[AsteriskBrasil] Digium (E1) TE110P não origina chamada
Demétrios Andrigo
andrigo em ntelecom.com.br
Segunda Abril 28 10:51:53 BRT 2008
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
>
-------------- Próxima Parte ----------
Um anexo em HTML foi limpo...
URL: http://listas.asteriskbrasil.org/pipermail/asteriskbrasil/attachments/20080428/cda4aca8/attachment-0001.htm
More information about the AsteriskBrasil
mailing list