[AsteriskBrasil] Demora para Completar ligação

Sexta Novembro 2 20:50:17 -03 2018

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 em 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 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Date: Fri, 02 Nov 2018 19:11:47 GMT
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 172.17.39.42>
> Remote-Party-ID: "Giliardy Arena" <sip:9770 em 172.17.39.42
> >;party=calling;screen=yes;privacy=off
> Contact: <sip:9770 em 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 em 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 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 172.17.37.129:5060>
> Content-Length: 0
>
>
> <------------>
>
> <--- SIP read from UDP:172.17.39.42:5060 --->
> INVITE sip:2001 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Date: Fri, 02 Nov 2018 19:11:48 GMT
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 172.17.39.42>
> Remote-Party-ID: "Giliardy Arena" <sip:9770 em 172.17.39.42
> >;party=calling;screen=yes;privacy=off
> Contact: <sip:9770 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 172.17.37.129:5060>
> Content-Length: 0
>
>
> <------------>
>
> <--- SIP read from UDP:172.17.39.42:5060 --->
> INVITE sip:2001 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Date: Fri, 02 Nov 2018 19:11:49 GMT
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 172.17.39.42>
> Remote-Party-ID: "Giliardy Arena" <sip:9770 em 172.17.39.42
> >;party=calling;screen=yes;privacy=off
> Contact: <sip:9770 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 172.17.37.129:5060>
> Content-Length: 0
>
>
> <------------>
>
> <--- SIP read from UDP:172.17.39.42:5060 --->
> INVITE sip:2001 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Date: Fri, 02 Nov 2018 19:11:51 GMT
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 172.17.39.42>
> Remote-Party-ID: "Giliardy Arena" <sip:9770 em 172.17.39.42
> >;party=calling;screen=yes;privacy=off
> Contact: <sip:9770 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 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 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Date: Fri, 02 Nov 2018 19:11:55 GMT
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 172.17.39.42>
> Remote-Party-ID: "Giliardy Arena" <sip:9770 em 172.17.39.42
> >;party=calling;screen=yes;privacy=off
> Contact: <sip:9770 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 172.17.39.42' in 32000 ms (Method:
> OPTIONS)
>
> <--- SIP read from UDP:172.17.39.42:5060 --->
> INVITE sip:2001 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Date: Fri, 02 Nov 2018 19:12:03 GMT
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 172.17.39.42>
> Remote-Party-ID: "Giliardy Arena" <sip:9770 em 172.17.39.42
> >;party=calling;screen=yes;privacy=off
> Contact: <sip:9770 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 172.17.39.42' in 32000 ms (Method:
> OPTIONS)
>     -- Executing [2001 em 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 em 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 em 172.17.37.129>;tag=as1b69e3fc
> To: <sip:2001 em 172.17.90.170:50147;rinstance=a175c2caa1292efd>
> Contact: <sip:9770 em 172.17.37.129:5060>
> Call-ID: 50915352165f67003a4ada5854da2a90 em 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 em 172.17.90.170:50147;rinstance=a175c2caa1292efd>
> To: "2001"<sip:2001 em 172.17.90.170:50147
> ;rinstance=a175c2caa1292efd>;tag=a0a27e40
> From: "Giliardy Arena" <sip:9770 em 172.17.37.129>;tag=as1b69e3fc
> Call-ID: 50915352165f67003a4ada5854da2a90 em 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 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>;tag=as109d5c95
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 172.17.37.129:5060>
> Content-Length: 0
>
>
> <------------>
>
> <--- SIP read from UDP:172.17.39.42:5060 --->
> CANCEL sip:2001 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>
> Date: Fri, 02 Nov 2018 19:12:03 GMT
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>;tag=as109d5c95
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>;tag=as109d5c95
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 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 em 172.17.37.129>;tag=as1b69e3fc
> To: <sip:2001 em 172.17.90.170:50147;rinstance=a175c2caa1292efd>
> Call-ID: 50915352165f67003a4ada5854da2a90 em 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 em 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 em 172.17.39.42
> >;tag=95126358~d382d894-1bee-48ff-9c06-0f1d00364c70-36093796
> To: <sip:2001 em 172.17.37.129>;tag=as109d5c95
> Date: Fri, 02 Nov 2018 19:12:03 GMT
> Call-ID: 237b6100-bdc1a173-3cf01d-2a2711ac em 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 em 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 em 172.17.90.170:50147;rinstance=a175c2caa1292efd>
> To: <sip:2001 em 172.17.90.170:50147;rinstance=a175c2caa1292efd>;tag=a0a27e40
> From: "Giliardy Arena" <sip:9770 em 172.17.37.129>;tag=as1b69e3fc
> Call-ID: 50915352165f67003a4ada5854da2a90 em 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 em 172.17.90.170:50147;rinstance=a175c2caa1292efd>;tag=a0a27e40
> From: "Giliardy Arena" <sip:9770 em 172.17.37.129>;tag=as1b69e3fc
> Call-ID: 50915352165f67003a4ada5854da2a90 em 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 em 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 em 172.17.37.129>;tag=as1b69e3fc
> To: <sip:2001 em 172.17.90.170:50147;rinstance=a175c2caa1292efd>;tag=a0a27e40
> Contact: <sip:9770 em 172.17.37.129:5060>
> Call-ID: 50915352165f67003a4ada5854da2a90 em 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 em 172.17.37.129>;tag=as1a8e4d0e
> To: <sip:172.17.39.41>
> Contact: <sip:asterisk em 172.17.37.129:5060>
> Call-ID: 3a7729aa25ef52bc43a1e90a591f08f5 em 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 em 172.17.37.129>;tag=as2d9ec9dd
> To: <sip:172.17.39.42>
> Contact: <sip:asterisk em 172.17.37.129:5060>
> Call-ID: 1147063b71e9d1762b714dfc40cd0c82 em 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 em 172.17.37.129>;tag=as2deca9a9
> To: <sip:172.17.39.43>
> Contact: <sip:asterisk em 172.17.37.129:5060>
> Call-ID: 1b0f7dba03cb6de82484db42174bfa17 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 172.17.39.41' in 32000 ms (Method:
> OPTIONS)
> Really destroying SIP dialog '
> 28d8ab80-bdc1a17c-208720-2b2711ac em 172.17.39.43' Method: OPTIONS
> Really destroying SIP dialog '
> 29714200-bdc1a17d-3cf01e-2a2711ac em 172.17.39.42' Method: OPTIONS
> Really destroying SIP dialog '
> 50915352165f67003a4ada5854da2a90 em 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 em 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 em 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 em 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 em 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 em 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 em 172.17.39.42' in 32000 ms (Method:
> OPTIONS)
> Really destroying SIP dialog '
> 41e15c80-bdc1a1a6-1c2eb1-292711ac em 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 em 172.17.37.129>;tag=as5db9427e
> To: <sip:172.17.39.42>
> Contact: <sip:asterisk em 172.17.37.129:5060>
> Call-ID: 3157a8b527eff76f6747c88c1b6b1125 em 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 em 172.17.37.129>;tag=as3dded9ad
> To: <sip:172.17.39.41>
> Contact: <sip:asterisk em 172.17.37.129:5060>
> Call-ID: 452981b8491a141f7b9c74da6e6d991c em 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 em 172.17.37.129>;tag=as773015ab
> To: <sip:172.17.39.43>
> Contact: <sip:asterisk em 172.17.37.129:5060>
> Call-ID: 37532137052ad7ea72c034fa1d87a29d em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 172.17.37.129:5060' Method: OPTIONS
> Really destroying SIP dialog '
> 37532137052ad7ea72c034fa1d87a29d em 172.17.37.129:5060' Method: OPTIONS
> Really destroying SIP dialog '
> 4d348800-bdc1a1b9-208733-2b2711ac em 172.17.39.43' Method: OPTIONS
> Really destroying SIP dialog '
> 4d348800-bdc1a1b9-3cf038-2a2711ac em 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 em 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 em 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 em 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 em 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 em 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 em 172.17.39.42' in 32000 ms (Method:
> OPTIONS)
> Really destroying SIP dialog '
> 65a4a280-bdc1a1e2-1c2ec2-292711ac em 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 em 172.17.37.129>;tag=as753534e0
> To: <sip:172.17.39.42>
> Contact: <sip:asterisk em 172.17.37.129:5060>
> Call-ID: 3a24108b5fbea78c3e231c8a01761c4e em 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 em 172.17.37.129>;tag=as2f7fde70
> To: <sip:172.17.39.41>
> Contact: <sip:asterisk em 172.17.37.129:5060>
> Call-ID: 06d4d73f1bd1c3a529d9c5337d3ee935 em 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 em 172.17.37.129>;tag=as2db68d44
> To: <sip:172.17.39.43>
> Contact: <sip:asterisk em 172.17.37.129:5060>
> Call-ID: 2a43e4d20faf2382671b73ec19170e4c em 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 em 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 em 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 em 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 em 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 em 172.17.37.129:5060' Method: OPTIONS
> Really destroying SIP dialog '
> 2a43e4d20faf2382671b73ec19170e4c em 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 em 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 em 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 em 172.17.37.129:5060' Method: OPTIONS
> Really destroying SIP dialog '
> 7228fb00-bdc1a1f7-3cf04c-2a2711ac em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 172.17.39.42' in 32000 ms (Method:
> OPTIONS)
> Really destroying SIP dialog '
> 8a007f00-bdc1a21f-1c2ed5-292711ac em 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 em 172.17.37.129>;tag=as24cc7f1d
> To: <sip:172.17.39.42>
> Contact: <sip:asterisk em 172.17.37.129:5060>
> Call-ID: 1beaeccb13e9e06b6b47bb851e4546f5 em 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 em 172.17.37.129>;tag=as324cd423
> To: <sip:172.17.39.43>
> Contact: <sip:asterisk em 172.17.37.129:5060>
> Call-ID: 0a4788ce0a17d618752e78666aae9d9c em 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 em 172.17.37.129>;tag=as50a2d78b
> To: <sip:172.17.39.41>
> Contact: <sip:asterisk em 172.17.37.129:5060>
> Call-ID: 6418b3cf7406e0f26ce814f13a4d2f78 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 172.17.37.129:5060 SIP/2.0
>>> 19:23:11.496042 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP:
>>> INVITE sip:2001 em 172.17.37.129:5060 SIP/2.0
>>> 19:23:12.507249 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP:
>>> INVITE sip:2001 em 172.17.37.129:5060 SIP/2.0
>>> 19:23:14.513145 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP:
>>> INVITE sip:2001 em 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 em 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 em 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 em 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 em 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 em 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 em 172.17.37.129:5060 SIP/2.0
>>>> 16:47:32.254307 IP cucmservice01.sip > asterisk.ogmaster.local.sip:
>>>> SIP: INVITE sip:2005 em 172.17.37.129:5060 SIP/2.0
>>>> 16:47:33.258050 IP cucmservice01.sip > asterisk.ogmaster.local.sip:
>>>> SIP: INVITE sip:2005 em 172.17.37.129:5060 SIP/2.0
>>>> 16:47:35.272582 IP cucmservice01.sip > asterisk.ogmaster.local.sip:
>>>> SIP: INVITE sip:2005 em 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 em 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 em 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 em 172.17.37.129:5060 SIP/2.0
>>>> 16:47:59.755067 IP cucmservice01.sip > asterisk.ogmaster.local.sip:
>>>> SIP: ACK sip:2005 em 172.17.37.129:5060 SIP/2.0
>>>> 16:47:59.756284 IP cucmservice01.sip > asterisk.ogmaster.local.sip:
>>>> SIP: ACK sip:2005 em 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 em 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 em 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 em 172.17.37.129:5060'
>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Initializing initreq for
>>>> method OPTIONS - callid
>>>> 68e59f75777e9a5c455eac993191add0 em 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 em 172.17.37.129:5060'
>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Initializing initreq for
>>>> method OPTIONS - callid
>>>> 16c1f43e5149fd8d1e2f27cc630f3ee8 em 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 em 172.17.37.129:5060'
>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Initializing initreq for
>>>> method OPTIONS - callid
>>>> 447c59563b0c41e72d5fd888396c6d5d em 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 em 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 em 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 em 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 em 172.17.37.129:5060' of Request 102:
>>>> Match Found
>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Destroying SIP dialog
>>>> 68e59f75777e9a5c455eac993191add0 em 172.17.37.129:5060
>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Destroying SIP dialog
>>>> 16c1f43e5149fd8d1e2f27cc630f3ee8 em 172.17.37.129:5060
>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: = Looking for  Call ID:
>>>> 447c59563b0c41e72d5fd888396c6d5d em 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 em 172.17.37.129:5060' of Request 102:
>>>> Match Found
>>>> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Destroying SIP dialog
>>>> 447c59563b0c41e72d5fd888396c6d5d em 172.17.37.129:5060
>>>> [Oct 31 15:35:36] DEBUG[31072] chan_sip.c: Auto destroying SIP dialog '
>>>> ab2e6780-bd91f5d4-1f9f50-2b2711ac em 172.17.39.43'
>>>> [Oct 31 15:35:36] DEBUG[31072] chan_sip.c: Destroying SIP dialog
>>>> ab2e6780-bd91f5d4-1f9f50-2b2711ac em 172.17.39.43
>>>> [Oct 31 15:35:43] DEBUG[31072] chan_sip.c: Auto destroying SIP dialog '
>>>> af5a8500-bd91f5db-1b63e6-292711ac em 172.17.39.41'
>>>> [Oct 31 15:35:43] DEBUG[31072] chan_sip.c: Destroying SIP dialog
>>>> af5a8500-bd91f5db-1b63e6-292711ac em 172.17.39.41
>>>> [Oct 31 15:35:52] DEBUG[31072] chan_sip.c: Auto destroying SIP dialog '
>>>> b4b7cf80-bd91f5e4-3b3ad1-2a2711ac em 172.17.39.42'
>>>> [Oct 31 15:35:52] DEBUG[31072] chan_sip.c: Destroying SIP dialog
>>>> b4b7cf80-bd91f5e4-3b3ad1-2a2711ac em 172.17.39.42
>>>> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: = Looking for  Call ID:
>>>> cef1ad80-bd91f610-1f9f6a-2b2711ac em 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 em 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 em 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 em 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 em 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 em 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 em 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 em 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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