[AsteriskBrasil] Ura não reconhece digitos.
Silvinho Barros
silvinho.b4rros em gmail.com
Terça Novembro 20 12:18:04 BRST 2012
Jorge,
O logger.conf está assim:
[logfiles]
;
; Format is "filename" and then "levels" of debugging to be included:
; debug
; notice
; warning
; error
; verbose
;
; Special filename "console" represents the system console
;
;debug => debug
; The DTMF log is very handy if you have issues with IVR's
;dtmf => dtmf
;console => notice,warning,error
console => notice,warning,error,debug,dtmf
;messages => notice,warning,error
full => notice,warning,error,debug,verbose,dtmf
;syslog keyword : This special keyword logs to syslog facility
;
;syslog.local0 => notice,warning,error
Log de uma ligação:
-- Accepting call from '8121386100' to '8811' on channel 0/21, span 2
-- Executing [8811 em from-pstn:1] Set("DAHDI/i2/8121386100-45",
"__FROM_DID=8811") in new stack
-- Executing [8811 em from-pstn:2] Gosub("DAHDI/i2/8121386100-45",
"app-blacklist-check,s,1") in new stack
-- Executing [s em app-blacklist-check:1] GotoIf("DAHDI/i2/8121386100-45",
"0?blacklisted") in new stack
-- Executing [s em app-blacklist-check:2] Set("DAHDI/i2/8121386100-45",
"CALLED_BLACKLIST=1") in new stack
-- Executing [s em app-blacklist-check:3] Return("DAHDI/i2/8121386100-45",
"") in new stack
-- Executing [8811 em from-pstn:3] ExecIf("DAHDI/i2/8121386100-45", "1
?Set(CALLERID(name)=8121386100)") in new stack
-- Executing [8811 em from-pstn:4] Set("DAHDI/i2/8121386100-45",
"__CALLINGPRES_SV=allowed") in new stack
-- Executing [8811 em from-pstn:5] Set("DAHDI/i2/8121386100-45",
"CALLERPRES()=allowed_not_screened") in new stack
-- Executing [8811 em from-pstn:6] Goto("DAHDI/i2/8121386100-45",
"ext-trunk,3,1") in new stack
-- Goto (ext-trunk,3,1)
-- Executing [3 em ext-trunk:1] Set("DAHDI/i2/8121386100-45",
"TDIAL_STRING=SIP/ura") in new stack
-- Executing [3 em ext-trunk:2] Set("DAHDI/i2/8121386100-45",
"DIAL_TRUNK=3") in new stack
-- Executing [3 em ext-trunk:3] Goto("DAHDI/i2/8121386100-45",
"ext-trunk,tcustom,1") in new stack
-- Goto (ext-trunk,tcustom,1)
-- Executing [tcustom em ext-trunk:1] Set("DAHDI/i2/8121386100-45",
"OUTBOUND_GROUP=OUT_3") in new stack
-- Executing [tcustom em ext-trunk:2] GotoIf("DAHDI/i2/8121386100-45",
"1?nomax") in new stack
-- Goto (ext-trunk,tcustom,4)
-- Executing [tcustom em ext-trunk:4] ExecIf("DAHDI/i2/8121386100-45",
"1?Set(CALLERPRES()=allowed)") in new stack
-- Executing [tcustom em ext-trunk:5] Set("DAHDI/i2/8121386100-45",
"DIAL_NUMBER=8811") in new stack
-- Executing [tcustom em ext-trunk:6] GosubIf("DAHDI/i2/8121386100-45",
"0?sub-flp-3,s,1") in new stack
-- Executing [tcustom em ext-trunk:7] Set("DAHDI/i2/8121386100-45",
"OUTNUM=8811") in new stack
-- Executing [tcustom em ext-trunk:8] Set("DAHDI/i2/8121386100-45",
"CALLERID(number)=8121386100") in new stack
-- Executing [tcustom em ext-trunk:9] Set("DAHDI/i2/8121386100-45",
"CALLERID(name)=8121386100") in new stack
-- Executing [tcustom em ext-trunk:10] Dial("DAHDI/i2/8121386100-45",
"SIP/ura,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/ura
-- SIP/74.3.210.188-00000d5e is ringing
-- SIP/74.3.210.188-00000d5e answered DAHDI/i2/8121386100-45
DEBUG:
É muita informação entrando na CLI coloquei o DEBUG 1:
[Nov 20 12:05:15] DEBUG[4320]: chan_sip.c:7769 sip_alloc: Allocating new
SIP dialog for 7659af540d953709360ba9d719fc993d em 127.0.0.1:5060 - OPTIONS
(No RTP)
[Nov 20 12:05:15] DEBUG[4320]: chan_sip.c:3073 initialize_initreq:
Initializing initreq for method OPTIONS - callid
3c963be87d921ee545af743d3c6729c7 em 192.168.12.10:5060
[Nov 20 12:05:15] DEBUG[23078]: res_rtp_asterisk.c:1791 ast_rtcp_read: Got
RTCP report of 176 bytes
[Nov 20 12:05:17] DEBUG[23549]: chan_dahdi.c:7716 dahdi_handle_dtmf: Begin
DTMF digit: 0x31 '1' on DAHDI/i2/8121386100-45
[Nov 20 12:05:17] DTMF[23549]: channel.c:4066 __ast_read: DTMF begin '1'
received on DAHDI/i2/8121386100-45
[Nov 20 12:05:17] DTMF[23549]: channel.c:4076 __ast_read: DTMF begin
passthrough '1' on DAHDI/i2/8121386100-45
[Nov 20 12:05:17] DEBUG[23549]: chan_dahdi.c:7716 dahdi_handle_dtmf: End
DTMF digit: 0x31 '1' on DAHDI/i2/8121386100-45
[Nov 20 12:05:17] DTMF[23549]: channel.c:3981 __ast_read: DTMF end '1'
received on DAHDI/i2/8121386100-45, duration 38 ms
[Nov 20 12:05:17] DTMF[23549]: channel.c:4021 __ast_read: DTMF end accepted
with begin '1' on DAHDI/i2/8121386100-45
[Nov 20 12:05:17] DTMF[23549]: channel.c:4036 __ast_read: DTMF end '1'
detected to have actual duration 79 on the wire, emulation will be
triggered on DAHDI/i2/8121386100-45
[Nov 20 12:05:17] DTMF[23549]: channel.c:4043 __ast_read: DTMF end '1' has
duration 79 but want minimum 80, emulating on DAHDI/i2/8121386100-45
[Nov 20 12:05:17] DTMF[23549]: channel.c:4143 __ast_read: DTMF end
emulation of '1' queued on DAHDI/i2/8121386100-45
[Nov 20 12:05:17] DEBUG[23549]: res_rtp_asterisk.c:1791 ast_rtcp_read: Got
RTCP report of 60 bytes
[Nov 20 12:05:19] DEBUG[23078]: res_rtp_asterisk.c:1791 ast_rtcp_read: Got
RTCP report of 176 bytes
[Nov 20 12:05:20] DEBUG[23549]: res_rtp_asterisk.c:1791 ast_rtcp_read: Got
RTCP report of 60 bytes
[Nov 20 12:05:22] DEBUG[23078]: res_rtp_asterisk.c:1791 ast_rtcp_read: Got
RTCP report of 176 bytes
[Nov 20 12:05:23] DEBUG[23549]: chan_dahdi.c:7716 dahdi_handle_dtmf: Begin
DTMF digit: 0x30 '0' on DAHDI/i2/8121386100-45
[Nov 20 12:05:23] DTMF[23549]: channel.c:4066 __ast_read: DTMF begin '0'
received on DAHDI/i2/8121386100-45
[Nov 20 12:05:23] DTMF[23549]: channel.c:4076 __ast_read: DTMF begin
passthrough '0' on DAHDI/i2/8121386100-45
[Nov 20 12:05:23] DEBUG[23549]: chan_dahdi.c:7716 dahdi_handle_dtmf: End
DTMF digit: 0x30 '0' on DAHDI/i2/8121386100-45
[Nov 20 12:05:23] DTMF[23549]: channel.c:3981 __ast_read: DTMF end '0'
received on DAHDI/i2/8121386100-45, duration 51 ms
[Nov 20 12:05:23] DTMF[23549]: channel.c:4021 __ast_read: DTMF end accepted
with begin '0' on DAHDI/i2/8121386100-45
[Nov 20 12:05:23] DTMF[23549]: channel.c:4043 __ast_read: DTMF end '0' has
duration 51 but want minimum 80, emulating on DAHDI/i2/8121386100-45
[Nov 20 12:05:23] DTMF[23549]: channel.c:4143 __ast_read: DTMF end
emulation of '0' queued on DAHDI/i2/8121386100-45
[Nov 20 12:05:25] DEBUG[23078]: res_rtp_asterisk.c:1791 ast_rtcp_read: Got
RTCP report of 176 bytes
[Nov 20 12:05:26] DEBUG[23549]: res_rtp_asterisk.c:1791 ast_rtcp_read: Got
RTCP report of 60 bytes
[Nov 20 12:05:27] DEBUG[4251]: chan_dahdi.c:4863 dahdi_enable_ec: Echo
cancellation already on
[Nov 20 12:05:27] DEBUG[23549]: channel.c:7113 ast_generic_bridge: Got a
FRAME_CONTROL (3) frame on channel DAHDI/i2/8121386100-45
[Nov 20 12:05:27] DEBUG[23549]: chan_dahdi.c:9366 dahdi_indicate: Requested
indication 20 on channel DAHDI/i2/8121386100-45
[Nov 20 12:05:27] DEBUG[23549]: channel.c:7531 ast_channel_bridge: Bridge
stops bridging channels DAHDI/i2/8121386100-45 and SIP/74.3.210.188-00000d5e
[Nov 20 12:05:27] DEBUG[23549]: channel.c:4495 ast_indicate_data: Driver
for channel 'SIP/74.3.210.188-00000d5e' does not support indication 3,
emulating it
[Nov 20 12:05:27] DEBUG[23549]: channel.c:5150 set_format: Set channel
SIP/74.3.210.188-00000d5e to write format slin
[Nov 20 12:05:27] DEBUG[23549]: channel.c:3503 ast_settimeout: Scheduling
timer at (50 requested / 50 actual) timer ticks per second
[Nov 20 12:05:27] DEBUG[23549]: chan_dahdi.c:9366 dahdi_indicate: Requested
indication 20 on channel DAHDI/i2/8121386100-45
[Nov 20 12:05:27] DEBUG[23549]: channel.c:7113 ast_generic_bridge: Got a
FRAME_CONTROL (14) frame on channel DAHDI/i2/8121386100-45
[Nov 20 12:05:27] DEBUG[23549]: chan_dahdi.c:9366 dahdi_indicate: Requested
indication 20 on channel DAHDI/i2/8121386100-45
[Nov 20 12:05:27] DEBUG[23549]: channel.c:7531 ast_channel_bridge: Bridge
stops bridging channels DAHDI/i2/8121386100-45 and SIP/74.3.210.188-00000d5e
[Nov 20 12:05:27] DEBUG[23549]: chan_dahdi.c:9366 dahdi_indicate: Requested
indication 20 on channel DAHDI/i2/8121386100-45
[Nov 20 12:05:27] DEBUG[23549]: channel.c:5150 set_format: Set channel
SIP/74.3.210.188-00000d5e to write format alaw
[Nov 20 12:05:27] DEBUG[23549]: channel.c:3503 ast_settimeout: Scheduling
timer at (0 requested / 0 actual) timer ticks per second
Em 20 de novembro de 2012 10:37, Jorge Silveira <jlrs19833 em gmail.com>escreveu:
> Bom dia;
>
> Estou acompanhando este assunto e gostaria de sugerir que fosse habilitado
> o log full do asterisk. e encaminhado o log da chamada para verificar se há
> algo errado.
>
> Encaminhe tb um log com o debug. às vezes é mais fácil visualizar um erro
> assim do que
>
> Em 20 de novembro de 2012 10:03, Silvinho Barros <
> silvinho.b4rros em gmail.com> escreveu:
>
> Não, temos apenas o asterisk(elastix) e no mesmo temos 3 troncos da vono e
>> um tronco e1 isdn da embratel.
>> No servidor a placa 2E1 da aligera(ap 402).
>> Existe um balun fazendo a conversão para RJ45.
>>
>> Atenciosamente.
>>
>> Em 20 de novembro de 2012 08:48, Rafael dos Santos Saraiva <
>> rafaelsnsa em gmail.com> escreveu:
>>
>>> O Asterisk está entre a operadora e um PABX?
>>>
>>> Att,
>>> Rafael Saraiva
>>> <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
>>>
>>>
>>>
>>> Em 20 de novembro de 2012 09:36, Silvinho Barros <
>>> silvinho.b4rros em gmail.com> escreveu:
>>>
>>> Oi rafael,
>>>>
>>>> Criei um ramal SIP em outro servidor asterisk e consegui interagir com
>>>> a URA, direto de uma linha PSNT ligada ao telefone também consigo, de um
>>>> outro servidor com placa tdm410p(4fxo) tambem consigo, apenas quando passa
>>>> pelo E1(A,ligera ap402) que não consego interagir com a URA(Cai na URA a
>>>> ligação é limpa mais não interagi).
>>>>
>>>> Um outro teste que fiz foi ligar para uma outra URA passando pelo E1,
>>>> funcionou.
>>>>
>>>> Estou realmente sem saber oque fazer.
>>>>
>>>> System.conf
>>>>
>>>> # Autogenerated by /usr/sbin/dahdi_genconf on Tue Nov 13 13:44:26 2012
>>>> # If you edit this file and execute /usr/sbin/dahdi_genconf again,
>>>> # your manual changes will be LOST.
>>>> # Dahdi Configuration File
>>>> #
>>>> # This file is parsed by the Dahdi Configurator, dahdi_cfg
>>>> #
>>>> # Span 1: AP402/0/1 "AP402 Card 1 Span 1" (MASTER)
>>>> span=1,1,0,ccs,hdb3,crc4
>>>> # termtype: te
>>>> bchan=1-15,17-31
>>>> dchan=16
>>>> #echocanceller=oslec,1-15,17-31
>>>>
>>>> # Span 2: AP402/0/2 "AP402 Card 1 Span 2"
>>>> span=2,2,0,ccs,hdb3,crc4
>>>> # termtype: te
>>>> bchan=32-46,48-62
>>>> dchan=47
>>>> #echocanceller=oslec,32-46,48-62
>>>>
>>>> # Global data
>>>>
>>>> ================
>>>>
>>>> chan_dahdi.conf
>>>>
>>>> [trunkgroups]
>>>> [channels]
>>>> language=br
>>>>
>>>> usecallerid=yes
>>>> callwaiting=yes
>>>> usecallingpres=yes
>>>> callwaitingcallerid=yes
>>>> threewaycalling=yes
>>>> transfer=yes
>>>> canpark=yes
>>>> cancallforward=yes
>>>> callreturn=yes
>>>> echocancel=yes
>>>> echocancelwhenbridged=no
>>>> ;Configuraç da primeira interface como tronco ISDN oriundo da operadora
>>>> switchtype=euroisdn
>>>> ;informa o nome do contexto usado em extensions.conf
>>>> group=1
>>>> signalling=pri_cpe
>>>> context=from-pstn
>>>> channel=>1-15,17-31
>>>>
>>>> signalling=pri_cpe
>>>> context=from-pstn
>>>> group=0
>>>> channel=>32-46,48-62
>>>>
>>>> ========================================
>>>>
>>>> dahdi_channels.conf
>>>>
>>>> ; Autogenerated by /usr/sbin/dahdi_genconf on Tue Nov 13 13:44:26 2012
>>>> ; If you edit this file and execute /usr/sbin/dahdi_genconf again,
>>>> ; your manual changes will be LOST.
>>>> ; Dahdi Channels Configurations (chan_dahdi.conf)
>>>> ;
>>>> ; This is not intended to be a complete chan_dahdi.conf. Rather, it is
>>>> intended
>>>> ; to be #include-d by /etc/chan_dahdi.conf that will include the global
>>>> settings
>>>> ;
>>>>
>>>> ; Span 1: AP402/0/1 "AP402 Card 1 Span 1" (MASTER)
>>>> group=0,11
>>>> context=from-pstn
>>>> switchtype = euroisdn
>>>> signalling = pri_cpe
>>>> channel => 1-15,17-31
>>>> context = default
>>>> group = 63
>>>>
>>>> ; Span 2: AP402/0/2 "AP402 Card 1 Span 2"
>>>> group=0,12
>>>> context=from-pstn
>>>> switchtype = euroisdn
>>>> signalling = pri_cpe
>>>> channel => 32-46,48-62
>>>> context = default
>>>> group = 63
>>>>
>>>>
>>>> Atenciosamente.
>>>>
>>>>
>>>>
>>>> Em 16 de novembro de 2012 11:49, Rafael dos Santos Saraiva <
>>>> rafaelsnsa em gmail.com> escreveu:
>>>>
>>>>> Bom, creio que seja alguma coisa no E1, bom ver se crc, clock está ok.
>>>>> Tenta logar com um ramal sip pra confirmar se o problema é no e1 ou no
>>>>> asterisk.
>>>>>
>>>>> Att,
>>>>> Rafael Saraiva
>>>>> <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
>>>>>
>>>>>
>>>>>
>>>>> Em 16 de novembro de 2012 12:41, Silvinho Barros <
>>>>> silvinho.b4rros em gmail.com> escreveu:
>>>>>
>>>>> Fiz a alteração restartei o dahdi e asterisk mas nao resolveu.
>>>>>>
>>>>>> De qualquer forma obrigado pela ajuda!
>>>>>>
>>>>>> Tem mais alguma sugestão ?
>>>>>>
>>>>>>
>>>>>> Em 16 de novembro de 2012 11:17, Rafael dos Santos Saraiva <
>>>>>> rafaelsnsa em gmail.com> escreveu:
>>>>>>
>>>>>>>
>>>>>>> Algo referente a duração do DTMF que não é suficiente para o
>>>>>>> Asterisk reconhecer.
>>>>>>> Não sei se resolveria, mas tente adicionar no chan_dahdi.conf:
>>>>>>>
>>>>>>> relaxdtmf=yes
>>>>>>>
>>>>>>>
>>>>>>> Att,
>>>>>>> Rafael Saraiva
>>>>>>> <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Em 16 de novembro de 2012 12:01, Silvinho Barros <
>>>>>>> silvinho.b4rros em gmail.com> escreveu:
>>>>>>>
>>>>>>> A placa e uma aligera ap402.
>>>>>>>> Foi necessario instalar o driver pois so com dahdi o asterisk
>>>>>>>> reconhecia mas nao funcionava.
>>>>>>>>
>>>>>>>> Segue log dtmf
>>>>>>>> [Nov 16 11:59:50] NOTICE[24692]: chan_sip.c:20788
>>>>>>>> handle_response_peerpoke: Peer '2003' is now Reachable. (3ms / 2000ms)
>>>>>>>> [Nov 16 11:59:54] DTMF[13770]: channel.c:4066 __ast_read: DTMF
>>>>>>>> begin '1' received on DAHDI/i2/8121386100-25
>>>>>>>> [Nov 16 11:59:54] DTMF[13770]: channel.c:4076 __ast_read: DTMF
>>>>>>>> begin passthrough '1' on DAHDI/i2/8121386100-25
>>>>>>>> [Nov 16 11:59:54] DTMF[13770]: channel.c:3981 __ast_read: DTMF end
>>>>>>>> '1' received on DAHDI/i2/8121386100-25, duration 12 ms
>>>>>>>> [Nov 16 11:59:54] DTMF[13770]: channel.c:4021 __ast_read: DTMF end
>>>>>>>> accepted with begin '1' on DAHDI/i2/8121386100-25
>>>>>>>> [Nov 16 11:59:54] DTMF[13770]: channel.c:4036 __ast_read: DTMF end
>>>>>>>> '1' detected to have actual duration 60 on the wire, emulation will be
>>>>>>>> triggered on DAHDI/i2/8121386100-25
>>>>>>>> [Nov 16 11:59:54] DTMF[13770]: channel.c:4043 __ast_read: DTMF end
>>>>>>>> '1' has duration 60 but want minimum 80, emulating on DAHDI/i2/8121386100-25
>>>>>>>> [Nov 16 11:59:54] DTMF[13770]: channel.c:4143 __ast_read: DTMF end
>>>>>>>> emulation of '1' queued on DAHDI/i2/8121386100-25
>>>>>>>>
>>>>>>>> Att, SIlvio de Barros.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Em 16 de novembro de 2012 09:23, Rafael dos Santos Saraiva <
>>>>>>>> rafaelsnsa em gmail.com> escreveu:
>>>>>>>>
>>>>>>>>> Faz o seguinte pra ver se esse dtmf ta chegando no Asterisk. No
>>>>>>>>> arquivo logger.conf, adiciona na linha que começa com "console =>" o
>>>>>>>>> parametro dtmf. Assim "console => notice,warning,error,dtmf"
>>>>>>>>>
>>>>>>>>> A interface E1 usa DAHDI?
>>>>>>>>>
>>>>>>>>> Att,
>>>>>>>>> Rafael Saraiva
>>>>>>>>> <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Em 16 de novembro de 2012 10:11, Silvinho Barros <
>>>>>>>>> silvinho.b4rros em gmail.com> escreveu:
>>>>>>>>>
>>>>>>>>> Rafael,
>>>>>>>>>>
>>>>>>>>>> A ligação entra pelo E1 que esta conectado diretamente ao
>>>>>>>>>> asterisk(nao tem central analogica), dai o contexto de entrada redireciona
>>>>>>>>>> o mesmo para essa URA.
>>>>>>>>>>
>>>>>>>>>> Em 14 de novembro de 2012 17:24, Rafael dos Santos Saraiva <
>>>>>>>>>> rafaelsnsa em gmail.com> escreveu:
>>>>>>>>>>
>>>>>>>>>>> Como é que está atualmente, a ligação entra pelo E1 e vai direto
>>>>>>>>>>> pra URA? Esse E1 é da operadora ou um PABX? O DTMF não funciona a partir de
>>>>>>>>>>> ligações vindas do E1 ou dos ramais SIP?
>>>>>>>>>>>
>>>>>>>>>>> Att,
>>>>>>>>>>> Rafael Saraiva
>>>>>>>>>>> <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Em 14 de novembro de 2012 17:35, Silvinho Barros <
>>>>>>>>>>> silvinho.b4rros em gmail.com> escreveu:
>>>>>>>>>>>
>>>>>>>>>>>> Boa tarde Pessoal.
>>>>>>>>>>>>
>>>>>>>>>>>> Depois de solucionar o problema do E1 no asterisk, apareceu
>>>>>>>>>>>> o próximo..
>>>>>>>>>>>> Fiz um redirecionamento de tudo que entra pelo E1 para um SIP
>>>>>>>>>>>> uqe é um URA, até ai beleza ele está redirecionando normal, esculto a URA
>>>>>>>>>>>> perfeitamente mas ele não reconhece nenhum dos digitos das opções.
>>>>>>>>>>>>
>>>>>>>>>>>> Alguém já passou por isso? estão me falando em dtmf.
>>>>>>>>>>>>
>>>>>>>>>>>> Alguma dica de por onde posso começar a resolver esse problema ?
>>>>>>>>>>>>
>>>>>>>>>>>> Grato pela atenção de todos.
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> KHOMP Inovação: External Board Series
>>>>>>>>>>>> Módulos de 1/2 rack e 1U para todas as interfaces e soluções
>>>>>>>>>>>> Asterisk e FreeSWITCH.
>>>>>>>>>>>> Tenha a External Series Experience na sua aplicação. Visite
>>>>>>>>>>>> www.khomp.com
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
>>>>>>>>>>>> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
>>>>>>>>>>>> Centro Treinamento - Curso de PABX IP - Asterisk - Site
>>>>>>>>>>>> www.digivoice.com.br
>>>>>>>>>>>> ________
>>>>>>>>>>>> YEALINK: Telefones IP e VídeoPhones IP com o melhor
>>>>>>>>>>>> custo/benefício do mercado.
>>>>>>>>>>>> email: yealink em commlogik.com.br | www.commlogik.com.br | (11)
>>>>>>>>>>>> 5503-1011
>>>>>>>>>>>> ______________________________________________
>>>>>>>>>>>> Para remover seu email desta lista, basta enviar um email em
>>>>>>>>>>>> branco para
>>>>>>>>>>>> asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> KHOMP Inovação: External Board Series
>>>>>>>>>>> Módulos de 1/2 rack e 1U para todas as interfaces e soluções
>>>>>>>>>>> Asterisk e FreeSWITCH.
>>>>>>>>>>> Tenha a External Series Experience na sua aplicação. Visite
>>>>>>>>>>> www.khomp.com
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
>>>>>>>>>>> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
>>>>>>>>>>> Centro Treinamento - Curso de PABX IP - Asterisk - Site
>>>>>>>>>>> www.digivoice.com.br
>>>>>>>>>>> ________
>>>>>>>>>>> YEALINK: Telefones IP e VídeoPhones IP com o melhor
>>>>>>>>>>> custo/benefício do mercado.
>>>>>>>>>>> email: yealink em commlogik.com.br | www.commlogik.com.br | (11)
>>>>>>>>>>> 5503-1011
>>>>>>>>>>> ______________________________________________
>>>>>>>>>>> Para remover seu email desta lista, basta enviar um email em
>>>>>>>>>>> branco para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> KHOMP Inovação: External Board Series
>>>>>>>>>> Módulos de 1/2 rack e 1U para todas as interfaces e soluções
>>>>>>>>>> Asterisk e FreeSWITCH.
>>>>>>>>>> Tenha a External Series Experience na sua aplicação. Visite
>>>>>>>>>> www.khomp.com
>>>>>>>>>> _______________________________________________
>>>>>>>>>> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
>>>>>>>>>> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
>>>>>>>>>> Centro Treinamento - Curso de PABX IP - Asterisk - Site
>>>>>>>>>> www.digivoice.com.br
>>>>>>>>>> ________
>>>>>>>>>> YEALINK: Telefones IP e VídeoPhones IP com o melhor
>>>>>>>>>> custo/benefício do mercado.
>>>>>>>>>> email: yealink em commlogik.com.br | www.commlogik.com.br | (11)
>>>>>>>>>> 5503-1011
>>>>>>>>>> ______________________________________________
>>>>>>>>>> Para remover seu email desta lista, basta enviar um email em
>>>>>>>>>> branco para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> KHOMP Inovação: External Board Series
>>>>>>>>> Módulos de 1/2 rack e 1U para todas as interfaces e soluções
>>>>>>>>> Asterisk e FreeSWITCH.
>>>>>>>>> Tenha a External Series Experience na sua aplicação. Visite
>>>>>>>>> www.khomp.com
>>>>>>>>> _______________________________________________
>>>>>>>>> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
>>>>>>>>> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
>>>>>>>>> Centro Treinamento - Curso de PABX IP - Asterisk - Site
>>>>>>>>> www.digivoice.com.br
>>>>>>>>> ________
>>>>>>>>> YEALINK: Telefones IP e VídeoPhones IP com o melhor
>>>>>>>>> custo/benefício do mercado.
>>>>>>>>> email: yealink em commlogik.com.br | www.commlogik.com.br | (11)
>>>>>>>>> 5503-1011
>>>>>>>>> ______________________________________________
>>>>>>>>> Para remover seu email desta lista, basta enviar um email em
>>>>>>>>> branco para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> KHOMP Inovação: External Board Series
>>>>>>>> Módulos de 1/2 rack e 1U para todas as interfaces e soluções
>>>>>>>> Asterisk e FreeSWITCH.
>>>>>>>> Tenha a External Series Experience na sua aplicação. Visite
>>>>>>>> www.khomp.com
>>>>>>>> _______________________________________________
>>>>>>>> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
>>>>>>>> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
>>>>>>>> Centro Treinamento - Curso de PABX IP - Asterisk - Site
>>>>>>>> www.digivoice.com.br
>>>>>>>> ________
>>>>>>>> YEALINK: Telefones IP e VídeoPhones IP com o melhor custo/benefício
>>>>>>>> do mercado.
>>>>>>>> email: yealink em commlogik.com.br | www.commlogik.com.br | (11)
>>>>>>>> 5503-1011
>>>>>>>> ______________________________________________
>>>>>>>> Para remover seu email desta lista, basta enviar um email em branco
>>>>>>>> para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> KHOMP Inovação: External Board Series
>>>>>>> Módulos de 1/2 rack e 1U para todas as interfaces e soluções
>>>>>>> Asterisk e FreeSWITCH.
>>>>>>> Tenha a External Series Experience na sua aplicação. Visite
>>>>>>> www.khomp.com
>>>>>>> _______________________________________________
>>>>>>> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
>>>>>>> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
>>>>>>> Centro Treinamento - Curso de PABX IP - Asterisk - Site
>>>>>>> www.digivoice.com.br
>>>>>>> ________
>>>>>>> YEALINK: Telefones IP e VídeoPhones IP com o melhor custo/benefício
>>>>>>> do mercado.
>>>>>>> email: yealink em commlogik.com.br | www.commlogik.com.br | (11)
>>>>>>> 5503-1011
>>>>>>> ______________________________________________
>>>>>>> Para remover seu email desta lista, basta enviar um email em branco
>>>>>>> para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> KHOMP Inovação: External Board Series
>>>>>> Módulos de 1/2 rack e 1U para todas as interfaces e soluções Asterisk
>>>>>> e FreeSWITCH.
>>>>>> Tenha a External Series Experience na sua aplicação. Visite
>>>>>> www.khomp.com
>>>>>> _______________________________________________
>>>>>> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
>>>>>> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
>>>>>> Centro Treinamento - Curso de PABX IP - Asterisk - Site
>>>>>> www.digivoice.com.br
>>>>>> ________
>>>>>> YEALINK: Telefones IP e VídeoPhones IP com o melhor custo/benefício
>>>>>> do mercado.
>>>>>> email: yealink em commlogik.com.br | www.commlogik.com.br | (11)
>>>>>> 5503-1011
>>>>>> ______________________________________________
>>>>>> Para remover seu email desta lista, basta enviar um email em branco
>>>>>> para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> KHOMP Inovação: External Board Series
>>>>> Módulos de 1/2 rack e 1U para todas as interfaces e soluções Asterisk
>>>>> e FreeSWITCH.
>>>>> Tenha a External Series Experience na sua aplicação. Visite
>>>>> www.khomp.com
>>>>> _______________________________________________
>>>>> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
>>>>> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
>>>>> Centro Treinamento - Curso de PABX IP - Asterisk - Site
>>>>> www.digivoice.com.br
>>>>> ________
>>>>> YEALINK: Telefones IP e VídeoPhones IP com o melhor custo/benefício do
>>>>> mercado.
>>>>> email: yealink em commlogik.com.br | www.commlogik.com.br | (11)
>>>>> 5503-1011
>>>>> ______________________________________________
>>>>> Para remover seu email desta lista, basta enviar um email em branco
>>>>> para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> KHOMP Inovação: External Board Series
>>>> Módulos de 1/2 rack e 1U para todas as interfaces e soluções Asterisk e
>>>> FreeSWITCH.
>>>> Tenha a External Series Experience na sua aplicação. Visite
>>>> www.khomp.com
>>>> _______________________________________________
>>>> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
>>>> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
>>>> Centro Treinamento - Curso de PABX IP - Asterisk - Site
>>>> www.digivoice.com.br
>>>> ________
>>>> YEALINK: Telefones IP e VídeoPhones IP com o melhor custo/benefício do
>>>> mercado.
>>>> email: yealink em commlogik.com.br | www.commlogik.com.br | (11) 5503-1011
>>>> ______________________________________________
>>>> Para remover seu email desta lista, basta enviar um email em branco
>>>> para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> KHOMP Inovação: External Board Series
>>> Módulos de 1/2 rack e 1U para todas as interfaces e soluções Asterisk e
>>> FreeSWITCH.
>>> Tenha a External Series Experience na sua aplicação. Visite
>>> www.khomp.com
>>> _______________________________________________
>>> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
>>> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
>>> Centro Treinamento - Curso de PABX IP - Asterisk - Site
>>> www.digivoice.com.br
>>> ________
>>> YEALINK: Telefones IP e VídeoPhones IP com o melhor custo/benefício do
>>> mercado.
>>> email: yealink em commlogik.com.br | www.commlogik.com.br | (11) 5503-1011
>>> ______________________________________________
>>> Para remover seu email desta lista, basta enviar um email em branco para
>>> asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>>
>>
>>
>> _______________________________________________
>> KHOMP Inovação: External Board Series
>> Módulos de 1/2 rack e 1U para todas as interfaces e soluções Asterisk e
>> FreeSWITCH.
>> Tenha a External Series Experience na sua aplicação. Visite www.khomp.com
>> _______________________________________________
>> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
>> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
>> Centro Treinamento - Curso de PABX IP - Asterisk - Site
>> www.digivoice.com.br
>> ________
>> YEALINK: Telefones IP e VídeoPhones IP com o melhor custo/benefício do
>> mercado.
>> email: yealink em commlogik.com.br | www.commlogik.com.br | (11) 5503-1011
>> ______________________________________________
>> Para remover seu email desta lista, basta enviar um email em branco para
>> asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>
>
>
> _______________________________________________
> KHOMP Inovação: External Board Series
> Módulos de 1/2 rack e 1U para todas as interfaces e soluções Asterisk e
> FreeSWITCH.
> Tenha a External Series Experience na sua aplicação. Visite www.khomp.com
> _______________________________________________
> DIGIVOICE Fabricante de Placas de Voz e Channel Bank
> 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
> Centro Treinamento - Curso de PABX IP - Asterisk - Site
> www.digivoice.com.br
> ________
> YEALINK: Telefones IP e VídeoPhones IP com o melhor custo/benefício do
> mercado.
> email: yealink em commlogik.com.br | www.commlogik.com.br | (11) 5503-1011
> ______________________________________________
> Para remover seu email desta lista, basta enviar um email em branco para
> asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>
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