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: gleidison.samp...@hotmail.com
To: asteriskbrasil@listas.asteriskbrasil.org
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=rfc2833host=dynamicincominglimit=1nat=neverport=5061qualify=yessecret=xxxxxxtype=friendusername=xxxx
REGISTER STRING:
xxxx:xx...@10.x.x.x:5061/xxxx





Date: Thu, 1 Jul 2010 00:50:18 -0300
From: rogerwin...@gmail.com
To: asteriskbrasil@listas.asteriskbrasil.org
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 <gleidison.samp...@hotmail.com> 
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 '0cb41fb06d71b2e0385c4f3b26424...@x.x.x.x' 
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 xxxxxx...@x.x.x.xreally destroying SIP dialog 
'6e03058055b63ec6034244496845d...@127.0.0.1' 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 '74f94492-a71b9...@x.x.x.x' 
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 '2c3a477-d9bc2...@x.x.x.x' 
Method: BYE
Really destroying SIP dialog '6e03058055b63ec6034244496845d...@127.0.0.1' 
Method: REGISTER




                                          
VEJA TODOS OS SEUS EMAILS DE VÁRIAS CONTAS COM UM SÓ LOGIN. CLIQUE AQUI E VEJA 
COMO.


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-- 
----

Roger Pitigliani
rogerwin...@gmail.com
msn: roger_pitigli...@hotmail.com
Gravataí - RS

                                          
PARA NAVEGAR COM MAIS PRIVACIDADE USE O INTERNET EXPLORER 8. INSTALE GRÁTIS.    
                                  
_________________________________________________________________
ACESSE SEUS EMAILS DE QUALQUER LUGAR PELO SEU CELULAR. CLIQUE E VEJA COMO FAZER 
ISSO.
http://celular.windowslive.com.br/hotmail.asp?produto=Hotmail&utm_source=Live_Hotmail&utm_medium=Tagline&utm_content=ACESSESEUS85&utm_campaign=MobileServices
_______________________________________________
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
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