Realmente estranho. Tu pode ativar o log full do Asterisk (/etc/asterisk/logger.conf) e depois recarregar o Asterisk e aih ativar o debug (core set debug 100) e fazer outro teste. Os logs vao ficar no /var/log/asterisk/full (o debug normalmente nao aparece no console).
Talvez assim tu encontre o problema. Concordo com o Bruno, DNS pode ser um fator complicante. []s Marcelo H. Terres <[email protected]> IM: [email protected] https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On Sat, 3 Nov 2018 at 00:13, Giliardy Arena <[email protected]> wrote: > > Oi ! > Obrigado pela resposta e pela ajuda. > Desculpe, não sei como enviar o arquivo. > > Nesta resposta estou tentando anexar via gmail. > Espero que funcione, mas se não funcionar e puder me indicar a maneira > correta. > > Utilizei a seguinte sintaxe : > > tcpdump -i ens192 src or dst 172.17.39.41 or 172.17.39.42 or 172.17.39.43 -w > capture4.cap > > > Sigo pesquisando =) > > > Atenciosamente, > Giliardy Correia Arena. > > > > > Em sex, 2 de nov de 2018 às 17:22, Giliardy Arena <[email protected]> > escreveu: >> >> Obrigado Rogerio. >> Esse comando não me ajudou muito ;/ >> Notei o comportamento parecido com do TCPdump , veja se consegue entender >> algo que possa explicar >> >> >> >> >> infoasterisk*CLI> >> >> >> Recebo esse INVITE logo quando faço a chamada do Call Manager para o Asterisk >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> INVITE sip:[email protected]:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Date: Fri, 02 Nov 2018 19:11:47 GMT >> Call-ID: [email protected] >> Supported: timer,resource-priority,replaces >> Min-SE: 1800 >> User-Agent: Cisco-CUCM10.5 >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> CSeq: 101 INVITE >> Expires: 180 >> Allow-Events: presence, kpml >> Supported: X-cisco-srtp-fallback,X-cisco-original-called >> Cisco-Guid: 0595288320-0000065536-0000222315-0707203500 >> Session-Expires: 1800 >> P-Asserted-Identity: "Giliardy Arena" <sip:[email protected]> >> Remote-Party-ID: "Giliardy Arena" >> <sip:[email protected]>;party=calling;screen=yes;privacy=off >> Contact: >> <sip:[email protected]:5060>;+u.sip!devicename.ccm.cisco.com="CSFGARENA";bfcp >> Max-Forwards: 69 >> Content-Type: application/sdp >> Content-Length: 206 >> >> v=0 >> o=CiscoSystemsCCM-SIP 95126358 1 IN IP4 172.17.39.42 >> s=SIP Call >> c=IN IP4 172.17.231.249 >> t=0 0 >> m=audio 18104 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> <-------------> >> --- (22 headers 9 lines) --- >> Sending to 172.17.39.42:5060 (no NAT) >> Sending to 172.17.39.42:5060 (no NAT) >> Using INVITE request as basis request - >> [email protected] >> Found peer 'callman02' for '9770' from 172.17.39.42:5060 >> == Using SIP RTP CoS mark 5 >> Found RTP audio format 0 >> Found RTP audio format 101 >> Found audio description format PCMU for ID 0 >> Found audio description format telephone-event for ID 101 >> Capabilities: us - (ulaw), peer - >> audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) >> vent|) >> > 0x7f9c840327f0 -- Strict RTP learning after remote address set to: >> 172.17.231.249:18104 >> Peer audio RTP is at port 172.17.231.249:18104 >> Looking for 2001 in ramais (domain 172.17.37.129) >> sip_route_dump: route/path hop: <sip:[email protected]:5060> >> >> >> >> Só me chamaram atenção o >> >> Found peer 'callman02' for '9770' from 172.17.39.42:5060 >> Looking for 2001 in ramais (domain 172.17.37.129) >> >> Mas não me parece anormal, pois não indica nada . >> >> >> >> >> Daqui para baixo, já é quando a chamada está tocando. >> Portanto, eu não enxergo o que está se passando na demora dos 30 segundos :( >> Só via TCPdump que vejo ele conversando com os servidores. >> >> >> >> >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba;received=172.17.39.42 >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 101 INVITE >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Session-Expires: 1800;refresher=uas >> Contact: <sip:[email protected]:5060> >> Content-Length: 0 >> >> >> <------------> >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> INVITE sip:[email protected]:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Date: Fri, 02 Nov 2018 19:11:48 GMT >> Call-ID: [email protected] >> Supported: timer,resource-priority,replaces >> Min-SE: 1800 >> User-Agent: Cisco-CUCM10.5 >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> CSeq: 101 INVITE >> Expires: 180 >> Allow-Events: presence, kpml >> Supported: X-cisco-srtp-fallback,X-cisco-original-called >> Cisco-Guid: 0595288320-0000065536-0000222315-0707203500 >> Session-Expires: 1800 >> P-Asserted-Identity: "Giliardy Arena" <sip:[email protected]> >> Remote-Party-ID: "Giliardy Arena" >> <sip:[email protected]>;party=calling;screen=yes;privacy=off >> Contact: >> <sip:[email protected]:5060>;+u.sip!devicename.ccm.cisco.com="CSFGARENA";bfcp >> Max-Forwards: 69 >> Content-Type: application/sdp >> Content-Length: 206 >> >> v=0 >> o=CiscoSystemsCCM-SIP 95126358 1 IN IP4 172.17.39.42 >> s=SIP Call >> c=IN IP4 172.17.231.249 >> t=0 0 >> m=audio 18104 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> <-------------> >> --- (22 headers 9 lines) --- >> Ignoring this INVITE request >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba;received=172.17.39.42 >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 101 INVITE >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Session-Expires: 1800;refresher=uas >> Contact: <sip:[email protected]:5060> >> Content-Length: 0 >> >> >> <------------> >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> INVITE sip:[email protected]:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Date: Fri, 02 Nov 2018 19:11:49 GMT >> Call-ID: [email protected] >> Supported: timer,resource-priority,replaces >> Min-SE: 1800 >> User-Agent: Cisco-CUCM10.5 >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> CSeq: 101 INVITE >> Expires: 180 >> Allow-Events: presence, kpml >> Supported: X-cisco-srtp-fallback,X-cisco-original-called >> Cisco-Guid: 0595288320-0000065536-0000222315-0707203500 >> Session-Expires: 1800 >> P-Asserted-Identity: "Giliardy Arena" <sip:[email protected]> >> Remote-Party-ID: "Giliardy Arena" >> <sip:[email protected]>;party=calling;screen=yes;privacy=off >> Contact: >> <sip:[email protected]:5060>;+u.sip!devicename.ccm.cisco.com="CSFGARENA";bfcp >> Max-Forwards: 69 >> Content-Type: application/sdp >> Content-Length: 206 >> >> v=0 >> o=CiscoSystemsCCM-SIP 95126358 1 IN IP4 172.17.39.42 >> s=SIP Call >> c=IN IP4 172.17.231.249 >> t=0 0 >> m=audio 18104 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> <-------------> >> --- (22 headers 9 lines) --- >> Ignoring this INVITE request >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba;received=172.17.39.42 >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 101 INVITE >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Session-Expires: 1800;refresher=uas >> Contact: <sip:[email protected]:5060> >> Content-Length: 0 >> >> >> <------------> >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> INVITE sip:[email protected]:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Date: Fri, 02 Nov 2018 19:11:51 GMT >> Call-ID: [email protected] >> Supported: timer,resource-priority,replaces >> Min-SE: 1800 >> User-Agent: Cisco-CUCM10.5 >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> CSeq: 101 INVITE >> Expires: 180 >> Allow-Events: presence, kpml >> Supported: X-cisco-srtp-fallback,X-cisco-original-called >> Cisco-Guid: 0595288320-0000065536-0000222315-0707203500 >> Session-Expires: 1800 >> P-Asserted-Identity: "Giliardy Arena" <sip:[email protected]> >> Remote-Party-ID: "Giliardy Arena" >> <sip:[email protected]>;party=calling;screen=yes;privacy=off >> Contact: >> <sip:[email protected]:5060>;+u.sip!devicename.ccm.cisco.com="CSFGARENA";bfcp >> Max-Forwards: 69 >> Content-Type: application/sdp >> Content-Length: 206 >> >> v=0 >> o=CiscoSystemsCCM-SIP 95126358 1 IN IP4 172.17.39.42 >> s=SIP Call >> c=IN IP4 172.17.231.249 >> t=0 0 >> m=audio 18104 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> <-------------> >> --- (22 headers 9 lines) --- >> Ignoring this INVITE request >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba;received=172.17.39.42 >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 101 INVITE >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Session-Expires: 1800;refresher=uas >> Contact: <sip:[email protected]:5060> >> Content-Length: 0 >> >> >> <------------> >> >> <--- SIP read from UDP:172.17.90.170:50147 ---> >> >> >> <-------------> >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> INVITE sip:[email protected]:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Date: Fri, 02 Nov 2018 19:11:55 GMT >> Call-ID: [email protected] >> Supported: timer,resource-priority,replaces >> Min-SE: 1800 >> User-Agent: Cisco-CUCM10.5 >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> CSeq: 101 INVITE >> Expires: 180 >> Allow-Events: presence, kpml >> Supported: X-cisco-srtp-fallback,X-cisco-original-called >> Cisco-Guid: 0595288320-0000065536-0000222315-0707203500 >> Session-Expires: 1800 >> P-Asserted-Identity: "Giliardy Arena" <sip:[email protected]> >> Remote-Party-ID: "Giliardy Arena" >> <sip:[email protected]>;party=calling;screen=yes;privacy=off >> Contact: >> <sip:[email protected]:5060>;+u.sip!devicename.ccm.cisco.com="CSFGARENA";bfcp >> Max-Forwards: 69 >> Content-Type: application/sdp >> Content-Length: 206 >> >> v=0 >> o=CiscoSystemsCCM-SIP 95126358 1 IN IP4 172.17.39.42 >> s=SIP Call >> c=IN IP4 172.17.231.249 >> t=0 0 >> m=audio 18104 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> <-------------> >> --- (22 headers 9 lines) --- >> Ignoring this INVITE request >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba;received=172.17.39.42 >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 101 INVITE >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Session-Expires: 1800;refresher=uas >> Contact: <sip:[email protected]:5060> >> Content-Length: 0 >> >> >> <------------> >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:11:56 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.43:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Sending to 172.17.39.43:5060 (no NAT) >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.43:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9;received=172.17.39.43 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129>;tag=as6cdc175e >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:11:56 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.43:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.43:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9;received=172.17.39.43 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129>;tag=as6cdc175e >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:11:57 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.42:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Sending to 172.17.39.42:5060 (no NAT) >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb;received=172.17.39.42 >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129>;tag=as3f81a07d >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:11:57 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.43:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.43:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9;received=172.17.39.43 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129>;tag=as6cdc175e >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:11:58 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.42:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb;received=172.17.39.42 >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129>;tag=as3f81a07d >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:11:59 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.42:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb;received=172.17.39.42 >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129>;tag=as3f81a07d >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:11:59 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.43:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.43:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9;received=172.17.39.43 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129>;tag=as6cdc175e >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:12:01 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.42:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb;received=172.17.39.42 >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129>;tag=as3f81a07d >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> INVITE sip:[email protected]:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Date: Fri, 02 Nov 2018 19:12:03 GMT >> Call-ID: [email protected] >> Supported: timer,resource-priority,replaces >> Min-SE: 1800 >> User-Agent: Cisco-CUCM10.5 >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> CSeq: 101 INVITE >> Expires: 180 >> Allow-Events: presence, kpml >> Supported: X-cisco-srtp-fallback,X-cisco-original-called >> Cisco-Guid: 0595288320-0000065536-0000222315-0707203500 >> Session-Expires: 1800 >> P-Asserted-Identity: "Giliardy Arena" <sip:[email protected]> >> Remote-Party-ID: "Giliardy Arena" >> <sip:[email protected]>;party=calling;screen=yes;privacy=off >> Contact: >> <sip:[email protected]:5060>;+u.sip!devicename.ccm.cisco.com="CSFGARENA";bfcp >> Max-Forwards: 69 >> Content-Type: application/sdp >> Content-Length: 206 >> >> v=0 >> o=CiscoSystemsCCM-SIP 95126358 1 IN IP4 172.17.39.42 >> s=SIP Call >> c=IN IP4 172.17.231.249 >> t=0 0 >> m=audio 18104 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> <-------------> >> --- (22 headers 9 lines) --- >> Ignoring this INVITE request >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba;received=172.17.39.42 >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 101 INVITE >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Session-Expires: 1800;refresher=uas >> Contact: <sip:[email protected]:5060> >> Content-Length: 0 >> >> >> <------------> >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:12:03 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.43:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.43:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9;received=172.17.39.43 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129>;tag=as6cdc175e >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:12:05 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.42:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb;received=172.17.39.42 >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129>;tag=as3f81a07d >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:12:07 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.43:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.43:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9;received=172.17.39.43 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129>;tag=as6cdc175e >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:12:09 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.42:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb;received=172.17.39.42 >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129>;tag=as3f81a07d >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:12:11 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.43:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.43:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.43:5060;branch=z9hG4bK2bff065528a6e9;received=172.17.39.43 >> From: <sip:172.17.39.43>;tag=80797582 >> To: <sip:172.17.37.129>;tag=as6cdc175e >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:12:13 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.42:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbfa298926fb;received=172.17.39.42 >> From: <sip:172.17.39.42>;tag=696000702 >> To: <sip:172.17.37.129>;tag=as3f81a07d >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> -- Executing [2001@ramais:1] Dial("SIP/callman02-00000091", "SIP/2001") >> in new stack >> == Using SIP RTP CoS mark 5 >> Audio is at 16502 >> Adding codec ulaw to SDP >> Adding non-codec 0x1 (telephone-event) to SDP >> Reliably Transmitting (no NAT) to 172.17.90.170:50147: >> INVITE sip:[email protected]:50147;rinstance=a175c2caa1292efd SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK7e5232b4 >> Max-Forwards: 70 >> From: "Giliardy Arena" <sip:[email protected]>;tag=as1b69e3fc >> To: <sip:[email protected]:50147;rinstance=a175c2caa1292efd> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 INVITE >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:12:20 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Type: application/sdp >> Content-Length: 252 >> >> v=0 >> o=root 388968980 388968980 IN IP4 172.17.37.129 >> s=Asterisk PBX 13.23.1 >> c=IN IP4 172.17.37.129 >> t=0 0 >> m=audio 16502 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=maxptime:150 >> a=sendrecv >> >> --- >> -- Called SIP/2001 >> << [ TYPE: Control (4) SUBCLASS: Unknown control '22' (22) ] >> [SIP/2001-00000092] >> >> <--- SIP read from UDP:172.17.90.170:50147 ---> >> SIP/2.0 180 Ringing >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK7e5232b4 >> Contact: <sip:[email protected]:50147;rinstance=a175c2caa1292efd> >> To: >> "2001"<sip:[email protected]:50147;rinstance=a175c2caa1292efd>;tag=a0a27e40 >> From: "Giliardy Arena" <sip:[email protected]>;tag=as1b69e3fc >> Call-ID: [email protected]:5060 >> CSeq: 102 INVITE >> User-Agent: X-Lite release 5.4.0 stamp 94388 >> Allow-Events: talk, hold >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> sip_route_dump: route/path hop: >> <sip:[email protected]:50147;rinstance=a175c2caa1292efd> >> << [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] >> [SIP/2001-00000092] >> << [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/2001-00000092] >> -- SIP/2001-00000092 is ringing >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 180 Ringing >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba;received=172.17.39.42 >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]>;tag=as109d5c95 >> Call-ID: [email protected] >> CSeq: 101 INVITE >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Session-Expires: 1800;refresher=uas >> Contact: <sip:[email protected]:5060> >> Content-Length: 0 >> >> >> <------------> >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> CANCEL sip:[email protected]:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]> >> Date: Fri, 02 Nov 2018 19:12:03 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 CANCEL >> Max-Forwards: 70 >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> Sending to 172.17.39.42:5060 (no NAT) >> >> <--- Reliably Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 487 Request Terminated >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba;received=172.17.39.42 >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]>;tag=as109d5c95 >> Call-ID: [email protected] >> CSeq: 101 INVITE >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> <------------> >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba;received=172.17.39.42 >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]>;tag=as109d5c95 >> Call-ID: [email protected] >> CSeq: 101 CANCEL >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> <------------> >> << [ HANGUP (NULL) ] [SIP/callman02-00000091] >> ) >> Reliably Transmitting (no NAT) to 172.17.90.170:50147: >> CANCEL sip:[email protected]:50147;rinstance=a175c2caa1292efd SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK7e5232b4 >> Max-Forwards: 70 >> From: "Giliardy Arena" <sip:[email protected]>;tag=as1b69e3fc >> To: <sip:[email protected]:50147;rinstance=a175c2caa1292efd> >> Call-ID: [email protected]:5060 >> CSeq: 102 CANCEL >> User-Agent: Asterisk PBX 13.23.1 >> Content-Length: 0 >> >> >> --- >> ) >> == Spawn extension (ramais, 2001, 1) exited non-zero on >> 'SIP/callman02-00000091' >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> ACK sip:[email protected]:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba >> From: "Giliardy Arena" >> <sip:[email protected]>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796 >> To: <sip:[email protected]>;tag=as109d5c95 >> Date: Fri, 02 Nov 2018 19:12:03 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> Max-Forwards: 70 >> CSeq: 101 ACK >> Allow-Events: presence, kpml >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Really destroying SIP dialog >> '[email protected]' Method: ACK >> >> <--- SIP read from UDP:172.17.90.170:50147 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK7e5232b4 >> Contact: <sip:[email protected]:50147;rinstance=a175c2caa1292efd> >> To: <sip:[email protected]:50147;rinstance=a175c2caa1292efd>;tag=a0a27e40 >> From: "Giliardy Arena" <sip:[email protected]>;tag=as1b69e3fc >> Call-ID: [email protected]:5060 >> CSeq: 102 CANCEL >> User-Agent: X-Lite release 5.4.0 stamp 94388 >> Content-Length: 0 >> >> <-------------> >> --- (9 headers 0 lines) --- >> >> <--- SIP read from UDP:172.17.90.170:50147 ---> >> SIP/2.0 487 Request Terminated >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK7e5232b4 >> To: <sip:[email protected]:50147;rinstance=a175c2caa1292efd>;tag=a0a27e40 >> From: "Giliardy Arena" <sip:[email protected]>;tag=as1b69e3fc >> Call-ID: [email protected]:5060 >> CSeq: 102 INVITE >> User-Agent: X-Lite release 5.4.0 stamp 94388 >> Content-Length: 0 >> >> <-------------> >> --- (8 headers 0 lines) --- >> Transmitting (no NAT) to 172.17.90.170:50147: >> ACK sip:[email protected]:50147;rinstance=a175c2caa1292efd SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK7e5232b4 >> Max-Forwards: 70 >> From: "Giliardy Arena" <sip:[email protected]>;tag=as1b69e3fc >> To: <sip:[email protected]:50147;rinstance=a175c2caa1292efd>;tag=a0a27e40 >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 ACK >> User-Agent: Asterisk PBX 13.23.1 >> Content-Length: 0 >> >> >> --- >> ) >> >> <--- SIP read from UDP:172.17.90.170:50147 ---> >> >> >> <-------------> >> Reliably Transmitting (no NAT) to 172.17.39.41:5060: >> OPTIONS sip:172.17.39.41 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK38f54aae >> Max-Forwards: 70 >> From: "asterisk" <sip:[email protected]>;tag=as1a8e4d0e >> To: <sip:172.17.39.41> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:12:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> Reliably Transmitting (no NAT) to 172.17.39.42:5060: >> OPTIONS sip:172.17.39.42 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK5b39be3c >> Max-Forwards: 70 >> From: "asterisk" <sip:[email protected]>;tag=as2d9ec9dd >> To: <sip:172.17.39.42> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:12:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> Reliably Transmitting (no NAT) to 172.17.39.43:5060: >> OPTIONS sip:172.17.39.43 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK64dda269 >> Max-Forwards: 70 >> From: "asterisk" <sip:[email protected]>;tag=as2deca9a9 >> To: <sip:172.17.39.43> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:12:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK5b39be3c >> From: "asterisk" <sip:[email protected]>;tag=as2d9ec9dd >> To: <sip:172.17.39.42>;tag=2130805835 >> Date: Fri, 02 Nov 2018 19:12:24 GMT >> Call-ID: [email protected]:5060 >> Server: Cisco-CUCM10.5 >> CSeq: 102 OPTIONS >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> Really destroying SIP dialog >> '[email protected]:5060' Method: OPTIONS >> >> <--- SIP read from UDP:172.17.39.41:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK38f54aae >> From: "asterisk" <sip:[email protected]>;tag=as1a8e4d0e >> To: <sip:172.17.39.41>;tag=1670426499 >> Date: Fri, 02 Nov 2018 19:12:24 GMT >> Call-ID: [email protected]:5060 >> Server: Cisco-CUCM10.5 >> CSeq: 102 OPTIONS >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> Really destroying SIP dialog >> '[email protected]:5060' Method: OPTIONS >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK64dda269 >> From: "asterisk" <sip:[email protected]>;tag=as2deca9a9 >> To: <sip:172.17.39.43>;tag=876720778 >> Date: Fri, 02 Nov 2018 19:12:24 GMT >> Call-ID: [email protected]:5060 >> Server: Cisco-CUCM10.5 >> CSeq: 102 OPTIONS >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> Really destroying SIP dialog >> '[email protected]:5060' Method: OPTIONS >> >> <--- SIP read from UDP:172.17.39.41:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.41:5060;branch=z9hG4bK1c78375818693a >> From: <sip:172.17.39.41>;tag=482859734 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:12:38 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.41:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Sending to 172.17.39.41:5060 (no NAT) >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.41:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.41:5060;branch=z9hG4bK1c78375818693a;received=172.17.39.41 >> From: <sip:172.17.39.41>;tag=482859734 >> To: <sip:172.17.37.129>;tag=as3cb6d00b >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> Really destroying SIP dialog >> '[email protected]' Method: OPTIONS >> Really destroying SIP dialog >> '[email protected]' Method: OPTIONS >> Really destroying SIP dialog >> '[email protected]:5060' Method: INVITE >> >> <--- SIP read from UDP:172.17.90.170:50147 ---> >> >> >> <-------------> >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.43:5060;branch=z9hG4bK2bff19a0c5d2b >> From: <sip:172.17.39.43>;tag=1681901178 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:12:57 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.43:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Sending to 172.17.39.43:5060 (no NAT) >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.43:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.43:5060;branch=z9hG4bK2bff19a0c5d2b;received=172.17.39.43 >> From: <sip:172.17.39.43>;tag=1681901178 >> To: <sip:172.17.37.129>;tag=as39195b67 >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cc1f3db4dd06 >> From: <sip:172.17.39.42>;tag=654360426 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:12:57 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.42:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Sending to 172.17.39.42:5060 (no NAT) >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cc1f3db4dd06;received=172.17.39.42 >> From: <sip:172.17.39.42>;tag=654360426 >> To: <sip:172.17.37.129>;tag=as130c9560 >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> Really destroying SIP dialog >> '[email protected]' Method: OPTIONS >> >> <--- SIP read from UDP:172.17.90.170:50147 ---> >> >> >> <-------------> >> Reliably Transmitting (no NAT) to 172.17.39.42:5060: >> OPTIONS sip:172.17.39.42 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK780abac5 >> Max-Forwards: 70 >> From: "asterisk" <sip:[email protected]>;tag=as5db9427e >> To: <sip:172.17.39.42> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:13:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> Reliably Transmitting (no NAT) to 172.17.39.41:5060: >> OPTIONS sip:172.17.39.41 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK6a5047da >> Max-Forwards: 70 >> From: "asterisk" <sip:[email protected]>;tag=as3dded9ad >> To: <sip:172.17.39.41> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:13:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> Reliably Transmitting (no NAT) to 172.17.39.43:5060: >> OPTIONS sip:172.17.39.43 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK55c27356 >> Max-Forwards: 70 >> From: "asterisk" <sip:[email protected]>;tag=as773015ab >> To: <sip:172.17.39.43> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:13:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK780abac5 >> From: "asterisk" <sip:[email protected]>;tag=as5db9427e >> To: <sip:172.17.39.42>;tag=304370098 >> Date: Fri, 02 Nov 2018 19:13:24 GMT >> Call-ID: [email protected]:5060 >> Server: Cisco-CUCM10.5 >> CSeq: 102 OPTIONS >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> Really destroying SIP dialog >> '[email protected]:5060' Method: OPTIONS >> >> <--- SIP read from UDP:172.17.39.41:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK6a5047da >> From: "asterisk" <sip:[email protected]>;tag=as3dded9ad >> To: <sip:172.17.39.41>;tag=383686183 >> Date: Fri, 02 Nov 2018 19:13:24 GMT >> Call-ID: [email protected]:5060 >> Server: Cisco-CUCM10.5 >> CSeq: 102 OPTIONS >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK55c27356 >> From: "asterisk" <sip:[email protected]>;tag=as773015ab >> To: <sip:172.17.39.43>;tag=715549747 >> Date: Fri, 02 Nov 2018 19:13:24 GMT >> Call-ID: [email protected]:5060 >> Server: Cisco-CUCM10.5 >> CSeq: 102 OPTIONS >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> Really destroying SIP dialog >> '[email protected]:5060' Method: OPTIONS >> Really destroying SIP dialog >> '[email protected]:5060' Method: OPTIONS >> Really destroying SIP dialog >> '[email protected]' Method: OPTIONS >> Really destroying SIP dialog >> '[email protected]' Method: OPTIONS >> >> <--- SIP read from UDP:172.17.39.41:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.41:5060;branch=z9hG4bK1c784863d1629c >> From: <sip:172.17.39.41>;tag=175949742 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:13:38 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.41:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Sending to 172.17.39.41:5060 (no NAT) >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.41:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.41:5060;branch=z9hG4bK1c784863d1629c;received=172.17.39.41 >> From: <sip:172.17.39.41>;tag=175949742 >> To: <sip:172.17.37.129>;tag=as37437605 >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.90.170:50147 ---> >> >> >> <-------------> >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cc374d521817 >> From: <sip:172.17.39.42>;tag=1442708621 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:13:59 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.42:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Sending to 172.17.39.42:5060 (no NAT) >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cc374d521817;received=172.17.39.42 >> From: <sip:172.17.39.42>;tag=1442708621 >> To: <sip:172.17.37.129>;tag=as31b8a209 >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> Really destroying SIP dialog >> '[email protected]' Method: OPTIONS >> >> <--- SIP read from UDP:172.17.90.170:50147 ---> >> >> >> <-------------> >> Reliably Transmitting (no NAT) to 172.17.39.42:5060: >> OPTIONS sip:172.17.39.42 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK3d7cd47c >> Max-Forwards: 70 >> From: "asterisk" <sip:[email protected]>;tag=as753534e0 >> To: <sip:172.17.39.42> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:14:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> Reliably Transmitting (no NAT) to 172.17.39.41:5060: >> OPTIONS sip:172.17.39.41 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK3a5093e6 >> Max-Forwards: 70 >> From: "asterisk" <sip:[email protected]>;tag=as2f7fde70 >> To: <sip:172.17.39.41> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:14:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> Reliably Transmitting (no NAT) to 172.17.39.43:5060: >> OPTIONS sip:172.17.39.43 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK5121b2c6 >> Max-Forwards: 70 >> From: "asterisk" <sip:[email protected]>;tag=as2db68d44 >> To: <sip:172.17.39.43> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:14:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK3d7cd47c >> From: "asterisk" <sip:[email protected]>;tag=as753534e0 >> To: <sip:172.17.39.42>;tag=917613056 >> Date: Fri, 02 Nov 2018 19:14:24 GMT >> Call-ID: [email protected]:5060 >> Server: Cisco-CUCM10.5 >> CSeq: 102 OPTIONS >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK5121b2c6 >> From: "asterisk" <sip:[email protected]>;tag=as2db68d44 >> To: <sip:172.17.39.43>;tag=1666345757 >> Date: Fri, 02 Nov 2018 19:14:24 GMT >> Call-ID: [email protected]:5060 >> Server: Cisco-CUCM10.5 >> CSeq: 102 OPTIONS >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> Really destroying SIP dialog >> '[email protected]:5060' Method: OPTIONS >> Really destroying SIP dialog >> '[email protected]:5060' Method: OPTIONS >> >> <--- SIP read from UDP:172.17.39.41:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK3a5093e6 >> From: "asterisk" <sip:[email protected]>;tag=as2f7fde70 >> To: <sip:172.17.39.41>;tag=1236514593 >> Date: Fri, 02 Nov 2018 19:14:24 GMT >> Call-ID: [email protected]:5060 >> Server: Cisco-CUCM10.5 >> CSeq: 102 OPTIONS >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> Really destroying SIP dialog >> '[email protected]:5060' Method: OPTIONS >> Really destroying SIP dialog >> '[email protected]' Method: OPTIONS >> >> <--- SIP read from UDP:172.17.39.41:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.41:5060;branch=z9hG4bK1c785b1bd77f3f >> From: <sip:172.17.39.41>;tag=1269215347 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:14:39 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.41:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Sending to 172.17.39.41:5060 (no NAT) >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.41:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.41:5060;branch=z9hG4bK1c785b1bd77f3f;received=172.17.39.41 >> From: <sip:172.17.39.41>;tag=1269215347 >> To: <sip:172.17.37.129>;tag=as1baa4254 >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.90.170:50147 ---> >> >> >> <-------------> >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.43:5060;branch=z9hG4bK2bff60347a9104 >> From: <sip:172.17.39.43>;tag=486133364 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:14:59 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.43:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Sending to 172.17.39.43:5060 (no NAT) >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.43:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.43:5060;branch=z9hG4bK2bff60347a9104;received=172.17.39.43 >> From: <sip:172.17.39.43>;tag=486133364 >> To: <sip:172.17.37.129>;tag=as33d34b95 >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cc56189b3648 >> From: <sip:172.17.39.42>;tag=279128362 >> To: <sip:172.17.37.129> >> Date: Fri, 02 Nov 2018 19:15:00 GMT >> Call-ID: [email protected] >> User-Agent: Cisco-CUCM10.5 >> CSeq: 101 OPTIONS >> Contact: <sip:172.17.39.42:5060> >> Max-Forwards: 0 >> Content-Length: 0 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Sending to 172.17.39.42:5060 (no NAT) >> Looking for s in ramais (domain 172.17.37.129) >> >> <--- Transmitting (no NAT) to 172.17.39.42:5060 ---> >> SIP/2.0 404 Not Found >> Via: SIP/2.0/UDP >> 172.17.39.42:5060;branch=z9hG4bK95cc56189b3648;received=172.17.39.42 >> From: <sip:172.17.39.42>;tag=279128362 >> To: <sip:172.17.37.129>;tag=as562795db >> Call-ID: [email protected] >> CSeq: 101 OPTIONS >> Server: Asterisk PBX 13.23.1 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Accept: application/sdp >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms (Method: >> OPTIONS) >> Really destroying SIP dialog >> '[email protected]' Method: OPTIONS >> >> <--- SIP read from UDP:172.17.90.170:50147 ---> >> >> >> <-------------> >> infoasterisk*CLI> >> infoasterisk*CLI> >> Reliably Transmitting (no NAT) to 172.17.39.42:5060: >> OPTIONS sip:172.17.39.42 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK765e1b02 >> Max-Forwards: 70 >> From: "asterisk" <sip:[email protected]>;tag=as24cc7f1d >> To: <sip:172.17.39.42> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:15:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> Reliably Transmitting (no NAT) to 172.17.39.43:5060: >> OPTIONS sip:172.17.39.43 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK18f5a08f >> Max-Forwards: 70 >> From: "asterisk" <sip:[email protected]>;tag=as324cd423 >> To: <sip:172.17.39.43> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:15:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> Reliably Transmitting (no NAT) to 172.17.39.41:5060: >> OPTIONS sip:172.17.39.41 SIP/2.0 >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK448e3148 >> Max-Forwards: 70 >> From: "asterisk" <sip:[email protected]>;tag=as50a2d78b >> To: <sip:172.17.39.41> >> Contact: <sip:[email protected]:5060> >> Call-ID: [email protected]:5060 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX 13.23.1 >> Date: Fri, 02 Nov 2018 19:15:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Content-Length: 0 >> >> >> --- >> >> <--- SIP read from UDP:172.17.39.42:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK765e1b02 >> From: "asterisk" <sip:[email protected]>;tag=as24cc7f1d >> To: <sip:172.17.39.42>;tag=1358087302 >> Date: Fri, 02 Nov 2018 19:15:24 GMT >> Call-ID: [email protected]:5060 >> Server: Cisco-CUCM10.5 >> CSeq: 102 OPTIONS >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> Really destroying SIP dialog >> '[email protected]:5060' Method: OPTIONS >> >> <--- SIP read from UDP:172.17.39.41:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK448e3148 >> From: "asterisk" <sip:[email protected]>;tag=as50a2d78b >> To: <sip:172.17.39.41>;tag=319483522 >> Date: Fri, 02 Nov 2018 19:15:24 GMT >> Call-ID: [email protected]:5060 >> Server: Cisco-CUCM10.5 >> CSeq: 102 OPTIONS >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> Really destroying SIP dialog >> '[email protected]:5060' Method: OPTIONS >> >> <--- SIP read from UDP:172.17.39.43:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK18f5a08f >> From: "asterisk" <sip:[email protected]>;tag=as324cd423 >> To: <sip:172.17.39.43>;tag=635128065 >> Date: Fri, 02 Nov 2018 19:15:24 GMT >> Call-ID: [email protected]:5060 >> Server: Cisco-CUCM10.5 >> CSeq: 102 OPTIONS >> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >> SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> Really destroying SIP dialog >> '[email protected]:5060' Method: OPTIONS >> infoasterisk*CLI> sip set debug off >> SIP Debugging Disabled >> infoasterisk*CLI> >> infoasterisk*CLI> >> infoasterisk*CLI> >> infoasterisk*CLI> >> >> >> Atenciosamente, >> Giliardy Correia Arena. >> >> >> >> >> Em sex, 2 de nov de 2018 às 13:14, Giliardy Arena <[email protected]> >> escreveu: >>> >>> Boa tarde. >>> Obrigado pela resposta, Rogerio. >>> >>> Sim , já testei como uma extensão simples e o cenário é o mesmo. >>> >>> No CLI eu só enxergo LOG quando a chamada é conectada. >>> Não consigo ver nada diferente antes desse momento. >>> >>> Via tcpdump eu vejo as tentativas, mas não consigo identificar a causa do >>> atraso através dele. >>> Me chamou atenção a tentativa do Asterisk em todos os IPs do Call Manager, >>> quando ele deveria se conectar diretamente ao que enviou a chamada. >>> >>> Você tem alguma sugestão que eu possa fazer no CLI para tentar enxergar a >>> tentativa desde o recebimento do INVITE ? >>> >>> Atenciosamente, >>> Giliardy Correia Arena. >>> >>> >>> >>> >>> Em qui, 1 de nov de 2018 às 19:24, Giliardy Arena >>> <[email protected]> escreveu: >>>> >>>> Sim ! >>>> >>>> Os ramais ficam no Cisco. Eu apenas vou ligar para um numero do Asterisk >>>> que vai gravar as ligações. >>>> Veja uma nova captura >>>> >>>> A troca de mensagens OPTION com os servidores que não possuem o ramal que >>>> eu estou chamado do Cisco que parece estar atrasando.... Mas não sei como >>>> resolver, pois já forcei apenas um servidor no sip.conf >>>> >>>> >>>> 19:23:10.984078 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> INVITE sip:[email protected]:5060 SIP/2.0 >>>> 19:23:11.496042 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> INVITE sip:[email protected]:5060 SIP/2.0 >>>> 19:23:12.507249 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> INVITE sip:[email protected]:5060 SIP/2.0 >>>> 19:23:14.513145 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> INVITE sip:[email protected]:5060 SIP/2.0 >>>> 19:23:15.983468 ARP, Request who-has asterisk.ogmaster.local tell >>>> cucmservice01, length 46 >>>> 19:23:15.983484 ARP, Reply asterisk.ogmaster.local is-at 00:50:56:90:dc:d1 >>>> (oui Unknown), length 28 >>>> 19:23:18.524150 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> INVITE sip:[email protected]:5060 SIP/2.0 >>>> 19:23:19.220165 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:19.726828 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:20.739614 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:22.706629 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:22.755062 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:23.213088 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:24.220115 ARP, Request who-has asterisk.ogmaster.local tell >>>> cucmservice02, length 46 >>>> 19:23:24.220130 ARP, Reply asterisk.ogmaster.local is-at 00:50:56:90:dc:d1 >>>> (oui Unknown), length 28 >>>> 19:23:24.224829 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:24.292071 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:24.808252 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:25.810898 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:26.240672 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:26.533679 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> INVITE sip:[email protected]:5060 SIP/2.0 >>>> 19:23:26.762741 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:27.827149 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:29.292152 ARP, Request who-has asterisk.ogmaster.local tell >>>> infocucmpub, length 46 >>>> 19:23:29.292168 ARP, Reply asterisk.ogmaster.local is-at 00:50:56:90:dc:d1 >>>> (oui Unknown), length 28 >>>> 19:23:30.247068 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:30.769748 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:31.835377 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:34.259328 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:34.784241 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:35.845668 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:38.268704 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>> 19:23:38.797238 ARP, Request who-has 172.17.39.48 tell cucmservice02, >>>> length 46 >>>> 19:23:38.989294 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 100 Trying >>>> 19:23:38.989552 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 100 Trying >>>> 19:23:38.989649 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 100 Trying >>>> 19:23:38.989743 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 100 Trying >>>> 19:23:38.989824 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 100 Trying >>>> 19:23:38.989979 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.990068 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.990155 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.990257 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.990339 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.990409 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.990505 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.990611 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.990688 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.990777 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.990878 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.990994 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 100 Trying >>>> 19:23:38.991069 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.991130 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.991218 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.991311 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.991460 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.991545 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.991636 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.991723 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:38.991807 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 404 Not Found >>>> 19:23:39.085356 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 180 Ringing >>>> 19:23:39.797232 ARP, Request who-has 172.17.39.48 tell cucmservice02, >>>> length 46 >>>> 19:23:40.768521 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> CANCEL sip:[email protected]:5060 SIP/2.0 >>>> 19:23:40.768819 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 487 Request Terminated >>>> 19:23:40.768869 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>> SIP/2.0 200 OK >>>> 19:23:40.771996 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>> ACK sip:[email protected]:5060 SIP/2.0 >>>> 19:23:40.797266 ARP, Request who-has 172.17.39.48 tell cucmservice02, >>>> length 46 >>>> >>>> Atenciosamente, >>>> Giliardy Correia Arena. >>>> >>>> >>>> >>>> >>>> Em qui, 1 de nov de 2018 às 17:30, Giliardy Arena >>>> <[email protected]> escreveu: >>>>> >>>>> Oi Luiz. >>>>> Estabeleci um SIP entre o Call Manager e o Asterisk. >>>>> O Call Manager possui um Publisher (39.41) e os Subscribers (39.42 e >>>>> 39.43), onde ficam os telefones registrados. >>>>> >>>>> Já testei tanto deixando todos os IPs possíveis do Call Manager, quanto >>>>> apenas a referente ao registro do meu telefone no Call Manager(39.42) e a >>>>> demora é a mesma. >>>>> >>>>> ;[callman01] >>>>> ;type=friend >>>>> ;context=ramais >>>>> ;host=172.17.39.41 >>>>> ;disallow=all >>>>> ;allow=ulaw >>>>> ;allow=alaw >>>>> ;nat=no >>>>> ;canreinvite=yes >>>>> ;qualify=yes >>>>> >>>>> [callman02] >>>>> type=friend >>>>> context=ramais >>>>> host=172.17.39.42 >>>>> disallow=all >>>>> allow=ulaw >>>>> allow=alaw >>>>> nat=no >>>>> canreinvite=yes >>>>> qualify=yes >>>>> >>>>> ;[callman03] >>>>> ;type=friend >>>>> ;context=ramais >>>>> ;host=172.17.39.43 >>>>> ;disallow=all >>>>> ;allow=ulaw >>>>> ;allow=alaw >>>>> ;nat=no >>>>> ;canreinvite=yes >>>>> ;qualify=yes >>>>> >>>>> >>>>> >>>>> Do lado do Call Manager está tudo configurado e eles estão falando UDP. >>>>> >>>>> >>>>> >>>>> >>>>> No lado do Asterisk , não consegui alguma captura especifica, mas peguei >>>>> via TCPDUMP que ele parece tentar todos antes de efetivamente fechar com >>>>> o primeiro , embora já tenha recebido INVITE do correto. >>>>> >>>>> >>>>> >>>>> tcpdump -i ens192 dst 172.17.37.129 and src 172.17.39.41 or 172.17.39.42 >>>>> or 172.17.39.43 >>>>> >>>>> >>>>> 16:47:31.740674 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>>> INVITE sip:[email protected]:5060 SIP/2.0 >>>>> 16:47:32.254307 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>>> INVITE sip:[email protected]:5060 SIP/2.0 >>>>> 16:47:33.258050 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>>> INVITE sip:[email protected]:5060 SIP/2.0 >>>>> 16:47:35.272582 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>>> INVITE sip:[email protected]:5060 SIP/2.0 >>>>> 16:47:38.225049 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:38.740848 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:39.282208 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>>> INVITE sip:[email protected]:5060 SIP/2.0 >>>>> 16:47:39.751717 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:41.754129 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:43.224610 ARP, Request who-has asterisk.ogmaster.local tell >>>>> infocucmpub, length 46 >>>>> 16:47:45.768670 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:46.055483 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:46.560533 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:47.292581 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>>> INVITE sip:[email protected]:5060 SIP/2.0 >>>>> 16:47:47.572900 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:49.587485 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:49.780979 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:51.054865 ARP, Request who-has asterisk.ogmaster.local tell >>>>> cucmservice02, length 46 >>>>> 16:47:52.292278 ARP, Request who-has asterisk.ogmaster.local tell >>>>> cucmservice01, length 46 >>>>> 16:47:53.596301 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:53.785687 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:57.607030 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:47:59.754553 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>>> ACK sip:[email protected]:5060 SIP/2.0 >>>>> 16:47:59.755067 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>>> ACK sip:[email protected]:5060 SIP/2.0 >>>>> 16:47:59.756284 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>>> ACK sip:[email protected]:5060 SIP/2.0 >>>>> 16:48:00.535923 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>>> OPTIONS sip:172.17.37.129:5060 SIP/2.0 >>>>> 16:48:02.126054 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: >>>>> SIP/2.0 200 OK >>>>> 16:48:02.220213 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: >>>>> SIP/2.0 200 OK >>>>> 16:48:02.220484 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: >>>>> SIP/2.0 200 OK >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> tcpdump -i ens192 src 172.17.37.129 and dst 172.17.39.41 or 172.17.39.42 >>>>> or 172.17.39.43 >>>>> >>>>> >>>>> 16:47:59.749555 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>>> SIP/2.0 100 Trying >>>>> 16:47:59.749932 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>>> SIP/2.0 100 Trying >>>>> 16:47:59.750055 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>>> SIP/2.0 100 Trying >>>>> 16:47:59.750181 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>>> SIP/2.0 100 Trying >>>>> 16:47:59.750348 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:47:59.750472 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:47:59.750514 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>>> SIP/2.0 200 OK >>>>> 16:47:59.750797 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>>> SIP/2.0 100 Trying >>>>> 16:47:59.750935 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>>> SIP/2.0 200 OK >>>>> 16:47:59.751084 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:47:59.751193 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:47:59.751293 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:47:59.751487 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:47:59.751608 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:47:59.751761 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>>> SIP/2.0 100 Trying >>>>> 16:47:59.751864 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>>> SIP/2.0 200 OK >>>>> 16:47:59.751998 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:47:59.752116 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:47:59.752230 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:47:59.752343 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:47:59.752458 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:47:59.752576 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:48:00.536313 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>>> SIP/2.0 404 Not Found >>>>> 16:48:02.124006 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: >>>>> OPTIONS sip:172.17.39.41 SIP/2.0 >>>>> 16:48:02.218575 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: >>>>> OPTIONS sip:172.17.39.43 SIP/2.0 >>>>> 16:48:02.218761 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>>> OPTIONS sip:172.17.39.42 SIP/2.0 >>>>> 16:48:02.589632 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: >>>>> SIP/2.0 200 OK >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Testei alguns Debugs que fui pesquisando na internet mas não consegui >>>>> compreender muito bem.... >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> [Oct 31 15:35:20] DEBUG[31072] chan_sip.c: = Looking for Call ID: >>>>> [email protected] (Checking From) --From tag >>>>> 1146601895 --To-tag >>>>> [Oct 31 15:35:20] DEBUG[31072] acl.c: For destination '172.17.39.42', our >>>>> source address is '172.17.37.129'. >>>>> [Oct 31 15:35:20] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with >>>>> address 172.17.37.129:5060 >>>>> [Oct 31 15:35:20] DEBUG[31072] netsock2.c: Splitting '172.17.39.42:5060' >>>>> into... >>>>> [Oct 31 15:35:20] DEBUG[31072] netsock2.c: ...host '172.17.39.42' and >>>>> port '5060'. >>>>> [Oct 31 15:35:20] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for >>>>> [email protected] - OPTIONS (No RTP) >>>>> [Oct 31 15:35:20] DEBUG[31072] chan_sip.c: **** Received OPTIONS (3) - >>>>> Command in SIP OPTIONS >>>>> [Oct 31 15:35:20] DEBUG[31072] netsock2.c: Splitting '172.17.37.129:5060' >>>>> into... >>>>> [Oct 31 15:35:20] DEBUG[31072] netsock2.c: ...host '172.17.37.129' and >>>>> port ''. >>>>> [Oct 31 15:35:20] DEBUG[31072] netsock2.c: Splitting '172.17.39.42' >>>>> into... >>>>> [Oct 31 15:35:20] DEBUG[31072] netsock2.c: ...host '172.17.39.42' and >>>>> port ''. >>>>> [Oct 31 15:35:20] DEBUG[31072] chan_sip.c: Trying to put 'SIP/2.0 404' >>>>> onto UDP socket destined for 172.17.39.42:5060 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for >>>>> 7eeb423d62baf89b2376864b55f025a9@[fe80::a0e0:69c4:bc8b:b417]:5060 - >>>>> OPTIONS (No RTP) >>>>> [Oct 31 15:35:21] DEBUG[31072] acl.c: For destination '172.17.39.43', our >>>>> source address is '172.17.37.129'. >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with >>>>> address 172.17.37.129:5060 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: SIP call-id changed from >>>>> '7eeb423d62baf89b2376864b55f025a9@[fe80::a0e0:69c4:bc8b:b417]:5060' to >>>>> '[email protected]:5060' >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Initializing initreq for >>>>> method OPTIONS - callid >>>>> [email protected]:5060 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Trying to put 'OPTIONS sip' >>>>> onto UDP socket destined for 172.17.39.43:5060 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for >>>>> 2b73bb0d2c3469fa0780743f3270ca4f@[fe80::a0e0:69c4:bc8b:b417]:5060 - >>>>> OPTIONS (No RTP) >>>>> [Oct 31 15:35:21] DEBUG[31072] acl.c: For destination '172.17.39.42', our >>>>> source address is '172.17.37.129'. >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with >>>>> address 172.17.37.129:5060 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: SIP call-id changed from >>>>> '2b73bb0d2c3469fa0780743f3270ca4f@[fe80::a0e0:69c4:bc8b:b417]:5060' to >>>>> '[email protected]:5060' >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Initializing initreq for >>>>> method OPTIONS - callid >>>>> [email protected]:5060 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Trying to put 'OPTIONS sip' >>>>> onto UDP socket destined for 172.17.39.42:5060 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for >>>>> 3c48a6e96480adda0d8af61a4d498fb7@[fe80::a0e0:69c4:bc8b:b417]:5060 - >>>>> OPTIONS (No RTP) >>>>> [Oct 31 15:35:21] DEBUG[31072] acl.c: For destination '172.17.39.41', our >>>>> source address is '172.17.37.129'. >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with >>>>> address 172.17.37.129:5060 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: SIP call-id changed from >>>>> '3c48a6e96480adda0d8af61a4d498fb7@[fe80::a0e0:69c4:bc8b:b417]:5060' to >>>>> '[email protected]:5060' >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Initializing initreq for >>>>> method OPTIONS - callid >>>>> [email protected]:5060 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Trying to put 'OPTIONS sip' >>>>> onto UDP socket destined for 172.17.39.41:5060 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: = Looking for Call ID: >>>>> [email protected]:5060 (Checking To) --From >>>>> tag as2ee346e2 --To-tag 348178859 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Stopping retransmission on >>>>> '[email protected]:5060' of Request 102: >>>>> Match Found >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: = Looking for Call ID: >>>>> [email protected]:5060 (Checking To) --From >>>>> tag as138ca155 --To-tag 802041871 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Stopping retransmission on >>>>> '[email protected]:5060' of Request 102: >>>>> Match Found >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Destroying SIP dialog >>>>> [email protected]:5060 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Destroying SIP dialog >>>>> [email protected]:5060 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: = Looking for Call ID: >>>>> [email protected]:5060 (Checking To) --From >>>>> tag as34b82738 --To-tag 605276003 >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Stopping retransmission on >>>>> '[email protected]:5060' of Request 102: >>>>> Match Found >>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Destroying SIP dialog >>>>> [email protected]:5060 >>>>> [Oct 31 15:35:36] DEBUG[31072] chan_sip.c: Auto destroying SIP dialog >>>>> '[email protected]' >>>>> [Oct 31 15:35:36] DEBUG[31072] chan_sip.c: Destroying SIP dialog >>>>> [email protected] >>>>> [Oct 31 15:35:43] DEBUG[31072] chan_sip.c: Auto destroying SIP dialog >>>>> '[email protected]' >>>>> [Oct 31 15:35:43] DEBUG[31072] chan_sip.c: Destroying SIP dialog >>>>> [email protected] >>>>> [Oct 31 15:35:52] DEBUG[31072] chan_sip.c: Auto destroying SIP dialog >>>>> '[email protected]' >>>>> [Oct 31 15:35:52] DEBUG[31072] chan_sip.c: Destroying SIP dialog >>>>> [email protected] >>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: = Looking for Call ID: >>>>> [email protected] (Checking From) --From tag >>>>> 1522038610 --To-tag >>>>> [Oct 31 15:36:04] DEBUG[31072] acl.c: For destination '172.17.39.43', our >>>>> source address is '172.17.37.129'. >>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with >>>>> address 172.17.37.129:5060 >>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with >>>>> address 172.17.37.129:5060 >>>>> [Oct 31 15:36:04] DEBUG[31072] netsock2.c: Splitting '172.17.39.43:5060' >>>>> into... >>>>> [Oct 31 15:36:04] DEBUG[31072] netsock2.c: ...host '172.17.39.43' and >>>>> port '5060'. >>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for >>>>> [email protected] - OPTIONS (No RTP) >>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: **** Received OPTIONS (3) - >>>>> Command in SIP OPTIONS >>>>> [Oct 31 15:36:04] DEBUG[31072] netsock2.c: Splitting '172.17.37.129:5060' >>>>> into... >>>>> [Oct 31 15:36:04] DEBUG[31072] netsock2.c: ...host '172.17.37.129' and >>>>> port ''. >>>>> [Oct 31 15:36:04] DEBUG[31072] netsock2.c: Splitting '172.17.39.43' >>>>> into... >>>>> [Oct 31 15:36:04] DEBUG[31072] netsock2.c: ...host '172.17.39.43' and >>>>> port ''. >>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: Trying to put 'SIP/2.0 404' >>>>> onto UDP socket destined for 172.17.39.43:5060 >>>>> [Oct 31 15:36:12] DEBUG[31072] chan_sip.c: = Looking for Call ID: >>>>> [email protected] (Checking From) --From tag >>>>> 639004019 --To-tag >>>>> [Oct 31 15:36:12] DEBUG[31072] acl.c: For destination '172.17.39.41', our >>>>> source address is '172.17.37.129'. >>>>> [Oct 31 15:36:12] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with >>>>> address 172.17.37.129:5060 >>>>> [Oct 31 15:36:12] DEBUG[31072] netsock2.c: Splitting '172.17.39.41:5060' >>>>> into... >>>>> [Oct 31 15:36:12] DEBUG[31072] netsock2.c: ...host '172.17.39.41' and >>>>> port '5060'. >>>>> [Oct 31 15:36:12] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for >>>>> [email protected] - OPTIONS (No RTP) >>>>> [Oct 31 15:36:12] DEBUG[31072] chan_sip.c: **** Received OPTIONS (3) - >>>>> Command in SIP OPTIONS >>>>> [Oct 31 15:36:12] DEBUG[31072] netsock2.c: Splitting '172.17.37.129:5060' >>>>> into... >>>>> [Oct 31 15:36:12] DEBUG[31072] netsock2.c: ...host '172.17.37.129' and >>>>> port ''. >>>>> [Oct 31 15:36:12] DEBUG[31072] netsock2.c: Splitting '172.17.39.41' >>>>> into... >>>>> [Oct 31 15:36:12] DEBUG[31072] netsock2.c: ...host '172.17.39.41' and >>>>> port ''. >>>>> [Oct 31 15:36:12] DEBUG[31072] chan_sip.c: Trying to put 'SIP/2.0 404' >>>>> onto UDP socket destined for 172.17.39.41:5060 >>>>> >>>>> >>>>> >>>>> >>>>> Atenciosamente, >>>>> Giliardy Correia Arena. >>>>> >>>>> >>>>> >>>>> >>>>> Em qui, 1 de nov de 2018 às 15:05, Giliardy Arena >>>>> <[email protected]> escreveu: >>>>>> >>>>>> Olá pessoal ! >>>>>> Alguma ajuda ? Alguma dica ? >>>>>> >>>>>> Obrigado >>>>>> >>>>>> >>>>>> Atenciosamente, >>>>>> Giliardy Correia Arena. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Em qua, 31 de out de 2018 às 10:58, Giliardy Arena >>>>>> <[email protected]> escreveu: >>>>>>> >>>>>>> Olá , bom dia. >>>>>>> >>>>>>> Alguém sugere alguma forma de eu rastrear a ligação desde a chegada da >>>>>>> requisicao SIP no servidor Asterisk , para entender o motivo de demorar >>>>>>> muito para conectar? Algum debug específico, um trace , um log... >>>>>>> >>>>>>> Obrigado >>>>>>> >>>>>>> Em ter, 30 de out de 2018 20:22, Giliardy Arena >>>>>>> <[email protected]> escreveu: >>>>>>>> >>>>>>>> Sylvio >>>>>>>> >>>>>>>> O waitforsilence é para identificar se não tiver mais conversação e >>>>>>>> encerrar a ligação. >>>>>>>> Para evitar ficar alguma chamada presa gravando eternamente. >>>>>>>> >>>>>>>> >>>>>>>> Atenciosamente, >>>>>>>> Giliardy Correia Arena. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Em ter, 30 de out de 2018 às 17:57, Giliardy Arena >>>>>>>> <[email protected]> escreveu: >>>>>>>>> >>>>>>>>> Caros, >>>>>>>>> Boa tarde. >>>>>>>>> >>>>>>>>> Estou aprendendo e estudando sobre o Asterisk. >>>>>>>>> Atualmente administro um Cisco Call Manager e a minha ideia é usar o >>>>>>>>> Asterisk para gravar ligações recebidas do Call Manager. >>>>>>>>> >>>>>>>>> Fiz a integração do Asterisk com o Call Manager com sucesso. >>>>>>>>> >>>>>>>>> Estou com problema para entender o motivo do Asterisk demorar para >>>>>>>>> conectar a ligação a uma extensão. Tenho pesquisado, mas com >>>>>>>>> dificuldades para entender como debugar. >>>>>>>>> >>>>>>>>> Criei a seguinte extensão, que atende sozinha e grava. >>>>>>>>> >>>>>>>>> exten => 2005,1,Answer() >>>>>>>>> exten => >>>>>>>>> 2005,n,MixMonitor(Ramal-${CALLERID(num)}-Em-${STRFTIME(${EPOCH},,%d-%m-%Y-%H-%M)}.wav) >>>>>>>>> exten => 2005,n,WaitForSilence(10000|6) >>>>>>>>> exten => 2005,n,Hangup >>>>>>>>> >>>>>>>>> >>>>>>>>> Também experimentei o mesmo sintoma através de uma extensão que criei >>>>>>>>> e loguei numa softphone. >>>>>>>>> >>>>>>>>> - Ativei Debug full , mas não tem nenhuma mensagem importante. Apenas >>>>>>>>> o que vejo na CLI do asterisk >>>>>>>>> >>>>>>>>> - Na CLI do Asterisk só vejo log quando a chamada efetivamente é >>>>>>>>> conectada, não sei se consigo ver desde o momento que ele recebe a >>>>>>>>> requisição. >>>>>>>>> >>>>>>>>> - Fiz um TCPDUMP e realmente me parece que é o Asterisk demorando a >>>>>>>>> conectar a extensão, mas via TCPDUMP não tenho detalhes para entender >>>>>>>>> e ajustar. Demora aproximadamente 30segundos após chamar do Call >>>>>>>>> Manager. >>>>>>>>> >>>>>>>>> >>>>>>>>> Alguém pode me dar um help de por onde eu posso rastrear para tentar >>>>>>>>> corrigir ? >>>>>>>>> >>>>>>>>> Obrigado! >>>>>>>>> >>>>>>>>> Atenciosamente, >>>>>>>>> Giliardy Correia Arena. >>>>>>>>> >>>>>>>>> > _______________________________________________ > KHOMP: completa linha de placas externas FXO, FXS, GSM e E1 > Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7 > Intercomunicador e acesso remoto via rede IP e telefones IP > Conheça todo o portfólio em www.Khomp.com > _______________________________________________ > Para remover seu email desta lista, basta enviar um email em branco para > [email protected] _______________________________________________ KHOMP: completa linha de placas externas FXO, FXS, GSM e E1 Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7 Intercomunicador e acesso remoto via rede IP e telefones IP Conheça todo o portfólio em www.Khomp.com _______________________________________________ Para remover seu email desta lista, basta enviar um email em branco para [email protected]
