Uma vez tive um problema semelhante, resolvi corrigindo o dtmf se nao me engano
coloquei inband.
Date: Thu, 1 Jul 2010 11:39:12 -0300
From: [email protected]
To: [email protected]
Subject: Re: [AsteriskBrasil] Elastix 1.6 + Linksys SPA 3102 MSG: The number
you have dialed is not in service please try again
Ele ta mandando pro tronco com 0 asntes do numero..
Tenta colocal
0|. na rota de saida, dae ele vai cortar o zero..
Em 1 de julho de 2010 09:24, Gleidison Sampaio <[email protected]>
escreveu:
Senhores
consegui fazer ligações através da linha PSTN, porém quando disco para um
numero de celular por exemplo a ligação vai para outro numero completamente
diferente, segue abaixo o log do momento da chamada.
Obs. os xxxxxx destacados em vermelho é o nome do meu trunk ou ip do Elastix
-- Executing [099215...@from-internal:1] Macro("SIP/12-088606a8",
"user-callerid|SKIPTTL|") in new stack
-- Executing [...@macro-user-callerid:1] Set("SIP/12-088606a8",
"AMPUSER=12") in new stack -- Executing [...@macro-user-callerid:2]
GotoIf("SIP/12-088606a8", "0?report") in new stack
-- Executing [...@macro-user-callerid:3] ExecIf("SIP/12-088606a8",
"1|Set|REALCALLERIDNUM=12") in new stack -- Executing
[...@macro-user-callerid:4] Set("SIP/12-088606a8", "AMPUSER=12") in new stack
-- Executing [...@macro-user-callerid:5] Set("SIP/12-088606a8",
"AMPUSERCIDNAME=Atendente") in new stack -- Executing
[...@macro-user-callerid:6] GotoIf("SIP/12-088606a8", "0?report") in new stack
-- Executing [...@macro-user-callerid:7] Set("SIP/12-088606a8",
"AMPUSERCID=12") in new stack -- Executing [...@macro-user-callerid:8]
Set("SIP/12-088606a8", "CALLERID(all)="Atendente" <12>") in new stack
-- Executing [...@macro-user-callerid:9] ExecIf("SIP/12-088606a8",
"0|Set|CHANNEL(language)=") in new stack -- Executing
[...@macro-user-callerid:10] GotoIf("SIP/12-088606a8", "1?continue") in new
stack
-- Goto (macro-user-callerid,s,19) -- Executing
[...@macro-user-callerid:19] NoOp("SIP/12-088606a8", "Using CallerID
"Atendente" <12>") in new stack -- Executing [099215...@from-internal:2]
Set("SIP/12-088606a8", "_NODEST=") in new stack
-- Executing [099215...@from-internal:3] Macro("SIP/12-088606a8",
"record-enable|12|OUT|") in new stack -- Executing
[...@macro-record-enable:1] GotoIf("SIP/12-088606a8", "1?check") in new stack
-- Goto (macro-record-enable,s,4) -- Executing
[...@macro-record-enable:4] AGI("SIP/12-088606a8",
"recordingcheck|20100701-085010|1277985010.14") in new stack -- Launched AGI
Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20100701-085010|1277985010.14: Outbound recording enabled.
recordingcheck|20100701-085010|1277985010.14:
CALLFILENAME=OUT12-20100701-085010-1277985010.14 -- AGI Script
recordingcheck completed, returning 0
-- Executing [...@macro-record-enable:999] MixMonitor("SIP/12-088606a8",
"OUT12-20100701-085010-1277985010.14.wav||") in new stack -- Executing
[099215...@from-internal:4] Macro("SIP/12-088606a8",
"dialout-trunk|1|099215415||") in new stack
-- Executing [...@macro-dialout-trunk:1] Set("SIP/12-088606a8",
"DIAL_TRUNK=1") in new stack -- Executing [...@macro-dialout-trunk:2]
GosubIf("SIP/12-088606a8", "0?sub-pincheck|s|1") in new stack
-- Executing [...@macro-dialout-trunk:3] GotoIf("SIP/12-088606a8",
"0?disabletrunk|1") in new stack -- Executing [...@macro-dialout-trunk:4]
Set("SIP/12-088606a8", "DIAL_NUMBER=099215415") in new stack
-- Executing [...@macro-dialout-trunk:5] Set("SIP/12-088606a8",
"DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing
[...@macro-dialout-trunk:6] Set("SIP/12-088606a8", "OUTBOUND_GROUP=OUT_1") in
new stack
-- Executing [...@macro-dialout-trunk:7] GotoIf("SIP/12-088606a8",
"0?nomax") in new stack -- Executing [...@macro-dialout-trunk:8]
GotoIf("SIP/12-088606a8", "0?chanfull") in new stack
-- Executing [...@macro-dialout-trunk:9] GotoIf("SIP/12-088606a8",
"0?skipoutcid") in new stack -- Executing [...@macro-dialout-trunk:10]
Set("SIP/12-088606a8", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [...@macro-dialout-trunk:11] Macro("SIP/12-088606a8",
"outbound-callerid|1") in new stack -- Executing
[...@macro-outbound-callerid:1] ExecIf("SIP/12-088606a8", "0|SetCallerPres|")
in new stack
-- Executing [...@macro-outbound-callerid:2] ExecIf("SIP/12-088606a8",
"0|Set|REALCALLERIDNUM=12") in new stack -- Executing
[...@macro-outbound-callerid:3] GotoIf("SIP/12-088606a8", "1?normcid") in new
stack
-- Goto (macro-outbound-callerid,s,6) -- Executing
[...@macro-outbound-callerid:6] Set("SIP/12-088606a8", "USEROUTCID=") in new
stack -- Executing [...@macro-outbound-callerid:7] Set("SIP/12-088606a8",
"EMERGENCYCID=") in new stack
-- Executing [...@macro-outbound-callerid:8] Set("SIP/12-088606a8",
"TRUNKOUTCID=<xxxxxxxxxx>") in new stack -- Executing
[...@macro-outbound-callerid:9] GotoIf("SIP/12-088606a8", "1?trunkcid") in new
stack
-- Goto (macro-outbound-callerid,s,12) -- Executing
[...@macro-outbound-callerid:12] ExecIf("SIP/12-088606a8",
"1|Set|CALLERID(all)=<xxxxxxxx>") in new stack
-- Executing [...@macro-outbound-callerid:13] ExecIf("SIP/12-088606a8",
"0|Set|CALLERID(all)=") in new stack -- Executing
[...@macro-outbound-callerid:14] ExecIf("SIP/12-088606a8",
"0|SetCallerPres|prohib_passed_screen") in new stack
-- Executing [...@macro-dialout-trunk:12] ExecIf("SIP/12-088606a8",
"0|AGI|fixlocalprefix") in new stack -- Executing
[...@macro-dialout-trunk:13] Set("SIP/12-088606a8", "OUTNUM=099215415") in new
stack
-- Executing [...@macro-dialout-trunk:14] Set("SIP/12-088606a8",
"custom=SIP/xxxxxxxxx") in new stack -- Executing
[...@macro-dialout-trunk:15] ExecIf("SIP/12-088606a8",
"0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
-- Executing [...@macro-dialout-trunk:16] Macro("SIP/12-088606a8",
"dialout-trunk-predial-hook|") in new stack -- Executing
[...@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/12-088606a8", "") in
new stack
-- Executing [...@macro-dialout-trunk:17] GotoIf("SIP/12-088606a8",
"0?bypass|1") in new stack -- Executing [...@macro-dialout-trunk:18]
GotoIf("SIP/12-088606a8", "0?customtrunk") in new stack
-- Executing [...@macro-dialout-trunk:19] Dial("SIP/12-088606a8",
"SIP/xxxxxxxx/099215415|300|") in new stack -- Called xxxxxxxxx/099215415
== Begin MixMonitor Recording SIP/12-088606a8 -- SIP/xxxxxxxxx-088646c8 is
ringing -- SIP/xxxxxxxxx-088646c8 answered SIP/12-088606a8
-- Executing [...@macro-dialout-trunk:1] Macro("SIP/12-088606a8",
"hangupcall|") in new stack -- Executing [...@macro-hangupcall:1]
GotoIf("SIP/12-088606a8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4) -- Executing [...@macro-hangupcall:4]
GotoIf("SIP/12-088606a8", "1?skipblkvm") in new stack -- Goto
(macro-hangupcall,s,7) -- Executing [...@macro-hangupcall:7]
GotoIf("SIP/12-088606a8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9]
Hangup("SIP/12-088606a8", "") in new stack == Spawn extension
(macro-hangupcall, s, 9) exited non-zero on 'SIP/12-088606a8' in macro
'hangupcall'
== Spawn h extension (macro-dialout-trunk, h, 1) exited non-zero on
'SIP/12-088606a8' == Spawn extension (macro-dialout-trunk, s, 19) exited
non-zero on 'SIP/12-088606a8' in macro 'dialout-trunk'
== Spawn extension (from-internal, 099215415, 4) exited non-zero on
'SIP/12-088606a8' == MixMonitor close filestream == End MixMonitor Recording
SIP/12-088606a8
From: [email protected]
To: [email protected]
Date: Thu, 1 Jul 2010 07:57:06 -0400
Subject: Re: [AsteriskBrasil] Elastix 1.6 + Linksys SPA 3102 MSG: The number
you have dialed is not in service please try again
Roger, bom dia
Primeiro obrigado pela ajuda, então depois de alterar para "from-internal" ele
aciona o tronco PSTN mais fica mudo e não faz a chamada, abaixo segue a
configuração do trunk...
PEER DETAILS:
disallow=allallow=ulawcanreinvite=nocontext=from-trunkdtmfmode=rfc2833
host=dynamicincominglimit=1nat=neverport=5061qualify=yessecret=xxxxxxtype=friendusername=xxxx
REGISTER STRING:
xxxx:[email protected]:5061/xxxx
Date: Thu, 1 Jul 2010 00:50:18 -0300
From: [email protected]
To: [email protected]
Subject: Re: [AsteriskBrasil] Elastix 1.6 + Linksys SPA 3102 MSG: The number
you have dialed is not in service please try again
Seu ramal "parece" estar com o context setado como "from-trunk"...Deveria ser
from-internal.. Dá uma conferida ae
Em 30 de junho de 2010 14:00, Gleidison Sampaio <[email protected]>
escreveu:
Boa tarde Srs,
Meu Elastix esta recebendo as ligações da minha linha PSTN tudo certinho, porém
não consigo originar chamadas para numeros nenhum, segue abaixo log que
capturei. se alguem tiver alguma ajuda.
-- Executing [98201...@from-trunk:1] Set("SIP/12-b72087d0",
"__FROM_DID=98201590") in new stack -- Executing [98201...@from-trunk:2]
NoOp("SIP/12-b72087d0", "Received an unknown call with DID set to 98201590") in
new stack
-- Executing [98201...@from-trunk:3] Goto("SIP/12-b72087d0", "s|a2") in new
stack -- Goto (from-trunk,s,2) -- Executing [...@from-trunk:2]
Answer("SIP/12-b72087d0", "") in new stack
-- Executing [...@from-trunk:3] Wait("SIP/12-b72087d0", "2") in new
stackReally destroying SIP dialog '[email protected]'
Method: OPTIONS
-- Executing [...@from-trunk:4] Playback("SIP/12-b72087d0", "ss-noservice")
in new stack -- <SIP/12-b72087d0> Playing 'ss-noservice' (language 'en')
REGISTER attempt 29 to [email protected] destroying SIP dialog
'[email protected]' Method: REGISTER
-- Executing [...@from-trunk:5] SayAlpha("SIP/12-b72087d0", "98201590") in
new stack -- <SIP/12-b72087d0> Playing 'digits/9' (language 'en') --
<SIP/12-b72087d0> Playing 'digits/8' (language 'en')
-- <SIP/12-b72087d0> Playing 'digits/2' (language 'en') == Spawn extension
(from-trunk, s, 5) exited non-zero on 'SIP/12-b72087d0' -- Executing
[...@from-trunk:1] Hangup("SIP/12-b72087d0", "") in new stack
== Spawn extension (from-trunk, h, 1) exited non-zero on
'SIP/12-b72087d0'Really destroying SIP dialog '[email protected]'
Method: BYE
-- Executing [98201...@from-trunk:1] Set("SIP/12-b7209df8",
"__FROM_DID=98201590") in new stack -- Executing [98201...@from-trunk:2]
NoOp("SIP/12-b7209df8", "Received an unknown call with DID set to 98201590") in
new stack
-- Executing [98201...@from-trunk:3] Goto("SIP/12-b7209df8", "s|a2") in new
stack -- Goto (from-trunk,s,2) -- Executing [...@from-trunk:2]
Answer("SIP/12-b7209df8", "") in new stack
-- Executing [...@from-trunk:3] Wait("SIP/12-b7209df8", "2") in new
stackReally destroying SIP dialog '794742104b0f4274' Method: REGISTER --
Executing [...@from-trunk:4] Playback("SIP/12-b7209df8", "ss-noservice") in new
stack
-- <SIP/12-b7209df8> Playing 'ss-noservice' (language 'en') == Spawn
extension (from-trunk, s, 4) exited non-zero on 'SIP/12-b7209df8' --
Executing [...@from-trunk:1] Hangup("SIP/12-b7209df8", "") in new stack
== Spawn extension (from-trunk, h, 1) exited non-zero on
'SIP/12-b7209df8'Really destroying SIP dialog '[email protected]'
Method: BYE
Really destroying SIP dialog '[email protected]'
Method: REGISTER
VEJA TODOS OS SEUS EMAILS DE VÁRIAS CONTAS COM UM SÓ LOGIN. CLIQUE AQUI E VEJA
COMO.
_______________________________________________
KHOMP: qualidade em placas de E1, GSM, FXS e FXO para Asterisk.
- Hardware com alta disponibilidade de recursos e qualidade KHOMP
- Suporte técnico local qualificado e gratuito
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_______________________________________________
Temos tudo para seu projeto VoIP com Asterisk!
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--
----
Roger Pitigliani
[email protected]
msn: [email protected]
Gravataí - RS
PARA NAVEGAR COM MAIS PRIVACIDADE USE O INTERNET EXPLORER 8. INSTALE GRÁTIS.
O INTERNET EXPLORER 8 AJUDA VOCÊ A FICAR LONGE DOS VÍRUS. DESCUBRA COMO.
_______________________________________________
KHOMP: qualidade em placas de E1, GSM, FXS e FXO para Asterisk.
- Hardware com alta disponibilidade de recursos e qualidade KHOMP
- Suporte técnico local qualificado e gratuito
Conheça a linha completa de produtos KHOMP em www.khomp.com.br
_______________________________________________
Temos tudo para seu projeto VoIP com Asterisk!
Descontos especiais para assinantes da AsteriskBrasil.org.
Registre-se e receba um cupom exclusivo de desconto!
Acesse agora www.voipmania.com.br
______________________________________________
Lista de discussões AsteriskBrasil.org
[email protected]
http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
--
----
Roger Pitigliani
[email protected]
msn: [email protected]
Gravataí - RS
_________________________________________________________________
CONVERSE COM SEUS AMIGOS E OS VEJA PELA WEBCAM NO MESSENGER. CLIQUE AQUI E VEJA
COMO.
http://www.windowslive.com.br/public/tip.aspx/view/84?product=2&ocid=WLCRM:Live:Hotmail:Tagline:senDimensao:CONVERSECO85:-_______________________________________________
KHOMP: qualidade em placas de E1, GSM, FXS e FXO para Asterisk.
- Hardware com alta disponibilidade de recursos e qualidade KHOMP
- Suporte técnico local qualificado e gratuito
Conheça a linha completa de produtos KHOMP em www.khomp.com.br
_______________________________________________
Temos tudo para seu projeto VoIP com Asterisk!
Descontos especiais para assinantes da AsteriskBrasil.org.
Registre-se e receba um cupom exclusivo de desconto!
Acesse agora www.voipmania.com.br
______________________________________________
Lista de discussões AsteriskBrasil.org
[email protected]
http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
