.104:64251;branch=z9hG4bK-524287-1---8885c6625fa3a315;received=192.168.130.104;rport=64251 From: ;tag=567d0976 To: ;tag=as637f5878 Call-ID: 9eJv0MT8qnzvkM1KyJ9EIA.. CSeq: 202 REGISTER Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Fri, 16 Jan 2015 23:01:23 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9eJv0MT8qnzvkM1KyJ9EIA..' in 32000 ms (Method: REGISTER) antarespbx-camus*CLI>  <--- SIP read from UDP:192.168.130.104:64251 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK64115960 Contact: To: ;tag=b26ed818 From: "asterisk" ;tag=as5955ab70 Call-ID: 2377ab882ed77f1c45b086454b04422f@192.168.130.68:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Zoiper r28827 Allow-Events: presence, kpml Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (14 headers 0 lines) --- antarespbx-camus*CLI> Really destroying SIP dialog '2377ab882ed77f1c45b086454b04422f@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> BYE sip:93147920634@192.168.130.68:5060;transport=WS SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKe7AosWZkf9hh32BHNL0MzuH6enplz0w4;rport From: "Antares Soporte";tag=1BwXMeSgrReJyIGlHlss To: ;tag=as2806ac81 Call-ID: 62b4e37a-f281-97d6-3c63-d469f75901dd CSeq: 42836 BYE Content-Length: 0 Route: Max-Forwards: 70 Accept-Contact: *;+g.oma.sip-im Accept-Contact: *;+sip.ice Accept-Contact: *;language="es,fr" Authorization: Digest username="er",realm="192.168.130.68",nonce="731840e7",uri="sip:93147920634@192.168.130.68:5060;transport=WS",response="89ca6e5c5e681b3bbbebe372532f2b7b",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: ER Technology LTDA <-------------> --- (15 headers 0 lines) --- Scheduling destruction of SIP dialog '62b4e37a-f281-97d6-3c63-d469f75901dd' in 6400 ms (Method: BYE) <--- Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKe7AosWZkf9hh32BHNL0MzuH6enplz0w4;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=1BwXMeSgrReJyIGlHlss To: ;tag=as2806ac81 Call-ID: 62b4e37a-f281-97d6-3c63-d469f75901dd CSeq: 42836 BYE Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> antarespbx-camus*CLI>  -- Channel SIP/er-00000018 left 'simple_bridge' basic-bridge antarespbx-camus*CLI>  -- Channel SIP/2062589054-00000019 left 'simple_bridge' basic-bridge antarespbx-camus*CLI>  -- Executing [caller@hangup-context:1] AGI("SIP/2062589054-00000019", "agi://127.0.0.1:4573/?app=posthangup&recordid=&duration=9") in new stack antarespbx-camus*CLI>  -- AGI Script Executing Application: (Goto) Options: (hangup-context,called,1) -- Goto (hangup-context,called,1) antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=posthangup&recordid=&duration=9: u'UNIQUEID:1421449247.49' antarespbx-camus*CLI>  -- AGI Script agi://127.0.0.1:4573/?app=mainDial&type=normal&dialscheme=voz2voiceer&dialrule=22&delete_initial_digits=1&delete_final_digits=0&prefix=57&sufix=&sipuri=sip:er@df7jal23ls0d.invalid&username=®isterid= completed, returning 4 [Jan 16 18:01:38] ERROR[20495][C-0000000c]: utils.c:1371 ast_carefulwrite: write() returned error: Broken pipe == Spawn extension (hangup-context, called, 0) exited non-zero on 'SIP/er-00000018' antarespbx-camus*CLI>  == MixMonitor close filestream (mixed) == End MixMonitor Recording SIP/er-00000018 antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=posthangup&recordid=&duration=9: u'callerid:93147920634' antarespbx-camus*CLI> agi://127.0.0.1:4573/?app=posthangup&recordid=&duration=9: u'calleridname:unknown' antarespbx-camus*CLI>  -- AGI Script Executing Application: (Hangup) Options: () antarespbx-camus*CLI>  -- AGI Script agi://127.0.0.1:4573/?app=posthangup&recordid=&duration=9 completed, returning 4 [Jan 16 18:01:38] ERROR[20498][C-0000000c]: utils.c:1371 ast_carefulwrite: write() returned error: Broken pipe == Spawn extension (hangup-context, caller, 1) exited non-zero on 'SIP/2062589054-00000019' Scheduling destruction of SIP dialog '4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 50.116.31.15:5060 Reliably Transmitting (no NAT) to 50.116.31.15:5060: BYE sip:573147920634@50.116.31.15:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK1262d63b Max-Forwards: 70 From: ;tag=as3393cee8 To: ;tag=as6b55c4d7 Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 13.0.0 Authorization: Digest username="2062589054", realm="asterisk", algorithm=MD5, uri="sip:573147920634@50.116.31.15:5060", nonce="3d41b320", response="d2324e240b66bed02404554da353d7c5" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK1262d63b;received=181.62.249.112;rport=2048 From: ;tag=as3393cee8 To: ;tag=as6b55c4d7 Call-ID: 4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060 CSeq: 104 BYE Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '4e280c050c01dda721bbaa2319f93615@192.168.130.68:5060' Method: INVITE antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> REGISTER sip:192.168.130.68 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKfQukH29VqAxt3JdiNJ2T9VymySMTrazu;rport From: "Antares Soporte";tag=aJEXkgub0eq4X2lwxkJc To: "Antares Soporte" Contact: "Antares Soporte";expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr" Call-ID: df282f12-a3ea-dc9d-8a07-08673dce3527 CSeq: 61174 REGISTER Content-Length: 0 Route: Max-Forwards: 70 Authorization: Digest username="er",realm="192.168.130.68",nonce="08bac52d",uri="sip:192.168.130.68",response="0165b1c6baa5bdd722d46a2f6620f7a2",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: ER Technology LTDA <-------------> --- (13 headers 0 lines) --- <--- Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKfQukH29VqAxt3JdiNJ2T9VymySMTrazu;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=aJEXkgub0eq4X2lwxkJc To: "Antares Soporte";tag=as5899cdf9 Call-ID: df282f12-a3ea-dc9d-8a07-08673dce3527 CSeq: 61174 REGISTER Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="192.168.130.68", nonce="0927932f" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'df282f12-a3ea-dc9d-8a07-08673dce3527' in 32000 ms (Method: REGISTER) antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> REGISTER sip:192.168.130.68 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKrpHGL95VBzAAryzcBdWTab9WWFoMuQMo;rport From: "Antares Soporte";tag=aJEXkgub0eq4X2lwxkJc To: "Antares Soporte" Contact: "Antares Soporte";expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr" Call-ID: df282f12-a3ea-dc9d-8a07-08673dce3527 CSeq: 61175 REGISTER Content-Length: 0 Route: Max-Forwards: 70 Authorization: Digest username="er",realm="192.168.130.68",nonce="0927932f",uri="sip:192.168.130.68",response="08de9ba62df2977caa42c57ce4e5f32f",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18 Organization: ER Technology LTDA <-------------> antarespbx-camus*CLI> --- (13 headers 0 lines) --- antarespbx-camus*CLI> Reliably Transmitting (NAT) to 192.168.130.102:57026: OPTIONS sip:er@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.130.68:5060;branch=z9hG4bK1cfdab57;rport Max-Forwards: 70 From: "asterisk" ;tag=as366569f4 To: Contact: Call-ID: 6a001a8568ca6eb07678fc3b49f2192d@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:01:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- Transmitting (NAT) to 192.168.130.102:57026 ---> SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKrpHGL95VBzAAryzcBdWTab9WWFoMuQMo;received=192.168.130.102;rport=57026 From: "Antares Soporte";tag=aJEXkgub0eq4X2lwxkJc To: "Antares Soporte";tag=as5899cdf9 Call-ID: df282f12-a3ea-dc9d-8a07-08673dce3527 CSeq: 61175 REGISTER Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 200 Contact: ;expires=200 Date: Fri, 16 Jan 2015 23:01:39 GMT Content-Length: 0 <------------> antarespbx-camus*CLI> Scheduling destruction of SIP dialog 'df282f12-a3ea-dc9d-8a07-08673dce3527' in 32000 ms (Method: REGISTER) antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> SIP/2.0 405 Method Not Allowed Via: SIP/2.0/WS 192.168.130.68:5060;rport=5060;branch=z9hG4bK1cfdab57 From: "asterisk";tag=as366569f4 To: Call-ID: 6a001a8568ca6eb07678fc3b49f2192d@192.168.130.68:5060 CSeq: 102 OPTIONS Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (7 headers 0 lines) --- antarespbx-camus*CLI> Really destroying SIP dialog '6a001a8568ca6eb07678fc3b49f2192d@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> OPTIONS sip:s@192.168.130.68:5060 SIP/2.0 Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK5d787919;rport Max-Forwards: 70 From: "asterisk" ;tag=as63b43cb0 To: Contact: Call-ID: 70c1370461a3643248fd0a8629ae3daa@50.116.31.15:5060 CSeq: 102 OPTIONS User-Agent: VozOrgSIP1 Date: Fri, 16 Jan 2015 23:01:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 50.116.31.15:5060 (no NAT) Looking for s in incoming (domain 192.168.130.68) <--- Transmitting (no NAT) to 50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK5d787919;received=50.116.31.15;rport=5060 From: "asterisk" ;tag=as63b43cb0 To: ;tag=as2020433a Call-ID: 70c1370461a3643248fd0a8629ae3daa@50.116.31.15:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '70c1370461a3643248fd0a8629ae3daa@50.116.31.15:5060' in 32000 ms (Method: OPTIONS) antarespbx-camus*CLI> Really destroying SIP dialog '62b4e37a-f281-97d6-3c63-d469f75901dd' Method: BYE antarespbx-camus*CLI> Really destroying SIP dialog '56280500777e6f8945c2cb5034e7f1eb@50.116.31.15:5060' Method: OPTIONS antarespbx-camus*CLI> Really destroying SIP dialog '9eJv0MT8qnzvkM1KyJ9EIA..' Method: REGISTER antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> OPTIONS sip:s@192.168.130.68:5060 SIP/2.0 Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK3be1b470;rport Max-Forwards: 70 From: "asterisk" ;tag=as399f3442 To: Contact: Call-ID: 4caa2716147789e5134a5a516a97b2c8@50.116.31.15:5060 CSeq: 102 OPTIONS User-Agent: VozOrgSIP1 Date: Fri, 16 Jan 2015 23:02:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 50.116.31.15:5060 (no NAT) Looking for s in incoming (domain 192.168.130.68) <--- Transmitting (no NAT) to 50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK3be1b470;received=50.116.31.15;rport=5060 From: "asterisk" ;tag=as399f3442 To: ;tag=as52a69369 Call-ID: 4caa2716147789e5134a5a516a97b2c8@50.116.31.15:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '4caa2716147789e5134a5a516a97b2c8@50.116.31.15:5060' in 32000 ms (Method: OPTIONS) antarespbx-camus*CLI> Reliably Transmitting (no NAT) to 208.66.193.46:5060: OPTIONS sip:sip.ipsofactum.com SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK623cf5fe Max-Forwards: 70 From: "asterisk" ;tag=as5b0ca140 To: Contact: Call-ID: 79fde3aa37cc85734359b6947d5e81f7@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:02:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI> Reliably Transmitting (no NAT) to 50.116.31.15:5060: OPTIONS sip:sip1.voztovoice.org SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK736e539b Max-Forwards: 70 From: "asterisk" ;tag=as301e0bb0 To: Contact: Call-ID: 55a3801760a486c306a43566032383ee@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:02:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:208.66.193.46:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK623cf5fe;received=181.62.249.112;rport=2048 From: "asterisk" ;tag=as5b0ca140 To: ;tag=as1aa95d34 Call-ID: 79fde3aa37cc85734359b6947d5e81f7@192.168.130.68:5060 CSeq: 102 OPTIONS Server: Ipsoswitch - VoIP Powered by IPSOFACTUM Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '79fde3aa37cc85734359b6947d5e81f7@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK736e539b;received=181.62.249.112;rport=2048 From: "asterisk" ;tag=as301e0bb0 To: ;tag=as24ddb5b7 Call-ID: 55a3801760a486c306a43566032383ee@192.168.130.68:5060 CSeq: 102 OPTIONS Server: VozOrgSIP1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '55a3801760a486c306a43566032383ee@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Reliably Transmitting (NAT) to 173.192.95.227:5060: OPTIONS sip:173.192.95.227 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK354afba5;rport Max-Forwards: 70 From: "asterisk" ;tag=as7497081b To: Contact: Call-ID: 4b8f4c4549948fa52b0e78e90da6140e@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:02:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:173.192.95.227:5060 ---> SIP/2.0 200 OK CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK354afba5;rport From: "asterisk" ;tag=as7497081b Call-ID: 4b8f4c4549948fa52b0e78e90da6140e@192.168.130.68:5060 To: ;tag=160158151738 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (8 headers 0 lines) --- antarespbx-camus*CLI> Really destroying SIP dialog '4b8f4c4549948fa52b0e78e90da6140e@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Reliably Transmitting (NAT) to 173.255.199.156:5060: OPTIONS sip:173.255.199.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK6fcc2858;rport Max-Forwards: 70 From: "asterisk" ;tag=as32fc2041 To: Contact: Call-ID: 4c942ddc30cb3b7640837e2713014f63@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:02:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:173.255.199.156:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK6fcc2858;received=181.62.249.112;rport=2048 From: "asterisk" ;tag=as32fc2041 To: ;tag=as2c6d228b Call-ID: 4c942ddc30cb3b7640837e2713014f63@192.168.130.68:5060 CSeq: 102 OPTIONS Server: VozOrgSIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (12 headers 0 lines) --- antarespbx-camus*CLI> Really destroying SIP dialog '4c942ddc30cb3b7640837e2713014f63@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Really destroying SIP dialog 'df282f12-a3ea-dc9d-8a07-08673dce3527' Method: REGISTER antarespbx-camus*CLI> Really destroying SIP dialog '70c1370461a3643248fd0a8629ae3daa@50.116.31.15:5060' Method: OPTIONS antarespbx-camus*CLI>  <--- SIP read from UDP:50.116.31.15:5060 ---> OPTIONS sip:s@192.168.130.68:5060 SIP/2.0 Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK52d8b968;rport Max-Forwards: 70 From: "asterisk" ;tag=as18868abd To: Contact: Call-ID: 4d093a080743691e4dadce7271f387c8@50.116.31.15:5060 CSeq: 102 OPTIONS User-Agent: VozOrgSIP1 Date: Fri, 16 Jan 2015 23:02:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 50.116.31.15:5060 (no NAT) Looking for s in incoming (domain 192.168.130.68) <--- Transmitting (no NAT) to 50.116.31.15:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 50.116.31.15:5060;branch=z9hG4bK52d8b968;received=50.116.31.15;rport=5060 From: "asterisk" ;tag=as18868abd To: ;tag=as646fe4a8 Call-ID: 4d093a080743691e4dadce7271f387c8@50.116.31.15:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '4d093a080743691e4dadce7271f387c8@50.116.31.15:5060' in 32000 ms (Method: OPTIONS) antarespbx-camus*CLI> Reliably Transmitting (no NAT) to 192.168.130.104:64251: OPTIONS sip:oeramos@192.168.130.104:64251;rinstance=f31723138230a356;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK012ec36c Max-Forwards: 70 From: "asterisk" ;tag=as62292842 To: Contact: Call-ID: 1115532c2d9140ff0889929d4d144fd3@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:02:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from UDP:192.168.130.104:64251 ---> REGISTER sip:192.168.130.68;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.130.104:64251;branch=z9hG4bK-524287-1---c651a9ed0dd80ce4;rport Max-Forwards: 70 Contact: To: From: ;tag=567d0976 Call-ID: 9eJv0MT8qnzvkM1KyJ9EIA.. CSeq: 203 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Zoiper r28827 Authorization: Digest username="oeramos",realm="192.168.130.68",nonce="0622645e",uri="sip:192.168.130.68;transport=UDP",response="9094e23bbe9408100bbe19629ff8c34b",algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to 192.168.130.104:64251 (no NAT) Sending to 192.168.130.104:64251 (no NAT) <--- Transmitting (no NAT) to 192.168.130.104:64251 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.130.104:64251;branch=z9hG4bK-524287-1---c651a9ed0dd80ce4;received=192.168.130.104;rport=64251 From: ;tag=567d0976 To: ;tag=as6b303d5b Call-ID: 9eJv0MT8qnzvkM1KyJ9EIA.. CSeq: 203 REGISTER Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="192.168.130.68", nonce="5935959d" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9eJv0MT8qnzvkM1KyJ9EIA..' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.130.104:64251 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK012ec36c Contact: To: ;tag=9c3d0e48 From: "asterisk" ;tag=as62292842 Call-ID: 1115532c2d9140ff0889929d4d144fd3@192.168.130.68:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Zoiper r28827 Allow-Events: presence, kpml Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '1115532c2d9140ff0889929d4d144fd3@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI>  <--- SIP read from UDP:192.168.130.104:64251 ---> REGISTER sip:192.168.130.68;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.130.104:64251;branch=z9hG4bK-524287-1---bb262fdc05e6ae43;rport Max-Forwards: 70 Contact: To: From: ;tag=567d0976 Call-ID: 9eJv0MT8qnzvkM1KyJ9EIA.. CSeq: 204 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Zoiper r28827 Authorization: Digest username="oeramos",realm="192.168.130.68",nonce="5935959d",uri="sip:192.168.130.68;transport=UDP",response="01f1c7bda588535c242b0864d5c793d8",algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to 192.168.130.104:64251 (no NAT) Reliably Transmitting (no NAT) to 192.168.130.104:64251: OPTIONS sip:oeramos@192.168.130.104:64251;rinstance=f31723138230a356;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK7a7d1ae4 Max-Forwards: 70 From: "asterisk" ;tag=as3b46b72b To: Contact: Call-ID: 1545cea33fdfe97e360c62064ee41b0a@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:02:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 192.168.130.104:64251 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.104:64251;branch=z9hG4bK-524287-1---bb262fdc05e6ae43;received=192.168.130.104;rport=64251 From: ;tag=567d0976 To: ;tag=as6b303d5b Call-ID: 9eJv0MT8qnzvkM1KyJ9EIA.. CSeq: 204 REGISTER Server: Asterisk PBX 13.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Fri, 16 Jan 2015 23:02:23 GMT Content-Length: 0 <------------> antarespbx-camus*CLI> Scheduling destruction of SIP dialog '9eJv0MT8qnzvkM1KyJ9EIA..' in 32000 ms (Method: REGISTER) antarespbx-camus*CLI>  <--- SIP read from UDP:192.168.130.104:64251 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.68:5060;branch=z9hG4bK7a7d1ae4 Contact: To: ;tag=91b3d65d From: "asterisk" ;tag=as3b46b72b Call-ID: 1545cea33fdfe97e360c62064ee41b0a@192.168.130.68:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri User-Agent: Zoiper r28827 Allow-Events: presence, kpml Content-Length: 0 <-------------> antarespbx-camus*CLI> --- (14 headers 0 lines) --- antarespbx-camus*CLI> Really destroying SIP dialog '1545cea33fdfe97e360c62064ee41b0a@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Really destroying SIP dialog '4caa2716147789e5134a5a516a97b2c8@50.116.31.15:5060' Method: OPTIONS antarespbx-camus*CLI> Reliably Transmitting (NAT) to 192.168.130.102:57026: OPTIONS sip:er@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.130.68:5060;branch=z9hG4bK28b71c9e;rport Max-Forwards: 70 From: "asterisk" ;tag=as36e9a0f6 To: Contact: Call-ID: 73ed6bb94a5624557adeb6f4744b78e0@192.168.130.68:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.0.0 Date: Fri, 16 Jan 2015 23:02:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- antarespbx-camus*CLI>  <--- SIP read from WS:192.168.130.102:57026 ---> SIP/2.0 405 Method Not Allowed Via: SIP/2.0/WS 192.168.130.68:5060;rport=5060;branch=z9hG4bK28b71c9e From: "asterisk";tag=as36e9a0f6 To: Call-ID: 73ed6bb94a5624557adeb6f4744b78e0@192.168.130.68:5060 CSeq: 102 OPTIONS Content-Length: 0 <-------------> --- (7 headers 0 lines) --- antarespbx-camus*CLI> Really destroying SIP dialog '73ed6bb94a5624557adeb6f4744b78e0@192.168.130.68:5060' Method: OPTIONS antarespbx-camus*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups