Todos os ramais ficam na cisco ?
Rogerio Santos ■Mobile
On Thu, Nov 1, 2018, 17:48 Giliardy Arena <giliardy.ar...@gmail.com> wrote:
> 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@172.17.37.129:5060 SIP/2.0
> 16:47:32.254307 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP:
> INVITE sip:2005@172.17.37.129:5060 SIP/2.0
> 16:47:33.258050 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP:
> INVITE sip:2005@172.17.37.129:5060 SIP/2.0
> 16:47:35.272582 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP:
> INVITE sip:2005@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@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@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@172.17.37.129:5060 SIP/2.0
> 16:47:59.755067 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP:
> ACK sip:2005@172.17.37.129:5060 SIP/2.0
> 16:47:59.756284 IP cucmservice01.sip > asterisk.ogmaster.local.sip: SIP:
> ACK sip:2005@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@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@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@172.17.37.129:5060'
> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Initializing initreq for method
> OPTIONS - callid 68e59f75777e9a5c455eac993191add0@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@172.17.37.129:5060'
> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Initializing initreq for method
> OPTIONS - callid 16c1f43e5149fd8d1e2f27cc630f3ee8@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@172.17.37.129:5060'
> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Initializing initreq for method
> OPTIONS - callid 447c59563b0c41e72d5fd888396c6d5d@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@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@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@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@172.17.37.129:5060' of Request 102:
> Match Found
> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Destroying SIP dialog
> 68e59f75777e9a5c455eac993191add0@172.17.37.129:5060
> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Destroying SIP dialog
> 16c1f43e5149fd8d1e2f27cc630f3ee8@172.17.37.129:5060
> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: = Looking for Call ID:
> 447c59563b0c41e72d5fd888396c6d5d@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@172.17.37.129:5060' of Request 102:
> Match Found
> [Oct 31 15:35:21] DEBUG[31072] chan_sip.c: Destroying SIP dialog
> 447c59563b0c41e72d5fd888396c6d5d@172.17.37.129:5060
> [Oct 31 15:35:36] DEBUG[31072] chan_sip.c: Auto destroying SIP dialog '
> ab2e6780-bd91f5d4-1f9f50-2b2711ac@172.17.39.43'
> [Oct 31 15:35:36] DEBUG[31072] chan_sip.c: Destroying SIP dialog
> ab2e6780-bd91f5d4-1f9f50-2b2711ac@172.17.39.43
> [Oct 31 15:35:43] DEBUG[31072] chan_sip.c: Auto destroying SIP dialog '
> af5a8500-bd91f5db-1b63e6-292711ac@172.17.39.41'
> [Oct 31 15:35:43] DEBUG[31072] chan_sip.c: Destroying SIP dialog
> af5a8500-bd91f5db-1b63e6-292711ac@172.17.39.41
> [Oct 31 15:35:52] DEBUG[31072] chan_sip.c: Auto destroying SIP dialog '
> b4b7cf80-bd91f5e4-3b3ad1-2a2711ac@172.17.39.42'
> [Oct 31 15:35:52] DEBUG[31072] chan_sip.c: Destroying SIP dialog
> b4b7cf80-bd91f5e4-3b3ad1-2a2711ac@172.17.39.42
> [Oct 31 15:36:04] DEBUG[31072] chan_sip.c: = Looking for Call ID:
> cef1ad80-bd91f610-1f9f6a-2b2711ac@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@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@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@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.ar...@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.ar...@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.ar...@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.ar...@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.
>>>>>
>>>>>
>>>>> _______________________________________________
> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1
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_______________________________________________
KHOMP: completa linha de placas externas FXO, FXS, GSM e E1
Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7
Intercomunicador e acesso remoto via rede IP e telefones IP
Conheça todo o portfólio em www.Khomp.com
_______________________________________________
Para remover seu email desta lista, basta enviar um email em branco para
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