[AsteriskBrasil] Demora para Completar ligação

Sábado Novembro 3 16:19:36 -03 2018

Realmente estranho.

Tu pode ativar o log full do Asterisk (/etc/asterisk/logger.conf) e
depois recarregar o Asterisk e aih ativar o debug (core set debug 100)
e fazer outro teste. Os logs vao ficar no /var/log/asterisk/full (o
debug normalmente nao aparece no console).

Talvez assim tu encontre o problema. Concordo com o Bruno, DNS pode
ser um fator complicante.

[]s

Marcelo H. Terres <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 Sat, 3 Nov 2018 at 00:13, Giliardy Arena <giliardy.arena at gmail.com> wrote:
>
> Oi !
> Obrigado pela resposta e pela ajuda.
> Desculpe, não sei como enviar o arquivo.
>
> Nesta resposta estou tentando anexar via gmail.
> Espero que funcione, mas se não funcionar e puder me indicar a maneira correta.
>
> Utilizei a seguinte sintaxe :
>
> tcpdump -i ens192 src or dst 172.17.39.41 or 172.17.39.42 or 172.17.39.43 -w capture4.cap
>
>
> Sigo pesquisando =)
>
>
> Atenciosamente,
> Giliardy Correia Arena.
>
>
>
>
> Em sex, 2 de nov de 2018 às 17:22, Giliardy Arena <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-CUCM10.5
>> CSeq: 101 OPTIONS
>> Contact: <sip:172.17.39.43:5060>
>> Max-Forwards: 0
>> Content-Length: 0
>>
>> <------------->
>> --- (11 headers 0 lines) ---
>> Sending to 172.17.39.43:5060 (no NAT)
>> Looking for s in ramais (domain 172.17.37.129)
>>
>> <--- Transmitting (no NAT) to 172.17.39.43:5060 --->
>> SIP/2.0 404 Not Found
>> Via: SIP/2.0/UDP 172.17.39.43:5060;branch=z9hG4bK2bff60347a9104;received=172.17.39.43
>> From: <sip:172.17.39.43>;tag=486133364
>> To: <sip:172.17.37.129>;tag=as33d34b95
>> Call-ID: 95ec4100-bdc1a233-208758-2b2711ac 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 '95ec4100-bdc1a233-208758-2b2711ac 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=z9hG4bK95cc56189b3648
>> From: <sip:172.17.39.42>;tag=279128362
>> To: <sip:172.17.37.129>
>> Date: Fri, 02 Nov 2018 19:15:00 GMT
>> Call-ID: 9684d780-bdc1a234-3cf05f-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=z9hG4bK95cc56189b3648;received=172.17.39.42
>> From: <sip:172.17.39.42>;tag=279128362
>> To: <sip:172.17.37.129>;tag=as562795db
>> Call-ID: 9684d780-bdc1a234-3cf05f-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 '9684d780-bdc1a234-3cf05f-2a2711ac at 172.17.39.42' in 32000 ms (Method: OPTIONS)
>> Really destroying SIP dialog '8a007f00-bdc1a21f-1c2ed5-292711ac at 172.17.39.41' Method: OPTIONS
>>
>> <--- SIP read from UDP:172.17.90.170:50147 --->
>>
>>
>> <------------->
>> infoasterisk*CLI>
>> infoasterisk*CLI>
>> Reliably Transmitting (no NAT) to 172.17.39.42:5060:
>> OPTIONS sip:172.17.39.42 SIP/2.0
>> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK765e1b02
>> Max-Forwards: 70
>> From: "asterisk" <sip:asterisk at 172.17.37.129>;tag=as24cc7f1d
>> To: <sip:172.17.39.42>
>> Contact: <sip:asterisk at 172.17.37.129:5060>
>> Call-ID: 1beaeccb13e9e06b6b47bb851e4546f5 at 172.17.37.129:5060
>> CSeq: 102 OPTIONS
>> User-Agent: Asterisk PBX 13.23.1
>> Date: Fri, 02 Nov 2018 19:15:28 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
>> Supported: replaces, timer
>> Content-Length: 0
>>
>>
>> ---
>> Reliably Transmitting (no NAT) to 172.17.39.43:5060:
>> OPTIONS sip:172.17.39.43 SIP/2.0
>> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK18f5a08f
>> Max-Forwards: 70
>> From: "asterisk" <sip:asterisk at 172.17.37.129>;tag=as324cd423
>> To: <sip:172.17.39.43>
>> Contact: <sip:asterisk at 172.17.37.129:5060>
>> Call-ID: 0a4788ce0a17d618752e78666aae9d9c at 172.17.37.129:5060
>> CSeq: 102 OPTIONS
>> User-Agent: Asterisk PBX 13.23.1
>> Date: Fri, 02 Nov 2018 19:15:28 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
>> Supported: replaces, timer
>> Content-Length: 0
>>
>>
>> ---
>> Reliably Transmitting (no NAT) to 172.17.39.41:5060:
>> OPTIONS sip:172.17.39.41 SIP/2.0
>> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK448e3148
>> Max-Forwards: 70
>> From: "asterisk" <sip:asterisk at 172.17.37.129>;tag=as50a2d78b
>> To: <sip:172.17.39.41>
>> Contact: <sip:asterisk at 172.17.37.129:5060>
>> Call-ID: 6418b3cf7406e0f26ce814f13a4d2f78 at 172.17.37.129:5060
>> CSeq: 102 OPTIONS
>> User-Agent: Asterisk PBX 13.23.1
>> Date: Fri, 02 Nov 2018 19:15:28 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
>> Supported: replaces, timer
>> Content-Length: 0
>>
>>
>> ---
>>
>> <--- SIP read from UDP:172.17.39.42:5060 --->
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 172.17.37.129:5060;branch=z9hG4bK765e1b02
>> From: "asterisk" <sip:asterisk at 172.17.37.129>;tag=as24cc7f1d
>> To: <sip:172.17.39.42>;tag=1358087302
>> Date: Fri, 02 Nov 2018 19:15:24 GMT
>> Call-ID: 1beaeccb13e9e06b6b47bb851e4546f5 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 '1beaeccb13e9e06b6b47bb851e4546f5 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=z9hG4bK448e3148
>> From: "asterisk" <sip:asterisk at 172.17.37.129>;tag=as50a2d78b
>> To: <sip:172.17.39.41>;tag=319483522
>> Date: Fri, 02 Nov 2018 19:15:24 GMT
>> Call-ID: 6418b3cf7406e0f26ce814f13a4d2f78 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 '6418b3cf7406e0f26ce814f13a4d2f78 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=z9hG4bK18f5a08f
>> From: "asterisk" <sip:asterisk at 172.17.37.129>;tag=as324cd423
>> To: <sip:172.17.39.43>;tag=635128065
>> Date: Fri, 02 Nov 2018 19:15:24 GMT
>> Call-ID: 0a4788ce0a17d618752e78666aae9d9c 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 '0a4788ce0a17d618752e78666aae9d9c at 172.17.37.129:5060' Method: OPTIONS
>> infoasterisk*CLI> sip set debug off
>> SIP Debugging Disabled
>> infoasterisk*CLI>
>> infoasterisk*CLI>
>> infoasterisk*CLI>
>> infoasterisk*CLI>
>>
>>
>> Atenciosamente,
>> Giliardy Correia Arena.
>>
>>
>>
>>
>> Em sex, 2 de nov de 2018 às 13:14, Giliardy Arena <giliardy.arena at gmail.com> escreveu:
>>>
>>> Boa tarde.
>>> Obrigado pela resposta, Rogerio.
>>>
>>> Sim , já testei como uma extensão simples e o cenário é o mesmo.
>>>
>>> No CLI eu só enxergo LOG quando a chamada é conectada.
>>> Não consigo ver nada diferente antes desse momento.
>>>
>>> Via tcpdump eu vejo as tentativas, mas não consigo identificar a causa do atraso através dele.
>>> Me chamou atenção a tentativa do Asterisk em todos os IPs do Call Manager, quando ele deveria se conectar diretamente ao que enviou a chamada.
>>>
>>> Você tem alguma sugestão que eu possa fazer no CLI para tentar enxergar a tentativa desde o recebimento do INVITE ?
>>>
>>> Atenciosamente,
>>> Giliardy Correia Arena.
>>>
>>>
>>>
>>>
>>> Em qui, 1 de nov de 2018 às 19:24, Giliardy Arena <giliardy.arena at gmail.com> escreveu:
>>>>
>>>> Sim !
>>>>
>>>> Os ramais ficam no Cisco. Eu apenas vou ligar para um numero do Asterisk que vai gravar as ligações.
>>>> Veja uma nova captura
>>>>
>>>> A troca de mensagens OPTION com os servidores que não possuem o ramal que eu estou chamado do Cisco que parece estar atrasando.... Mas não sei como resolver, pois já forcei apenas um servidor no sip.conf
>>>>
>>>>
>>>> 19:23:10.984078 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: INVITE sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>> 19:23:11.496042 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: INVITE sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>> 19:23:12.507249 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: INVITE sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>> 19:23:14.513145 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: INVITE sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>> 19:23:15.983468 ARP, Request who-has asterisk.ogmaster.local tell cucmservice01, length 46
>>>> 19:23:15.983484 ARP, Reply asterisk.ogmaster.local is-at 00:50:56:90:dc:d1 (oui Unknown), length 28
>>>> 19:23:18.524150 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: INVITE sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>> 19:23:19.220165 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:19.726828 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:20.739614 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:22.706629 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:22.755062 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:23.213088 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:24.220115 ARP, Request who-has asterisk.ogmaster.local tell cucmservice02, length 46
>>>> 19:23:24.220130 ARP, Reply asterisk.ogmaster.local is-at 00:50:56:90:dc:d1 (oui Unknown), length 28
>>>> 19:23:24.224829 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:24.292071 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:24.808252 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:25.810898 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:26.240672 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:26.533679 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: INVITE sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>> 19:23:26.762741 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:27.827149 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:29.292152 ARP, Request who-has asterisk.ogmaster.local tell infocucmpub, length 46
>>>> 19:23:29.292168 ARP, Reply asterisk.ogmaster.local is-at 00:50:56:90:dc:d1 (oui Unknown), length 28
>>>> 19:23:30.247068 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:30.769748 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:31.835377 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:34.259328 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:34.784241 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:35.845668 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:38.268704 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>> 19:23:38.797238 ARP, Request who-has 172.17.39.48 tell cucmservice02, length 46
>>>> 19:23:38.989294 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 100 Trying
>>>> 19:23:38.989552 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 100 Trying
>>>> 19:23:38.989649 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 100 Trying
>>>> 19:23:38.989743 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 100 Trying
>>>> 19:23:38.989824 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 100 Trying
>>>> 19:23:38.989979 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.990068 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.990155 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.990257 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.990339 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.990409 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.990505 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.990611 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.990688 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.990777 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.990878 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.990994 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 100 Trying
>>>> 19:23:38.991069 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.991130 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.991218 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.991311 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.991460 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.991545 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.991636 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.991723 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:38.991807 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 404 Not Found
>>>> 19:23:39.085356 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 180 Ringing
>>>> 19:23:39.797232 ARP, Request who-has 172.17.39.48 tell cucmservice02, length 46
>>>> 19:23:40.768521 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: CANCEL sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>> 19:23:40.768819 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 487 Request Terminated
>>>> 19:23:40.768869 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 200 OK
>>>> 19:23:40.771996 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: ACK sip:2001 at 172.17.37.129:5060 SIP/2.0
>>>> 19:23:40.797266 ARP, Request who-has 172.17.39.48 tell cucmservice02, length 46
>>>>
>>>> Atenciosamente,
>>>> Giliardy Correia Arena.
>>>>
>>>>
>>>>
>>>>
>>>> Em qui, 1 de nov de 2018 às 17:30, Giliardy Arena <giliardy.arena at gmail.com> escreveu:
>>>>>
>>>>> Oi Luiz.
>>>>> Estabeleci um SIP entre o Call Manager e o Asterisk.
>>>>> O Call Manager possui um Publisher (39.41) e os Subscribers (39.42 e 39.43), onde ficam os telefones registrados.
>>>>>
>>>>> Já testei tanto deixando todos os IPs possíveis do Call Manager, quanto apenas a referente ao registro do meu telefone no Call Manager(39.42) e a demora é a mesma.
>>>>>
>>>>> ;[callman01]
>>>>> ;type=friend
>>>>> ;context=ramais
>>>>> ;host=172.17.39.41
>>>>> ;disallow=all
>>>>> ;allow=ulaw
>>>>> ;allow=alaw
>>>>> ;nat=no
>>>>> ;canreinvite=yes
>>>>> ;qualify=yes
>>>>>
>>>>> [callman02]
>>>>> type=friend
>>>>> context=ramais
>>>>> host=172.17.39.42
>>>>> disallow=all
>>>>> allow=ulaw
>>>>> allow=alaw
>>>>> nat=no
>>>>> canreinvite=yes
>>>>> qualify=yes
>>>>>
>>>>> ;[callman03]
>>>>> ;type=friend
>>>>> ;context=ramais
>>>>> ;host=172.17.39.43
>>>>> ;disallow=all
>>>>> ;allow=ulaw
>>>>> ;allow=alaw
>>>>> ;nat=no
>>>>> ;canreinvite=yes
>>>>> ;qualify=yes
>>>>>
>>>>>
>>>>>
>>>>> Do lado do Call Manager está tudo configurado e eles estão falando UDP.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> No lado do Asterisk , não consegui alguma captura especifica, mas peguei via TCPDUMP que ele parece tentar todos antes de efetivamente fechar com o primeiro , embora já tenha recebido INVITE do correto.
>>>>>
>>>>>
>>>>>
>>>>> tcpdump -i ens192 dst 172.17.37.129 and src 172.17.39.41 or 172.17.39.42 or 172.17.39.43
>>>>>
>>>>>
>>>>> 16:47:31.740674 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: INVITE sip:2005 at 172.17.37.129:5060 SIP/2.0
>>>>> 16:47:32.254307 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: INVITE sip:2005 at 172.17.37.129:5060 SIP/2.0
>>>>> 16:47:33.258050 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: INVITE sip:2005 at 172.17.37.129:5060 SIP/2.0
>>>>> 16:47:35.272582 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: INVITE sip:2005 at 172.17.37.129:5060 SIP/2.0
>>>>> 16:47:38.225049 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:38.740848 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:39.282208 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: INVITE sip:2005 at 172.17.37.129:5060 SIP/2.0
>>>>> 16:47:39.751717 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:41.754129 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:43.224610 ARP, Request who-has asterisk.ogmaster.local tell infocucmpub, length 46
>>>>> 16:47:45.768670 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:46.055483 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:46.560533 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:47.292581 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: INVITE sip:2005 at 172.17.37.129:5060 SIP/2.0
>>>>> 16:47:47.572900 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:49.587485 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:49.780979 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:51.054865 ARP, Request who-has asterisk.ogmaster.local tell cucmservice02, length 46
>>>>> 16:47:52.292278 ARP, Request who-has asterisk.ogmaster.local tell cucmservice01, length 46
>>>>> 16:47:53.596301 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:53.785687 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:57.607030 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:47:59.754553 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: ACK sip:2005 at 172.17.37.129:5060 SIP/2.0
>>>>> 16:47:59.755067 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: ACK sip:2005 at 172.17.37.129:5060 SIP/2.0
>>>>> 16:47:59.756284 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: ACK sip:2005 at 172.17.37.129:5060 SIP/2.0
>>>>> 16:48:00.535923 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: OPTIONS sip:172.17.37.129:5060 SIP/2.0
>>>>> 16:48:02.126054 IP infocucmpub.sip > asterisk.ogmaster.local.sip: SIP: SIP/2.0 200 OK
>>>>> 16:48:02.220213 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP: SIP/2.0 200 OK
>>>>> 16:48:02.220484 IP cucmservice02.sip > asterisk.ogmaster.local.sip: SIP: SIP/2.0 200 OK
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> tcpdump -i ens192 src 172.17.37.129 and dst 172.17.39.41 or 172.17.39.42 or 172.17.39.43
>>>>>
>>>>>
>>>>> 16:47:59.749555 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 100 Trying
>>>>> 16:47:59.749932 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 100 Trying
>>>>> 16:47:59.750055 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 100 Trying
>>>>> 16:47:59.750181 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 100 Trying
>>>>> 16:47:59.750348 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:47:59.750472 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:47:59.750514 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 200 OK
>>>>> 16:47:59.750797 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 100 Trying
>>>>> 16:47:59.750935 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 200 OK
>>>>> 16:47:59.751084 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:47:59.751193 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:47:59.751293 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:47:59.751487 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:47:59.751608 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:47:59.751761 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 100 Trying
>>>>> 16:47:59.751864 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 200 OK
>>>>> 16:47:59.751998 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:47:59.752116 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:47:59.752230 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:47:59.752343 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:47:59.752458 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:47:59.752576 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:48:00.536313 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 404 Not Found
>>>>> 16:48:02.124006 IP asterisk.ogmaster.local.sip > infocucmpub.sip: SIP: OPTIONS sip:172.17.39.41 SIP/2.0
>>>>> 16:48:02.218575 IP asterisk.ogmaster.local.sip > cucmservice02.sip: SIP: OPTIONS sip:172.17.39.43 SIP/2.0
>>>>> 16:48:02.218761 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: OPTIONS sip:172.17.39.42 SIP/2.0
>>>>> 16:48:02.589632 IP asterisk.ogmaster.local.sip > cucmservice01.sip: SIP: SIP/2.0 200 OK
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Testei alguns Debugs que fui pesquisando na internet mas não consegui compreender muito bem....
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> [Oct 31 15:35:20] DEBUG[31072] chan_sip.c: = Looking for  Call ID: b4b7cf80-bd91f5e4-3b3ad1-2a2711ac at 172.17.39.42 (Checking From) --From tag 1146601895 --To-tag
>>>>> [Oct 31 15:35:20] DEBUG[31072] acl.c: For destination '172.17.39.42', our source address is '172.17.37.129'.
>>>>> [Oct 31 15:35:20] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with address 172.17.37.129:5060
>>>>> [Oct 31 15:35:20] DEBUG[31072] netsock2.c: Splitting '172.17.39.42:5060' into...
>>>>> [Oct 31 15:35:20] DEBUG[31072] netsock2.c: ...host '172.17.39.42' and port '5060'.
>>>>> [Oct 31 15:35:20] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for b4b7cf80-bd91f5e4-3b3ad1-2a2711ac at 172.17.39.42 - OPTIONS (No RTP)
>>>>> [Oct 31 15:35:20] DEBUG[31072] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
>>>>> [Oct 31 15:35:20] DEBUG[31072] netsock2.c: Splitting '172.17.37.129:5060' into...
>>>>> [Oct 31 15:35:20] DEBUG[31072] netsock2.c: ...host '172.17.37.129' and port ''.
>>>>> [Oct 31 15:35:20] DEBUG[31072] netsock2.c: Splitting '172.17.39.42' into...
>>>>> [Oct 31 15:35:20] DEBUG[31072] netsock2.c: ...host '172.17.39.42' and port ''.
>>>>> [Oct 31 15:35:20] DEBUG[31072] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 172.17.39.42:5060
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for 7eeb423d62baf89b2376864b55f025a9@[fe80::a0e0:69c4:bc8b:b417]:5060 - OPTIONS (No RTP)
>>>>> [Oct 31 15:35:21] DEBUG[31072] acl.c: For destination '172.17.39.43', our source address is '172.17.37.129'.
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with address 172.17.37.129:5060
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: SIP call-id changed from '7eeb423d62baf89b2376864b55f025a9@[fe80::a0e0:69c4:bc8b:b417]:5060' to '68e59f75777e9a5c455eac993191add0 at 172.17.37.129:5060'
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Initializing initreq for method OPTIONS - callid 68e59f75777e9a5c455eac993191add0 at 172.17.37.129:5060
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.17.39.43:5060
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for 2b73bb0d2c3469fa0780743f3270ca4f@[fe80::a0e0:69c4:bc8b:b417]:5060 - OPTIONS (No RTP)
>>>>> [Oct 31 15:35:21] DEBUG[31072] acl.c: For destination '172.17.39.42', our source address is '172.17.37.129'.
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with address 172.17.37.129:5060
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: SIP call-id changed from '2b73bb0d2c3469fa0780743f3270ca4f@[fe80::a0e0:69c4:bc8b:b417]:5060' to '16c1f43e5149fd8d1e2f27cc630f3ee8 at 172.17.37.129:5060'
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Initializing initreq for method OPTIONS - callid 16c1f43e5149fd8d1e2f27cc630f3ee8 at 172.17.37.129:5060
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.17.39.42:5060
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for 3c48a6e96480adda0d8af61a4d498fb7@[fe80::a0e0:69c4:bc8b:b417]:5060 - OPTIONS (No RTP)
>>>>> [Oct 31 15:35:21] DEBUG[31072] acl.c: For destination '172.17.39.41', our source address is '172.17.37.129'.
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with address 172.17.37.129:5060
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: SIP call-id changed from '3c48a6e96480adda0d8af61a4d498fb7@[fe80::a0e0:69c4:bc8b:b417]:5060' to '447c59563b0c41e72d5fd888396c6d5d at 172.17.37.129:5060'
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Initializing initreq for method OPTIONS - callid 447c59563b0c41e72d5fd888396c6d5d at 172.17.37.129:5060
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.17.39.41:5060
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: = Looking for  Call ID: 68e59f75777e9a5c455eac993191add0 at 172.17.37.129:5060 (Checking To) --From tag as2ee346e2 --To-tag 348178859
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Stopping retransmission on '68e59f75777e9a5c455eac993191add0 at 172.17.37.129:5060' of Request 102: Match Found
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: = Looking for  Call ID: 16c1f43e5149fd8d1e2f27cc630f3ee8 at 172.17.37.129:5060 (Checking To) --From tag as138ca155 --To-tag 802041871
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Stopping retransmission on '16c1f43e5149fd8d1e2f27cc630f3ee8 at 172.17.37.129:5060' of Request 102: Match Found
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Destroying SIP dialog 68e59f75777e9a5c455eac993191add0 at 172.17.37.129:5060
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Destroying SIP dialog 16c1f43e5149fd8d1e2f27cc630f3ee8 at 172.17.37.129:5060
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: = Looking for  Call ID: 447c59563b0c41e72d5fd888396c6d5d at 172.17.37.129:5060 (Checking To) --From tag as34b82738 --To-tag 605276003
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Stopping retransmission on '447c59563b0c41e72d5fd888396c6d5d at 172.17.37.129:5060' of Request 102: Match Found
>>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Destroying SIP dialog 447c59563b0c41e72d5fd888396c6d5d at 172.17.37.129:5060
>>>>> [Oct 31 15:35:36] DEBUG[31072] chan_sip.c: Auto destroying SIP dialog 'ab2e6780-bd91f5d4-1f9f50-2b2711ac at 172.17.39.43'
>>>>> [Oct 31 15:35:36] DEBUG[31072] chan_sip.c: Destroying SIP dialog ab2e6780-bd91f5d4-1f9f50-2b2711ac at 172.17.39.43
>>>>> [Oct 31 15:35:43] DEBUG[31072] chan_sip.c: Auto destroying SIP dialog 'af5a8500-bd91f5db-1b63e6-292711ac at 172.17.39.41'
>>>>> [Oct 31 15:35:43] DEBUG[31072] chan_sip.c: Destroying SIP dialog af5a8500-bd91f5db-1b63e6-292711ac at 172.17.39.41
>>>>> [Oct 31 15:35:52] DEBUG[31072] chan_sip.c: Auto destroying SIP dialog 'b4b7cf80-bd91f5e4-3b3ad1-2a2711ac at 172.17.39.42'
>>>>> [Oct 31 15:35:52] DEBUG[31072] chan_sip.c: Destroying SIP dialog b4b7cf80-bd91f5e4-3b3ad1-2a2711ac at 172.17.39.42
>>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: = Looking for  Call ID: cef1ad80-bd91f610-1f9f6a-2b2711ac at 172.17.39.43 (Checking From) --From tag 1522038610 --To-tag
>>>>> [Oct 31 15:36:04] DEBUG[31072] acl.c: For destination '172.17.39.43', our source address is '172.17.37.129'.
>>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with address 172.17.37.129:5060
>>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with address 172.17.37.129:5060
>>>>> [Oct 31 15:36:04] DEBUG[31072] netsock2.c: Splitting '172.17.39.43:5060' into...
>>>>> [Oct 31 15:36:04] DEBUG[31072] netsock2.c: ...host '172.17.39.43' and port '5060'.
>>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for cef1ad80-bd91f610-1f9f6a-2b2711ac at 172.17.39.43 - OPTIONS (No RTP)
>>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
>>>>> [Oct 31 15:36:04] DEBUG[31072] netsock2.c: Splitting '172.17.37.129:5060' into...
>>>>> [Oct 31 15:36:04] DEBUG[31072] netsock2.c: ...host '172.17.37.129' and port ''.
>>>>> [Oct 31 15:36:04] DEBUG[31072] netsock2.c: Splitting '172.17.39.43' into...
>>>>> [Oct 31 15:36:04] DEBUG[31072] netsock2.c: ...host '172.17.39.43' and port ''.
>>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 172.17.39.43:5060
>>>>> [Oct 31 15:36:12] DEBUG[31072] chan_sip.c: = Looking for  Call ID: d3b66180-bd91f618-1b63f9-292711ac at 172.17.39.41 (Checking From) --From tag 639004019 --To-tag
>>>>> [Oct 31 15:36:12] DEBUG[31072] acl.c: For destination '172.17.39.41', our source address is '172.17.37.129'.
>>>>> [Oct 31 15:36:12] DEBUG[31072] chan_sip.c: Setting AST_TRANSPORT_UDP with address 172.17.37.129:5060
>>>>> [Oct 31 15:36:12] DEBUG[31072] netsock2.c: Splitting '172.17.39.41:5060' into...
>>>>> [Oct 31 15:36:12] DEBUG[31072] netsock2.c: ...host '172.17.39.41' and port '5060'.
>>>>> [Oct 31 15:36:12] DEBUG[31072] chan_sip.c: Allocating new SIP dialog for d3b66180-bd91f618-1b63f9-292711ac at 172.17.39.41 - OPTIONS (No RTP)
>>>>> [Oct 31 15:36:12] DEBUG[31072] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
>>>>> [Oct 31 15:36:12] DEBUG[31072] netsock2.c: Splitting '172.17.37.129:5060' into...
>>>>> [Oct 31 15:36:12] DEBUG[31072] netsock2.c: ...host '172.17.37.129' and port ''.
>>>>> [Oct 31 15:36:12] DEBUG[31072] netsock2.c: Splitting '172.17.39.41' into...
>>>>> [Oct 31 15:36:12] DEBUG[31072] netsock2.c: ...host '172.17.39.41' and port ''.
>>>>> [Oct 31 15:36:12] DEBUG[31072] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 172.17.39.41:5060
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Atenciosamente,
>>>>> Giliardy Correia Arena.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Em qui, 1 de nov de 2018 às 15:05, Giliardy Arena <giliardy.arena at gmail.com> escreveu:
>>>>>>
>>>>>> Olá pessoal !
>>>>>> Alguma ajuda ?  Alguma dica ?
>>>>>>
>>>>>> Obrigado
>>>>>>
>>>>>>
>>>>>> Atenciosamente,
>>>>>> Giliardy Correia Arena.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Em qua, 31 de out de 2018 às 10:58, Giliardy Arena <giliardy.arena at gmail.com> escreveu:
>>>>>>>
>>>>>>> Olá , bom dia.
>>>>>>>
>>>>>>> Alguém sugere alguma forma de eu rastrear a ligação desde a chegada da requisicao SIP no servidor Asterisk , para entender o motivo de demorar muito para conectar? Algum debug específico, um trace , um log...
>>>>>>>
>>>>>>> Obrigado
>>>>>>>
>>>>>>> Em ter, 30 de out de 2018 20:22, Giliardy Arena <giliardy.arena at gmail.com> escreveu:
>>>>>>>>
>>>>>>>> Sylvio
>>>>>>>>
>>>>>>>> O waitforsilence é para identificar se não tiver mais conversação e encerrar a ligação.
>>>>>>>> Para evitar ficar alguma chamada presa gravando eternamente.
>>>>>>>>
>>>>>>>>
>>>>>>>> Atenciosamente,
>>>>>>>> Giliardy Correia Arena.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Em ter, 30 de out de 2018 às 17:57, Giliardy Arena <giliardy.arena at gmail.com> escreveu:
>>>>>>>>>
>>>>>>>>> Caros,
>>>>>>>>> Boa tarde.
>>>>>>>>>
>>>>>>>>> Estou aprendendo e estudando sobre o Asterisk.
>>>>>>>>> Atualmente administro um Cisco Call Manager e a minha ideia é usar o Asterisk para gravar ligações recebidas do Call Manager.
>>>>>>>>>
>>>>>>>>> Fiz a integração do Asterisk com o Call Manager com sucesso.
>>>>>>>>>
>>>>>>>>> Estou com problema para entender o motivo do Asterisk demorar para conectar a ligação a uma extensão. Tenho pesquisado, mas com dificuldades para entender como debugar.
>>>>>>>>>
>>>>>>>>> Criei a seguinte extensão, que atende sozinha e grava.
>>>>>>>>>
>>>>>>>>> exten => 2005,1,Answer()
>>>>>>>>> exten => 2005,n,MixMonitor(Ramal-${CALLERID(num)}-Em-${STRFTIME(${EPOCH},,%d-%m-%Y-%H-%M)}.wav)
>>>>>>>>> exten => 2005,n,WaitForSilence(10000|6)
>>>>>>>>> exten => 2005,n,Hangup
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Também experimentei o mesmo sintoma através de uma extensão que criei e loguei numa softphone.
>>>>>>>>>
>>>>>>>>> - Ativei Debug full , mas não tem nenhuma mensagem importante. Apenas o que vejo na CLI do asterisk
>>>>>>>>>
>>>>>>>>> - Na CLI do Asterisk só vejo log quando a chamada efetivamente é conectada, não sei se consigo ver desde o momento que ele recebe a requisição.
>>>>>>>>>
>>>>>>>>> - Fiz um TCPDUMP e realmente me parece que é o Asterisk demorando a conectar a extensão, mas via TCPDUMP não tenho detalhes para entender e ajustar. Demora aproximadamente 30segundos após chamar do Call Manager.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Alguém pode me dar um help de por onde eu posso rastrear para tentar corrigir ?
>>>>>>>>>
>>>>>>>>> Obrigado!
>>>>>>>>>
>>>>>>>>> Atenciosamente,
>>>>>>>>> Giliardy Correia Arena.
>>>>>>>>>
>>>>>>>>>
> _______________________________________________
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> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7
> Intercomunicador e acesso remoto via rede IP e telefones IP
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