[AsteriskBrasil] Demora para Completar ligação

Quarta Novembro 7 13:36:02 -02 2018

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Abracos,

Marcelo H. Terres <mhterres at gmail.com>
IM: mhterres at jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

On Tue, 6 Nov 2018 at 22:55, Giliardy Arena <giliardy.arena at gmail.com> 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 df5bc980-be210e28-400108-2a2711ac at 172.17.39.42 - 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 <giliardy.arena at gmail.com> 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 <giliardy.arena at gmail.com> 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 <giliardy.arena at gmail.com> 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 <giliardy.arena at gmail.com> 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:2001 at 172.17.37.129:5060 SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Date: Fri, 02 Nov 2018 19:11:47 GMT
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:9770 at 172.17.39.42>
>>>>> Remote-Party-ID: "Giliardy Arena" <sip:9770 at 172.17.39.42>;party=calling;screen=yes;privacy=off
>>>>> Contact: <sip:9770 at 172.17.39.42:5060>;+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 - 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:9770 at 172.17.39.42:5060>
>>>>>
>>>>>
>>>>>
>>>>> 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:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:2001 at 172.17.37.129:5060>
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> <------------>
>>>>>
>>>>> <--- SIP read from UDP:172.17.39.42:5060 --->
>>>>> INVITE sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Date: Fri, 02 Nov 2018 19:11:48 GMT
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:9770 at 172.17.39.42>
>>>>> Remote-Party-ID: "Giliardy Arena" <sip:9770 at 172.17.39.42>;party=calling;screen=yes;privacy=off
>>>>> Contact: <sip:9770 at 172.17.39.42:5060>;+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:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:2001 at 172.17.37.129:5060>
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> <------------>
>>>>>
>>>>> <--- SIP read from UDP:172.17.39.42:5060 --->
>>>>> INVITE sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Date: Fri, 02 Nov 2018 19:11:49 GMT
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:9770 at 172.17.39.42>
>>>>> Remote-Party-ID: "Giliardy Arena" <sip:9770 at 172.17.39.42>;party=calling;screen=yes;privacy=off
>>>>> Contact: <sip:9770 at 172.17.39.42:5060>;+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:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:2001 at 172.17.37.129:5060>
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> <------------>
>>>>>
>>>>> <--- SIP read from UDP:172.17.39.42:5060 --->
>>>>> INVITE sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Date: Fri, 02 Nov 2018 19:11:51 GMT
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:9770 at 172.17.39.42>
>>>>> Remote-Party-ID: "Giliardy Arena" <sip:9770 at 172.17.39.42>;party=calling;screen=yes;privacy=off
>>>>> Contact: <sip:9770 at 172.17.39.42:5060>;+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:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:2001 at 172.17.37.129:5060>
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> <------------>
>>>>>
>>>>> <--- SIP read from UDP:172.17.90.170:50147 --->
>>>>>
>>>>>
>>>>> <------------->
>>>>>
>>>>> <--- SIP read from UDP:172.17.39.42:5060 --->
>>>>> INVITE sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Date: Fri, 02 Nov 2018 19:11:55 GMT
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:9770 at 172.17.39.42>
>>>>> Remote-Party-ID: "Giliardy Arena" <sip:9770 at 172.17.39.42>;party=calling;screen=yes;privacy=off
>>>>> Contact: <sip:9770 at 172.17.39.42:5060>;+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:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:2001 at 172.17.37.129:5060>
>>>>> 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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 '28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43' 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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 '28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43' 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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 '29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42' 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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 '28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43' 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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 '29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42' 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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 '29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42' 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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 '28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43' 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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 '29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42' in 32000 ms (Method: OPTIONS)
>>>>>
>>>>> <--- SIP read from UDP:172.17.39.42:5060 --->
>>>>> INVITE sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Date: Fri, 02 Nov 2018 19:12:03 GMT
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:9770 at 172.17.39.42>
>>>>> Remote-Party-ID: "Giliardy Arena" <sip:9770 at 172.17.39.42>;party=calling;screen=yes;privacy=off
>>>>> Contact: <sip:9770 at 172.17.39.42:5060>;+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:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:2001 at 172.17.37.129:5060>
>>>>> 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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 '28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43' 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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 '29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42' 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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 '28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43' 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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 '29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42' 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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: 28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43
>>>>> 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 '28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43' 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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: 29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42
>>>>> 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 '29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42' in 32000 ms (Method: OPTIONS)
>>>>>     -- Executing [2001 at 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:2001 at 172.17.90.170:50147;rinstance=a175c2caa1292efd SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK7e5232b4
>>>>> Max-Forwards: 70
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.37.129>;tag=as1b69e3fc
>>>>> To: <sip:2001 at 172.17.90.170:50147;rinstance=a175c2caa1292efd>
>>>>> Contact: <sip:9770 at 172.17.37.129:5060>
>>>>> Call-ID: 50915352165f67003a4ada5854da2a90 at 172.17.37.129:5060
>>>>> 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:2001 at 172.17.90.170:50147;rinstance=a175c2caa1292efd>
>>>>> To: "2001"<sip:2001 at 172.17.90.170:50147;rinstance=a175c2caa1292efd>;tag=a0a27e40
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.37.129>;tag=as1b69e3fc
>>>>> Call-ID: 50915352165f67003a4ada5854da2a90 at 172.17.37.129:5060
>>>>> 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:2001 at 172.17.90.170:50147;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:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>;tag=as109d5c95
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:2001 at 172.17.37.129:5060>
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> <------------>
>>>>>
>>>>> <--- SIP read from UDP:172.17.39.42:5060 --->
>>>>> CANCEL sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>
>>>>> Date: Fri, 02 Nov 2018 19:12:03 GMT
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>;tag=as109d5c95
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>;tag=as109d5c95
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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:2001 at 172.17.90.170:50147;rinstance=a175c2caa1292efd SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK7e5232b4
>>>>> Max-Forwards: 70
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.37.129>;tag=as1b69e3fc
>>>>> To: <sip:2001 at 172.17.90.170:50147;rinstance=a175c2caa1292efd>
>>>>> Call-ID: 50915352165f67003a4ada5854da2a90 at 172.17.37.129:5060
>>>>> 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:2001 at 172.17.37.129:5060 SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.17.39.42:5060;branch=z9hG4bK95cbf93f9f4ba
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.39.42>;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
>>>>> To: <sip:2001 at 172.17.37.129>;tag=as109d5c95
>>>>> Date: Fri, 02 Nov 2018 19:12:03 GMT
>>>>> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42
>>>>> 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 '237b6100-bdc1a173-3cf01d-2a2711ac at 172.17.39.42' 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:2001 at 172.17.90.170:50147;rinstance=a175c2caa1292efd>
>>>>> To: <sip:2001 at 172.17.90.170:50147;rinstance=a175c2caa1292efd>;tag=a0a27e40
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.37.129>;tag=as1b69e3fc
>>>>> Call-ID: 50915352165f67003a4ada5854da2a90 at 172.17.37.129:5060
>>>>> 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:2001 at 172.17.90.170:50147;rinstance=a175c2caa1292efd>;tag=a0a27e40
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.37.129>;tag=as1b69e3fc
>>>>> Call-ID: 50915352165f67003a4ada5854da2a90 at 172.17.37.129:5060
>>>>> 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:2001 at 172.17.90.170:50147;rinstance=a175c2caa1292efd SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK7e5232b4
>>>>> Max-Forwards: 70
>>>>> From: "Giliardy Arena" <sip:9770 at 172.17.37.129>;tag=as1b69e3fc
>>>>> To: <sip:2001 at 172.17.90.170:50147;rinstance=a175c2caa1292efd>;tag=a0a27e40
>>>>> Contact: <sip:9770 at 172.17.37.129:5060>
>>>>> Call-ID: 50915352165f67003a4ada5854da2a90 at 172.17.37.129:5060
>>>>> 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:asterisk at 172.17.37.129>;tag=as1a8e4d0e
>>>>> To: <sip:172.17.39.41>
>>>>> Contact: <sip:asterisk at 172.17.37.129:5060>
>>>>> Call-ID: 3a7729aa25ef52bc43a1e90a591f08f5 at 172.17.37.129:5060
>>>>> 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:asterisk at 172.17.37.129>;tag=as2d9ec9dd
>>>>> To: <sip:172.17.39.42>
>>>>> Contact: <sip:asterisk at 172.17.37.129:5060>
>>>>> Call-ID: 1147063b71e9d1762b714dfc40cd0c82 at 172.17.37.129:5060
>>>>> 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:asterisk at 172.17.37.129>;tag=as2deca9a9
>>>>> To: <sip:172.17.39.43>
>>>>> Contact: <sip:asterisk at 172.17.37.129:5060>
>>>>> Call-ID: 1b0f7dba03cb6de82484db42174bfa17 at 172.17.37.129:5060
>>>>> 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:asterisk at 172.17.37.129>;tag=as2d9ec9dd
>>>>> To: <sip:172.17.39.42>;tag=2130805835
>>>>> Date: Fri, 02 Nov 2018 19:12:24 GMT
>>>>> Call-ID: 1147063b71e9d1762b714dfc40cd0c82 at 172.17.37.129:5060
>>>>> 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 '1147063b71e9d1762b714dfc40cd0c82 at 172.17.37.129:5060' 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:asterisk at 172.17.37.129>;tag=as1a8e4d0e
>>>>> To: <sip:172.17.39.41>;tag=1670426499
>>>>> Date: Fri, 02 Nov 2018 19:12:24 GMT
>>>>> Call-ID: 3a7729aa25ef52bc43a1e90a591f08f5 at 172.17.37.129:5060
>>>>> 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 '3a7729aa25ef52bc43a1e90a591f08f5 at 172.17.37.129:5060' 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:asterisk at 172.17.37.129>;tag=as2deca9a9
>>>>> To: <sip:172.17.39.43>;tag=876720778
>>>>> Date: Fri, 02 Nov 2018 19:12:24 GMT
>>>>> Call-ID: 1b0f7dba03cb6de82484db42174bfa17 at 172.17.37.129:5060
>>>>> 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 '1b0f7dba03cb6de82484db42174bfa17 at 172.17.37.129:5060' 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: 41e15c80-bdc1a1a6-1c2eb1-292711ac at 172.17.39.41
>>>>> 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: 41e15c80-bdc1a1a6-1c2eb1-292711ac at 172.17.39.41
>>>>> 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 '41e15c80-bdc1a1a6-1c2eb1-292711ac at 172.17.39.41' in 32000 ms (Method: OPTIONS)
>>>>> Really destroying SIP dialog '28d8ab80-bdc1a17c-208720-2b2711ac at 172.17.39.43' Method: OPTIONS
>>>>> Really destroying SIP dialog '29714200-bdc1a17d-3cf01e-2a2711ac at 172.17.39.42' Method: OPTIONS
>>>>> Really destroying SIP dialog '50915352165f67003a4ada5854da2a90 at 172.17.37.129:5060' 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: 4d348800-bdc1a1b9-208733-2b2711ac at 172.17.39.43
>>>>> 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: 4d348800-bdc1a1b9-208733-2b2711ac at 172.17.39.43
>>>>> 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 '4d348800-bdc1a1b9-208733-2b2711ac at 172.17.39.43' 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: 4d348800-bdc1a1b9-3cf038-2a2711ac at 172.17.39.42
>>>>> 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: 4d348800-bdc1a1b9-3cf038-2a2711ac at 172.17.39.42
>>>>> 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 '4d348800-bdc1a1b9-3cf038-2a2711ac at 172.17.39.42' in 32000 ms (Method: OPTIONS)
>>>>> Really destroying SIP dialog '41e15c80-bdc1a1a6-1c2eb1-292711ac at 172.17.39.41' 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:asterisk at 172.17.37.129>;tag=as5db9427e
>>>>> To: <sip:172.17.39.42>
>>>>> Contact: <sip:asterisk at 172.17.37.129:5060>
>>>>> Call-ID: 3157a8b527eff76f6747c88c1b6b1125 at 172.17.37.129:5060
>>>>> 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:asterisk at 172.17.37.129>;tag=as3dded9ad
>>>>> To: <sip:172.17.39.41>
>>>>> Contact: <sip:asterisk at 172.17.37.129:5060>
>>>>> Call-ID: 452981b8491a141f7b9c74da6e6d991c at 172.17.37.129:5060
>>>>> 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:asterisk at 172.17.37.129>;tag=as773015ab
>>>>> To: <sip:172.17.39.43>
>>>>> Contact: <sip:asterisk at 172.17.37.129:5060>
>>>>> Call-ID: 37532137052ad7ea72c034fa1d87a29d at 172.17.37.129:5060
>>>>> 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:asterisk at 172.17.37.129>;tag=as5db9427e
>>>>> To: <sip:172.17.39.42>;tag=304370098
>>>>> Date: Fri, 02 Nov 2018 19:13:24 GMT
>>>>> Call-ID: 3157a8b527eff76f6747c88c1b6b1125 at 172.17.37.129:5060
>>>>> 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 '3157a8b527eff76f6747c88c1b6b1125 at 172.17.37.129:5060' 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:asterisk at 172.17.37.129>;tag=as3dded9ad
>>>>> To: <sip:172.17.39.41>;tag=383686183
>>>>> Date: Fri, 02 Nov 2018 19:13:24 GMT
>>>>> Call-ID: 452981b8491a141f7b9c74da6e6d991c at 172.17.37.129:5060
>>>>> 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:asterisk at 172.17.37.129>;tag=as773015ab
>>>>> To: <sip:172.17.39.43>;tag=715549747
>>>>> Date: Fri, 02 Nov 2018 19:13:24 GMT
>>>>> Call-ID: 37532137052ad7ea72c034fa1d87a29d at 172.17.37.129:5060
>>>>> 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 '452981b8491a141f7b9c74da6e6d991c at 172.17.37.129:5060' Method: OPTIONS
>>>>> Really destroying SIP dialog '37532137052ad7ea72c034fa1d87a29d at 172.17.37.129:5060' Method: OPTIONS
>>>>> Really destroying SIP dialog '4d348800-bdc1a1b9-208733-2b2711ac at 172.17.39.43' Method: OPTIONS
>>>>> Really destroying SIP dialog '4d348800-bdc1a1b9-3cf038-2a2711ac at 172.17.39.42' 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: 65a4a280-bdc1a1e2-1c2ec2-292711ac at 172.17.39.41
>>>>> 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: 65a4a280-bdc1a1e2-1c2ec2-292711ac at 172.17.39.41
>>>>> 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 '65a4a280-bdc1a1e2-1c2ec2-292711ac at 172.17.39.41' 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: 7228fb00-bdc1a1f7-3cf04c-2a2711ac at 172.17.39.42
>>>>> 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: 7228fb00-bdc1a1f7-3cf04c-2a2711ac at 172.17.39.42
>>>>> 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 '7228fb00-bdc1a1f7-3cf04c-2a2711ac at 172.17.39.42' in 32000 ms (Method: OPTIONS)
>>>>> Really destroying SIP dialog '65a4a280-bdc1a1e2-1c2ec2-292711ac at 172.17.39.41' 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:asterisk at 172.17.37.129>;tag=as753534e0
>>>>> To: <sip:172.17.39.42>
>>>>> Contact: <sip:asterisk at 172.17.37.129:5060>
>>>>> Call-ID: 3a24108b5fbea78c3e231c8a01761c4e at 172.17.37.129:5060
>>>>> 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:asterisk at 172.17.37.129>;tag=as2f7fde70
>>>>> To: <sip:172.17.39.41>
>>>>> Contact: <sip:asterisk at 172.17.37.129:5060>
>>>>> Call-ID: 06d4d73f1bd1c3a529d9c5337d3ee935 at 172.17.37.129:5060
>>>>> 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:asterisk at 172.17.37.129>;tag=as2db68d44
>>>>> To: <sip:172.17.39.43>
>>>>> Contact: <sip:asterisk at 172.17.37.129:5060>
>>>>> Call-ID: 2a43e4d20faf2382671b73ec19170e4c at 172.17.37.129:5060
>>>>> 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:asterisk at 172.17.37.129>;tag=as753534e0
>>>>> To: <sip:172.17.39.42>;tag=917613056
>>>>> Date: Fri, 02 Nov 2018 19:14:24 GMT
>>>>> Call-ID: 3a24108b5fbea78c3e231c8a01761c4e at 172.17.37.129:5060
>>>>> 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:asterisk at 172.17.37.129>;tag=as2db68d44
>>>>> To: <sip:172.17.39.43>;tag=1666345757
>>>>> Date: Fri, 02 Nov 2018 19:14:24 GMT
>>>>> Call-ID: 2a43e4d20faf2382671b73ec19170e4c at 172.17.37.129:5060
>>>>> 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 '3a24108b5fbea78c3e231c8a01761c4e at 172.17.37.129:5060' Method: OPTIONS
>>>>> Really destroying SIP dialog '2a43e4d20faf2382671b73ec19170e4c at 172.17.37.129:5060' 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:asterisk at 172.17.37.129>;tag=as2f7fde70
>>>>> To: <sip:172.17.39.41>;tag=1236514593
>>>>> Date: Fri, 02 Nov 2018 19:14:24 GMT
>>>>> Call-ID: 06d4d73f1bd1c3a529d9c5337d3ee935 at 172.17.37.129:5060
>>>>> 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 '06d4d73f1bd1c3a529d9c5337d3ee935 at 172.17.37.129:5060' Method: OPTIONS
>>>>> Really destroying SIP dialog '7228fb00-bdc1a1f7-3cf04c-2a2711ac at 172.17.39.42' 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: 8a007f00-bdc1a21f-1c2ed5-292711ac at 172.17.39.41
>>>>> 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: 8a007f00-bdc1a21f-1c2ed5-292711ac at 172.17.39.41
>>>>> 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 '8a007f00-bdc1a21f-1c2ed5-292711ac at 172.17.39.41' 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: 95ec4100-bdc1a233-208758-2b2711ac at 172.17.39.43
>>>>> User-Agent: Cisco-CU
>
> _______________________________________________
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> Intercomunicador e acesso remoto via rede IP e telefones IP
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