[AsteriskBrasil] Demora para Completar ligação
Giliardy Arena
giliardy.arena em gmail.com
Quinta Novembro 1 17:30:00 -03 2018
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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