[AsteriskBrasil] X100p Clone continua ...
Emanuel dos Reis Rodrigues
emanueldosreis em gmail.com
Terça Novembro 11 20:52:19 BRST 2008
Tiago Durante wrote:
> oi,
> não sei se isso pode ajudar, mas testa mudar isso na conf da placa:
>
>
> loadzone= us
> defaultzone = us
>
> para
>
> loadzone= br
> defaultzone = br
>
>
> []s
>
>
>
> 2008/11/11 Emanuel dos Reis Rodrigues <emanueldosreis em gmail.com>:
>
>> Tiago Durante wrote:
>>
>>> Desculpa, me expressei mal... Qdo disse clone na verdade quis dizer x100p...
>>>
>>> Vc tem algum erro na inicializacão do Asterisk?
>>> Alguma mensagem no dmesg... Saida do dahdi_cfg...
>>>
>>> Vai ter que ver bastante coisa... eu estou meio sem ideia no momento.. =(
>>>
>>> se vc quiser me chama no gtalk que eu vejo se posso te ajudar em algo...
>>>
>>>
>>> []!
>>>
>>>
>>> 2008/11/11 Emanuel dos Reis Rodrigues <emanueldosreis em gmail.com>:
>>>
>>>
>>>> Tiago Durante wrote:
>>>>
>>>>
>>>>> cara... muito estranho..
>>>>>
>>>>> vc ja usou essa placa com outro asterisk antes?
>>>>>
>>>>> ja pensou em comprar uma clone? rs
>>>>>
>>>>>
>>>>>
>>>>> []s
>>>>>
>>>>>
>>>>> 2008/11/11 Emanuel dos Reis Rodrigues <emanueldosreis em gmail.com>:
>>>>>
>>>>>
>>>>>
>>>>>> segue meu interrupts
>>>>>>
>>>>>>
>>>>>> asterisk:~# cat /proc/interrupts
>>>>>> CPU0
>>>>>> 0: 1001168 IO-APIC-edge timer
>>>>>> 1: 229 IO-APIC-edge i8042
>>>>>> 6: 5 IO-APIC-edge floppy
>>>>>> 7: 2 IO-APIC-edge parport0
>>>>>> 8: 1 IO-APIC-edge rtc
>>>>>> 9: 0 IO-APIC-level acpi
>>>>>> 12: 113 IO-APIC-edge i8042
>>>>>> 14: 2620 IO-APIC-edge ide0
>>>>>> 15: 12023 IO-APIC-edge ide1
>>>>>> 169: 0 IO-APIC-level uhci_hcd:usb1, uhci_hcd:usb2,
>>>>>> uhci_hcd:usb3, ehci_hcd:usb4
>>>>>> 177: 15704 IO-APIC-level eth0
>>>>>> 185: 0 IO-APIC-level VIA8233
>>>>>> 193: 153228 IO-APIC-level wcfxo
>>>>>> NMI: 0
>>>>>> LOC: 1001124
>>>>>> ERR: 0
>>>>>> MIS: 0
>>>>>>
>>>>>> As chamadas entrantes não funcionam, ele atende mas fica fazendo um zunido
>>>>>> .... parecido com de fax só que mais fino ....
>>>>>>
>>>>>> Emanuel
>>>>>>
>>>>>>
>>>>>> 2008/11/11 Tiago Durante <tiagodurante em gmail.com>
>>>>>>
>>>>>>
>>>>>>
>>>>>>> Oi,
>>>>>>>
>>>>>>> Sim, manda seu interrups... Just in case...
>>>>>>>
>>>>>>>
>>>>>>> # cat /proc/interrupts
>>>>>>>
>>>>>>>
>>>>>>> Chamadas entrantes funcionam?
>>>>>>>
>>>>>>>
>>>>>>> []s
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 2008/11/11 Emanuel dos Reis Rodrigues <emanueldosreis em gmail.com>:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Tiago Durante wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> Emanuel,
>>>>>>>>>
>>>>>>>>> Pelo dialplan deveria sim estar fazendo a ligacão.
>>>>>>>>> Ele não chega nem a tocar o telefone que vc chama?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> []s
>>>>>>>>>
>>>>>>>>> 2008/11/11 Emanuel dos Reis Rodrigues <emanueldosreis em gmail.com>:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> Helius Ferreira wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> Oi!! Emanuel...
>>>>>>>>>>> Você pode tentar fazer o boot com algum Asterisk LiveCD (pelo
>>>>>>>>>>> exemplo:
>>>>>>>>>>> Ubunterisk, Celliax ou qualquer outro), para testar seu hardware e
>>>>>>>>>>> software.
>>>>>>>>>>> Ou como qualquer outra dificuldade nas configurações ..do dahdi..
>>>>>>>>>>>
>>>>>>>>>>> Boa, sorte...viu!!
>>>>>>>>>>> Helius
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>> então, gostaria de saber se pelo meu extensions.conf, qual o
>>>>>>>>>> comportamento ? por exemplo era para discar normalmente ?
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> Emanuel dos Reis Rodrigues
>>>>>>>>>> Senior Level Linux Professional (LPIC-3)
>>>>>>>>>> emanueldosreis(No*SpAm)gmail.com
>>>>>>>>>> +55 95 8112-9628
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> 2 a 4 de Dezembro - IPComm 2008
>>>>>>>>>> Com presença de engenheiros da Digium e Jon "maddog" Hall.
>>>>>>>>>> http://www.ipcomm2008.com.br
>>>>>>>>>>
>>>>>>>>>> Compre uma camiseta da AsteriskBrasil.org!
>>>>>>>>>> http://www.voipmania.com.br
>>>>>>>>>>
>>>>>>>>>> Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro
>>>>>>>>>> na rede Freenode.net: #asterisk-br
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Lista de discussões AsteriskBrasil.org
>>>>>>>>>> AsteriskBrasil em listas.asteriskbrasil.org
>>>>>>>>>> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>> não cara não toca, no console aparece isso:
>>>>>>>>
>>>>>>>> Using SIP RTP CoS mark 5
>>>>>>>> Executing [0numerodiscado]@default:1
>>>>>>>> Dial("SIP/101-082452f8","DAHDI/1/numerodiscado") in new stack
>>>>>>>> Called 1/numerodiscado
>>>>>>>> DAHDI/1-1 answered SIP/101-082452f8
>>>>>>>>
>>>>>>>>
>>>>>>>> Fica neste ponto até que eu desligo, não sai disso.
>>>>>>>>
>>>>>>>> verifico no console e indica que o ramal está usando o canal ... no
>>>>>>>> entanto fica lá ....
>>>>>>>>
>>>>>>>> tem mais alguma informação que posso postar na lista ?
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Emanuel dos Reis Rodrigues
>>>>>>>> Senior Level Linux Professional (LPIC-3)
>>>>>>>> emanueldosreis(No*SpAm)gmail.com
>>>>>>>> +55 95 8112-9628
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> 2 a 4 de Dezembro - IPComm 2008
>>>>>>>> Com presença de engenheiros da Digium e Jon "maddog" Hall.
>>>>>>>> http://www.ipcomm2008.com.br
>>>>>>>>
>>>>>>>> Compre uma camiseta da AsteriskBrasil.org!
>>>>>>>> http://www.voipmania.com.br
>>>>>>>>
>>>>>>>> Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro
>>>>>>>> na rede Freenode.net: #asterisk-br
>>>>>>>> _______________________________________________
>>>>>>>> Lista de discussões AsteriskBrasil.org
>>>>>>>> AsteriskBrasil em listas.asteriskbrasil.org
>>>>>>>> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>> --
>>>>>>> Tiago Durante
>>>>>>>
>>>>>>> ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,
>>>>>>> Perseverance is the hard work you do after you
>>>>>>> get tired of doing the hard work you already did.
>>>>>>> -- Newt Gingrich
>>>>>>> _______________________________________________
>>>>>>> 2 a 4 de Dezembro - IPComm 2008
>>>>>>> Com presença de engenheiros da Digium e Jon "maddog" Hall.
>>>>>>> http://www.ipcomm2008.com.br
>>>>>>>
>>>>>>> Compre uma camiseta da AsteriskBrasil.org!
>>>>>>> http://www.voipmania.com.br
>>>>>>>
>>>>>>> Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na
>>>>>>> rede Freenode.net: #asterisk-br
>>>>>>> _______________________________________________
>>>>>>> Lista de discussões AsteriskBrasil.org
>>>>>>> AsteriskBrasil em listas.asteriskbrasil.org
>>>>>>> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> --
>>>>>> Emanuel dos Reis Rodrigues
>>>>>> Linux Professional Certified
>>>>>>
>>>>>> _______________________________________________
>>>>>> 2 a 4 de Dezembro - IPComm 2008
>>>>>> Com presença de engenheiros da Digium e Jon "maddog" Hall.
>>>>>> http://www.ipcomm2008.com.br
>>>>>>
>>>>>> Compre uma camiseta da AsteriskBrasil.org!
>>>>>> http://www.voipmania.com.br
>>>>>>
>>>>>> Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na
>>>>>> rede Freenode.net: #asterisk-br
>>>>>> _______________________________________________
>>>>>> Lista de discussões AsteriskBrasil.org
>>>>>> AsteriskBrasil em listas.asteriskbrasil.org
>>>>>> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>> mais esta á uma clone ( Intel Ambiente MD3200), na verdade já tentei 3,
>>>> estou comprando uma x100p original ... mas esta placa funciona ...
>>>> estou usando para testar ... mas irei comprar uma tdm410p com 4 modulos
>>>> FXO .... mas gostaria de me certificar de algumas funcionalidades ....
>>>> testes etc ....
>>>>
>>>> tem algun debug que pode ajudar ?
>>>>
>>>>
>>>> --
>>>> Emanuel dos Reis Rodrigues
>>>> Senior Level Linux Professional (LPIC-3)
>>>> emanueldosreis(No*SpAm)gmail.com
>>>> +55 95 8112-9628
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 2 a 4 de Dezembro - IPComm 2008
>>>> Com presença de engenheiros da Digium e Jon "maddog" Hall.
>>>> http://www.ipcomm2008.com.br
>>>>
>>>> Compre uma camiseta da AsteriskBrasil.org!
>>>> http://www.voipmania.com.br
>>>>
>>>> Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na rede Freenode.net: #asterisk-br
>>>> _______________________________________________
>>>> Lista de discussões AsteriskBrasil.org
>>>> AsteriskBrasil em listas.asteriskbrasil.org
>>>> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>>>>
>>>>
>>>>
>>>
>>>
>>>
>> asterisk:/etc/asterisk# dahdi_cfg -vvvvv
>> DAHDI Tools Version - SVN-trunk-r5160
>>
>> DAHDI Version: SVN-trunk-r5207
>> Echo Canceller(s): MG2
>> Configuration
>> ======================
>>
>>
>> Channel map:
>>
>> Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
>>
>> 1 channels to configure.
>>
>> Setting echocan for channel 1 to mg2
>>
>>
>>
>>
>> dahdi: Telephony Interface Registered on major 196
>> dahdi: Version: SVN-trunk-r5207
>> ACPI: PCI Interrupt 0000:00:08.0[A] -> GSI 16 (level, low) -> IRQ 193
>> Found a Wildcard FXO: Generic Clone
>> INFO-xpp: revision trunk-r6056 MAX_XPDS=64 (8*8)
>> INFO-xpp: FEATURE: without BRISTUFF support
>> INFO-xpp: FEATURE: with PROTOCOL_DEBUG
>> INFO-xpp: FEATURE: with sync_tick() from DAHDI
>> INFO-xpp_usb: revision trunk-r6056
>> usbcore: registered new driver xpp_usb
>> dahdi_echocan_mg2: Registered echo canceler 'MG2'
>> dahdi: Registered tone zone 0 (United States / North America)
>>
>>
>> tudo certo ....
>>
>>
>>
>> --
>> Emanuel dos Reis Rodrigues
>> Senior Level Linux Professional (LPIC-3)
>> emanueldosreis(No*SpAm)gmail.com
>> +55 95 8112-9628
>>
>>
>>
>>
>> _______________________________________________
>> 2 a 4 de Dezembro - IPComm 2008
>> Com presença de engenheiros da Digium e Jon "maddog" Hall.
>> http://www.ipcomm2008.com.br
>>
>> Compre uma camiseta da AsteriskBrasil.org!
>> http://www.voipmania.com.br
>>
>> Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na rede Freenode.net: #asterisk-br
>> _______________________________________________
>> Lista de discussões AsteriskBrasil.org
>> AsteriskBrasil em listas.asteriskbrasil.org
>> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>>
>>
>
>
>
>
fiz o que pediu e aumentei o nivel de log .... veja:
quando eu ligo para o modem ...
Starting simple switch on 'DAHDI/1-1'
[Nov 11 17:00:33] ERROR[2896]: callerid.c:562 callerid_feed: No start
bit found in fsk data.
[Nov 11 17:00:33] WARNING[2896]: chan_dahdi.c:7154 ss_thread: CallerID
feed failed: Success
[Nov 11 17:00:33] WARNING[2896]: chan_dahdi.c:7258 ss_thread: CallerID
returned with error on channel 'DAHDI/1-1'
-- Executing [s em default:1] Answer("DAHDI/1-1", "") in new stack
-- Executing [s em default:2] Dial("DAHDI/1-1", "SIP/101&SIP/102") in
new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called 102
-- SIP/101-08233e48 is ringing
-- SIP/102-08235440 is ringing
No entanto não toca no ramai SIP ....
Tento discar, disco 0meunumero, fica mudo ....
Quando disca
--
Emanuel dos Reis Rodrigues
Senior Level Linux Professional (LPIC-3)
emanueldosreis(No*SpAm)gmail.com
+55 95 8112-9628
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