Show! Abracos,
Marcelo H. Terres <[email protected]> IM: [email protected] https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On Tue, 6 Nov 2018 at 22:55, Giliardy Arena <[email protected]> wrote: > > Pessoal, > Consegui ! Graças a ajuda de todos. > O problema realmente era DNS. > > > Tentei novamente há pouco as capturas e me chamou atenção um TIMEOUT > > [Nov 6 19:57:01] DEBUG[31072] acl.c: For destination '172.17.39.42', our > source address is '172.17.37.129'. > [Nov 6 19:57:01] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with > address 172.17.37.129:5060 > [Nov 6 19:57:01] DEBUG[31072] netsock2.c: Splitting '172.17.39.42:5060' > into... > [Nov 6 19:57:01] DEBUG[31072] netsock2.c: ...host '172.17.39.42' and port > '5060'. > [Nov 6 19:57:01] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for > [email protected] - INVITE (No RTP) > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] chan_sip.c: **** Received INVITE > (5) - Command in SIP INVITE > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] sip/reqresp_parser.c: Begin: > parsing SIP "Supported: timer,resource-priority,replaces" > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] sip/reqresp_parser.c: Found SIP > option: -timer- > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] sip/reqresp_parser.c: Matched SIP > option: timer > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] sip/reqresp_parser.c: Found SIP > option: -resource-priority- > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] sip/reqresp_parser.c: Matched SIP > option: resource-priority > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] sip/reqresp_parser.c: Found SIP > option: -replaces- > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] sip/reqresp_parser.c: Matched SIP > option: replaces > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] sip/reqresp_parser.c: Begin: > parsing SIP "Supported: X-cisco-srtp-fallback,X-cisco-original-called" > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] sip/reqresp_parser.c: Found SIP > option: -X-cisco-srtp-fallback- > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] sip/reqresp_parser.c: Found > private SIP option, not supported: X-cisco-srtp-fallback > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] sip/reqresp_parser.c: Found SIP > option: -X-cisco-original-called- > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] sip/reqresp_parser.c: Found > private SIP option, not supported: X-cisco-original-called > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] netsock2.c: Splitting > '172.17.39.42:5060' into... > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] netsock2.c: ...host '172.17.39.42' > and port '5060'. > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] netsock2.c: Splitting > '172.17.39.42' into... > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] netsock2.c: ...host '172.17.39.42' > and port ''. > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] rtp_engine.c: Using engine > 'asterisk' for RTP instance '0x7f9c8402ac60' > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] res_rtp_asterisk.c: Allocated port > 10480 for RTP instance '0x7f9c8402ac60' > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] rtp_engine.c: RTP instance > '0x7f9c8402ac60' is setup and ready to go > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] netsock2.c: Splitting > 'infoasterisk' into... > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] netsock2.c: ...host 'infoasterisk' > and port ''. > > > Um pouco depois recebi os logs abaixo de Timeout que até então não tinha > reparado.... > E isso me chamou atenção > > [Nov 6 19:57:13] DEBUG[19060] threadpool.c: Worker thread idle timeout > reached. Dying. > [Nov 6 19:57:13] DEBUG[31032] threadpool.c: Destroying worker thread 1727 > [Nov 6 19:57:13] DEBUG[19062] threadpool.c: Worker thread idle timeout > reached. Dying. > [Nov 6 19:57:13] DEBUG[19061] threadpool.c: Worker thread idle timeout > reached. Dying. > [Nov 6 19:57:13] DEBUG[31032] threadpool.c: Destroying worker thread 1729 > [Nov 6 19:57:13] DEBUG[31032] threadpool.c: Destroying worker thread 1728 > > > E depois seguiu a demora e a ligação completou : > > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] acl.c: Multiple addresses. Using > the first only > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] res_rtp_asterisk.c: Setup RTCP on > RTP instance '0x7f9c8402ac60' > [Nov 6 19:57:29] VERBOSE[31072][C-0000009a] netsock2.c: Using SIP RTP CoS > mark 5 > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] chan_sip.c: Setting NAT on RTP to > Off > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] chan_sip.c: Processing > session-level SDP v=0... UNSUPPORTED OR FAILED. > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] chan_sip.c: Processing > session-level SDP o=CiscoSystemsCCM-SIP 100133925 1 IN IP4 172.17.39.42... OK. > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] chan_sip.c: Processing > session-level SDP s=SIP Call... UNSUPPORTED OR FAILED. > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] netsock2.c: Splitting > '172.17.231.249' into... > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] netsock2.c: ...host > '172.17.231.249' and port ''. > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] chan_sip.c: Processing > session-level SDP c=IN IP4 172.17.231.249... OK. > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] chan_sip.c: Processing > session-level SDP t=0 0... UNSUPPORTED OR FAILED. > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] rtp_engine.c: Setting tx payload > type 0 based on m type on 0x7f9c38d3a390 > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] rtp_engine.c: Setting tx payload > type 101 based on m type on 0x7f9c38d3a390 > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] chan_sip.c: Processing media-level > (audio) SDP a=rtpmap:0 PCMU/8000... OK. > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] chan_sip.c: Processing media-level > (audio) SDP a=rtpmap:101 telephone-event/8000... OK. > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] chan_sip.c: Processing media-level > (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. > [Nov 6 19:57:29] DEBUG[31072][C-0000009a] acl.c: For destination > '172.17.231.249', our source address is '172.17.37.129'. > > > > > Eu tinha no meu DNS o nome "Asterisk" cadastrado. > > > [Nov 6 19:57:01] DEBUG[31072][C-0000009a] netsock2.c: ...host 'infoasterisk' > and port '' > > > > Por algum motivo , o Asterisk utiliza o nome do host. > Para tirar a duvida , cadastrei também o nome "infoasterisk" no DNS e > funcionou de primeira. > A ligação conecta automaticamente. > > > Gostaria de agradecer a todos por todas as dicas. > > > > Atenciosamente, > Giliardy Correia Arena. > > > > > Em seg, 5 de nov de 2018 às 17:16, Giliardy Arena <[email protected]> > escreveu: >> >> Infelizmente ainda não. >> Eu vejo bater , e depois só loga mensagens quando chama no ramal. >> Então não vejo no meio tempo o que o Asterisk está tentando fazer. >> Se tiverem alguma dica de debug especifico.. >> >> Já tentei sip debug, sip debug peer, já mudei os core verbose e debug .... >> >> Vejam se conseguem visualizar o post que abri na comunidade do asterisk. >> Lá compartilhei as imagens com as explicações. >> https://community.asterisk.org/t/asterisk-register-on-invite/74776/5 >> >> >> >> https://imgur.com/a/yW9tM89 >> https://imgur.com/a/9wkdO2B >> >> https://imgur.com/a/Cq9opqc >> https://imgur.com/a/ukNAZx5 >> https://imgur.com/a/ukNAZx5 >> Atenciosamente, >> Giliardy Correia Arena. >> >> >> >> >> Em seg, 5 de nov de 2018 às 15:25, Giliardy Arena <[email protected]> >> escreveu: >>> >>> Oi ! >>> Não consegui ainda. Mas aparentemente não é problema de DNS, pelo tcpdump >>> que tenho. >>> Preciso entender o que se passa no Asterisk após receber o INVITE, que >>> ainda não consegui visualizar. >>> >>> Como faço para enviar imagens no fórum. É possível? >>> Ou devo hospedar num site qualquer e enviar o link ? >>> >>> Fica mais facil para entenderem. >>> >>> >>> Atenciosamente, >>> Giliardy Correia Arena. >>> >>> >>> >>> >>> Em sex, 2 de nov de 2018 às 20:50, Giliardy Arena >>> <[email protected]> escreveu: >>>> >>>> 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-CU > > _______________________________________________ > 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]
