[AsteriskBrasil] Digium (E1) TE110P não origina chamada
Alexandre Cavalcante Alencar
alexandre.alencar em gmail.com
Segunda Abril 28 11:01:13 BRT 2008
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/
More information about the AsteriskBrasil
mailing list