[AsteriskBrasil] Digium (E1) TE110P não origina chamada
Demétrios Andrigo
andrigo em ntelecom.com.br
Segunda Abril 28 11:15:04 BRT 2008
Foi minha primeira hipótese. Problemas com a Operadora. Entretanto o link
funciona com um PABX ISDN numa boa. Fazendo e recebendo ligações.
Tem outro fator interessante que gostaria de compartilhar. Esta E1 tem 30
linhas e se eu tento discar pra qualquer dos números dela vai que é uma
maravilha.
Sobre o callerid já tirei da configuração.
----- Original Message -----
From: "Alexandre Cavalcante Alencar" <alexandre.alencar em gmail.com>
To: <asteriskbrasil em listas.asteriskbrasil.org>
Sent: Monday, April 28, 2008 11:01 AM
Subject: Re: [AsteriskBrasil]Digium (E1) TE110P não origina chamada
> Olá,
>
> A operadora bloqueou a chamada... o circuito já foi liberado para
> originar chamadas? Lembro que isto já ocorreu antes com um de meus
> sistemas e era a operadora que não havia configurado meu circuito para
> originar chamadas.
>
> PS: Não tente setar o callerid...
>
> Ats
>
> On Mon, Apr 28, 2008 at 10:51 AM, Demétrios Andrigo
> <andrigo em ntelecom.com.br> wrote:
>>
>>
>> Obrigado pela ajuda, mas insistiu no mesmo erro.
>> Coloquei debug no span 1 para ver mais detalhes e segue abaixo o que ele
>> tenta fazer no momento que efetuou uma chamada externa:
>>
>> -- Executing [s em macro-dialout-trunk:20] Dial("SIP/6000-09628100",
>> "ZAP/g0/24095213|300|") in new stack
>> -- Making new call for cr 32771
>> -- Requested transfer capability: 0x00 - SPEECH
>> > Protocol Discriminator: Q.931 (8) len=30
>> > Call Ref: len= 2 (reference 3/0x3) (Originator)
>> > Message type: SETUP (5)
>> > [04 03 80 90 a3]
>> > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
>> capability: Speech (0)
>> > Ext: 1 Trans mode/rate: 64kbps,
>> > circuit-mode
>> (16)
>> > Ext: 1 User information layer 1: A-Law
>> > (35)
>> > [18 03 a9 83 81]
>> > Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive
>> Dchan: 0
>> > ChanSel: Reserved
>> > Ext: 1 Coding: 0 Number Specified Channel
>> > Type: 3
>> > Ext: 1 Channel: 1 ]
>> > [6c 02 00 c3]
>> > Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
>> Unknown Number Plan (0)
>> > Presentation: Number not available (67) '' ]
>> > [70 09 a1 32 34 30 39 35 32 31 33]
>> > Called Number (len=11) [ Ext: 1 TON: National Number (2) NPI:
>> ISDN/Telephony Numbering Plan (E.164/E.163) (1) '24095213' ]
>> q931.c:2881 q931_setup: call 32771 on channel 1 enters state 1 (Call
>> Initiated)
>> -- Called g0/24095213
>> < Protocol Discriminator: Q.931 (8) len=10
>> < Call Ref: len= 2 (reference 3/0x3) (Terminator)
>> < Message type: CALL PROCEEDING (2)
>> < [18 03 a9 83 81]
>> < Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive
>> Dchan: 0
>> < ChanSel: Reserved
>> < Ext: 1 Coding: 0 Number Specified Channel
>> Type: 3
>> < Ext: 1 Channel: 1 ]
>> -- Processing IE 24 (cs0, Channel Identification)
>> q931.c:3428 q931_receive: call 32771 on channel 1 enters state 3
>> (Outgoing
>> call Proceeding)
>> -- Zap/1-1 is proceeding passing it to SIP/6000-09628100
>> < Protocol Discriminator: Q.931 (8) len=13
>> < Call Ref: len= 2 (reference 3/0x3) (Terminator)
>> < Message type: DISCONNECT (69)
>> < [08 02 82 9f]
>> < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
>> Location: Public network serving the local user (2)
>> < Ext: 1 Cause: Normal, unspecified (31), class =
>> Normal
>> Event (1) ]
>> < [1e 02 82 88]
>> < Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0)
>> 0:
>> 0 Location: Public network serving the local user (2)
>> < Ext: 1 Progress Description: Inband
>> information or appropriate pattern now available. (8) ]
>> -- Processing IE 8 (cs0, Cause)
>> -- Processing IE 30 (cs0, Progress Indicator)
>> q931.c:3563 q931_receive: call 32771 on channel 1 enters state 12
>> (Disconnect Indication)
>> -- Channel 0/1, span 1 got hangup request, cause 31
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
>> peerstate Disconnect Request
>> q931.c:2716 q931_release: call 32771 on channel 1 enters state 19
>> (Release
>> Request)
>> > Protocol Discriminator: Q.931 (8) len=9
>> > Call Ref: len= 2 (reference 3/0x3) (Originator)
>> > Message type: RELEASE (77)
>> > [08 02 81 9f]
>> > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
>> Location: Private network serving the local user (1)
>> > Ext: 1 Cause: Normal, unspecified (31), class =
>> > Normal
>> Event (1) ]
>> -- Hungup 'Zap/1-1'
>> == Everyone is busy/congested at this time (1:0/0/1)
>> -- Executing [s em macro-dialout-trunk:21] Goto("SIP/6000-09628100",
>> "s-CHANUNAVAIL|1") in new stack
>> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
>> -- Executing [s-CHANUNAVAIL em macro-dialout-trunk:1]
>> GotoIf("SIP/6000-09628100", "1?noreport") in new stack
>> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
>> -- Executing [s-CHANUNAVAIL em macro-dialout-trunk:3]
>> NoOp("SIP/6000-09628100", "TRUNK Dial failed due to CHANUNAVAIL - failing
>> through to other trunks") in new stack
>> -- Executing [024095213 em from-internal:5] Macro("SIP/6000-09628100",
>> "outisbusy|") in new stack
>> -- Executing [s em macro-outisbusy:1] Playback("SIP/6000-09628100",
>> "all-circuits-busy-now|noanswer") in new stack
>> -- <SIP/6000-09628100> Playing 'all-circuits-busy-now' (language
>> 'pt_BR')
>> < Protocol Discriminator: Q.931 (8) len=5
>> < Call Ref: len= 2 (reference 3/0x3) (Terminator)
>> < Message type: RELEASE COMPLETE (90)
>> q931.c:3503 q931_receive: call 32771 on channel 1 enters state 0 (Null)
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
>> NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
>>
>> Obrigado mais uma vez.
>>
>>
>> ----- Original Message -----
>> From: "Alexandre Cavalcante Alencar" <alexandre.alencar em gmail.com>
>> To: <asteriskbrasil em listas.asteriskbrasil.org>
>> Sent: Monday, April 28, 2008 10:34 AM
>> Subject: Re: [AsteriskBrasil]Digium (E1) TE110P não origina chamada
>>
>> > Olá,
>> >
>> > Altere seu zapata.conf para o conteúdo abaixo e veja se resolve seu
>> > problema. Aqui é um Oi ISDN.
>> >
>> > [channels]
>> > language = br
>> > usecallerid = yes
>> > hidecallerid = yes
>> > callwaiting = yes
>> > callwaitingcallerid = yes
>> > restrictcid = no
>> > usecallingpres = yes
>> > threewaycalling = yes
>> > callreturn = yes
>> > transfer = yes
>> > cancallforward = yes
>> > echocancelwhenbridged = yes
>> > echocancel = yes
>> > musiconhold = default
>> > overlapdial = yes
>> > immediate = no
>> >
>> > ; Digium Wildcard TE110P
>> > context = oi ; <<<< Lembre de mudar o
>> contexto
>> > switchtype = euroisdn
>> > signalling = pri_cpe
>> > group = 0
>> > callgroup = 1
>> > pickupgroup = 1
>> > channel => 1-15,17-31
>> >
>> >
>> > On Mon, Apr 28, 2008 at 10:11 AM, Demétrios Andrigo
>> > <andrigo em ntelecom.com.br> wrote:
>> >>
>> >>
>> >>
>> >> Olá a todos,
>> >>
>> >> Tenho configurado um asterisk 1.4.18 com uma placa Digium TE110P (E1
>> ISDN),
>> >> em um Trixbox. Temos contratado o serviços da CTBC.
>> >>
>> >> Consigo receber ligações sem problema, porém quando tento originar uma
>> >> chamada, me retorna o erro:
>> >>
>> >> E a gravação diz que todos os circuitos estão ocupados, porém, as
>> >> linhas
>> >> estão livres. Abaixo seguem meus arquivos de conf.
>> >>
>> >> [zaptel.conf]
>> >> # Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER)
>> >> HDB3/CCS/CRC4
>> >> span=1,1,0,ccs,hdb3,crc4
>> >> # termtype: te
>> >> bchan=1-15,17-31
>> >> dchan=16
>> >>
>> >> # Global data
>> >> loadzone=br
>> >> defaultzone=br
>> >>
>> >> [zapata.conf]
>> >> [trunkgroups]
>> >>
>> >> [channels]
>> >> pridialplan=unknown
>> >> prilocaldialplan=unknown
>> >> language = br
>> >> resetinterval = never
>> >> usecallerid = yes
>> >> hidecallerid = no
>> >> callwaiting = no
>> >> usecallingpres = yes
>> >> callwaitingcallerid = yes
>> >> threewaycalling = yes
>> >> transfer = yes
>> >> canpark = yes
>> >> cancallforward = yes
>> >> callreturn = yes
>> >> echocancelwhenbridged = yes
>> >> echocancel = yes
>> >> rxgain = 0.0
>> >> txgain = 0.0
>> >> overlapdial=no
>> >> callprogress=no
>> >> busydetec=yes
>> >> pulsedial=yes
>> >>
>> >> group=0
>> >> callgroup = 0
>> >> pickupgroup = 0
>> >> context=from-zaptel
>> >> switchtype = euroisdn
>> >> dtmfmode=rfc2833
>> >> signalling=pri_cpe
>> >> channel = 1-15,17-31
>> >> callerid=<nosso_num>
>> >>
>> >> [extensions.conf]
>> >>
>> >> [outrt-001-0_outside]
>> >> include => outrt-001-0_outside-custom
>> >> exten => _0.,1,Macro(user-callerid,SKIPTTL,)
>> >> exten => _0.,n,Set(_NODEST=)
>> >> exten => _0.,n,Macro(record-enable,${AMPUSER},OUT,)
>> >> exten => _0.,n,Macro(dialout-trunk,1,${EXTEN:1},,)
>> >> exten => _0.,n,Macro(outisbusy,)
>> >>
>> >> ; end of [outrt-001-0_outside]
>> >>
>> >> Obrigado pela ajuda.
>> >> _______________________________________________
>> >> Compre uma camiseta da AsteriskBrasil.org!
>> >> http://www.voipmania.com.br
>> >> == VoIPMania.com.br ==
>> >>
>> >> _______________________________________________
>> >> Lista de discussões AsteriskBrasil.org
>> >> AsteriskBrasil em listas.asteriskbrasil.org
>> >> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>> >>
>> >
>> >
>> >
>> > --
>> > Alexandre C Alencar (Skarmeth)
>> > http://blog.alexandrealencar.net/
>> > http://www.alexandrealencar.net/
>> > http://people.debian-ce.org/skarmeth/
>> > _______________________________________________
>> > Compre uma camiseta da AsteriskBrasil.org!
>> > http://www.voipmania.com.br
>> > == VoIPMania.com.br ==
>> >
>> > _______________________________________________
>> > Lista de discussões AsteriskBrasil.org
>> > AsteriskBrasil em listas.asteriskbrasil.org
>> > http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>> >
>> _______________________________________________
>> Compre uma camiseta da AsteriskBrasil.org!
>> http://www.voipmania.com.br
>> == VoIPMania.com.br ==
>>
>> _______________________________________________
>> Lista de discussões AsteriskBrasil.org
>> AsteriskBrasil em listas.asteriskbrasil.org
>> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>>
>
>
>
> --
> Alexandre C Alencar (Skarmeth)
> http://blog.alexandrealencar.net/
> http://www.alexandrealencar.net/
> http://people.debian-ce.org/skarmeth/
> _______________________________________________
> Compre uma camiseta da AsteriskBrasil.org!
> http://www.voipmania.com.br
> == VoIPMania.com.br ==
>
> _______________________________________________
> Lista de discussões AsteriskBrasil.org
> AsteriskBrasil em listas.asteriskbrasil.org
> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>
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