[AsteriskBrasil] E1 e SIP Trunk !

[Emerson Mochon Borsatti]

Pessoal,

Já resolvi, faltava esse trecho no meu extensions.conf

[macro-stdexten]

exten => s,1,Dial(${ARG2},20)
exten => s,2,Goto(s-${DIALSTATUS},1)

Emerson

2010/11/29 Emerson Mochon Borsatti <borsatti.em em gmail.com>

> Boa tarde pessoal,
>
> Tenho 2 servidores com Asterisk e em um deles esta meu E1, chamadas de
> saída funcionam perfeitamente, o que não funciona é a chamada de entrada.
> Nem tenho o contexto default no meu extensions.conf e no servidor onde
> estão os ramais ele diz que o contexto é o default.
> A ligação entre eles é através de um SIP Trunk. Eis abaixo os erros:
>
> No Asterisk com E1:
>
> MFC/R2 call has been accepted on backward channel 24
>     -- Executing [3108 em imti-broad:1] Dial("DAHDI/24-1",
> "SIP/sipproxy01/3108,30") in new stack
>   == Using SIP RTP CoS mark 5
>     -- Called sipproxy01/3108
>     -- Got SIP response 603 "Declined" back from 172.17.0.30
>     -- SIP/sipproxy01-00000021 is busy
>   == Everyone is busy/congested at this time (1:1/0/0)
>
> No Asterisk onde estão os ramais:
>
> == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>     -- Executing [3108 em default:1] Macro("SIP/voipe1-000000cc",
> "stdexten,3108,SIP/3108") in new stack
> [Nov 29 14:10:42] WARNING[26445]: app_macro.c:304 _macro_exec: No such
> context 'macro-stdexten' for macro 'stdexten'
>     -- Auto fallthrough, channel 'SIP/voipe1-000000cc' status is 'UNKNOWN'
>
> No meu Asterisk que estão os ramais: extensions.conf
>
>
> [imti-pmcg]
>
> exten=>_31XX,1,Dial(SIP/${EXTEN},30)
>
> include = embratel-out
>
>
> [imti-broad]
>
> exten=>_31XX,1,Dial(SIP/${EXTEN},30)
>
> include = imti-pmcg
> include = broad-out-local
> include = broad-out-ddd
> include = broad-out-inter
>
> [broad-out-local]
> exten=>_9[012345]XXXXXXX,1,Dial(SIP/0115567${EXTEN:1}@sip.broadvoice.com<EXTEN%3A1%7D em sip.broadvoice.com>
> ,30)
> exten=>_9[012345]XXXXXXX,2,Playback(invalid)
> exten=>_9[012345]XXXXXXX,3,Hangup
>
> exten=>_9XX[012345]XXXXXXX,1,Dial(SIP/01155${EXTEN:1}@sip.broadvoice.com<EXTEN%3A1%7D em sip.broadvoice.com>
> ,30)
> exten=>_9XX[012345]XXXXXXX,2,Playback(invalid)
> exten=>_9XX[012345]XXXXXXX,3,Hangup
>
> exten=>_9XXXX[012345]XXXXXXX,1,Dial(SIP/011${EXTEN:1}@sip.broadvoice.com<EXTEN%3A1%7D em sip.broadvoice.com>
> ,30)
> exten=>_9XXXX[012345]XXXXXXX,2,Playback(invalid)
> exten=>_9XXXX[012345]XXXXXXX,3,Hangup
>
> [broad-out-ddd]
> exten=>_8XX[012345]XXXXXXX,1,Dial(SIP/01155${EXTEN:1}@sip.broadvoice.com<EXTEN%3A1%7D em sip.broadvoice.com>
> ,30)
> exten=>_8XX[012345]XXXXXXX,2,Playback(invalid)
> exten=>_8XX[012345]XXXXXXX,3,Hangup
>
> [broad-out-inter]
> exten=>_7XXXX[012345]XXXXXXX,1,Dial(SIP/011${EXTEN:1}@sip.broadvoice.com<EXTEN%3A1%7D em sip.broadvoice.com>
> ,30)
> exten=>_7XXXX[012345]XXXXXXX,2,Playback(invalid)
> exten=>_7XXXX[012345]XXXXXXX,3,Hangup
>
> [embratel-out]
> exten => _0XXXXXXXX,1,Dial(SIP/voipe1/${EXTEN:1})
> exten => _0XXXXXXXX,2,Playback(invalid)
> exten => _0XXXXXXXX,3,Hangup()
>
>
-------------- Próxima Parte ----------
Um anexo em HTML foi limpo...
URL: http://listas.asteriskbrasil.org/pipermail/asteriskbrasil/attachments/20101129/f1b6c20c/attachment.htm