[AsteriskBrasil] Demora para Completar ligação
Giliardy Arena
giliardy.arena em gmail.com
Sexta Novembro 2 17:22:26 -03 2018
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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