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 < gleidison.samp...@hotmail.com> 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: 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=all > allow=ulaw > canreinvite=no > context=from-trunk > dtmfmode=rfc2833 > host=dynamic > incominglimit=1 > nat=never > port=5061 > qualify=yes > secret=xxxxxx > type=friend > username=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 > stack > Really destroying SIP dialog > '0cb41fb06d71b2e0385c4f3b26424...@x<0cb41fb06d71b2e0385c4f3b26424...@192.168.2.131>.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 xxxxxxxxx@ <4430170...@192.168.2.131>x.x.x.x > Really 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<74f94492-a71b9...@192.168.2.222>.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 > stack > Really 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<2c3a477-d9bc2...@192.168.2.222>.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.<http://www.windowslive.com.br/public/tip.aspx/view/16?product=1&ocid=Hotmail:Live:Hotmail:Tagline:1x1:VEJATODOSO84:-> > > _______________________________________________ > 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 > AsteriskBrasil@listas.asteriskbrasil.org > http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil > > > > > -- > ---- > Roger Pitigliani > rogerwin...@gmail.com > msn: roger_pitigli...@hotmail.com > Gravataí - RS > > ------------------------------ > PARA NAVEGAR COM MAIS PRIVACIDADE USE O INTERNET EXPLORER 8. INSTALE > GRÁTIS.<http://www.microsoft.com/brasil/windows/internet-explorer/default.aspx?WT.mc_id=1633> > ------------------------------ > O INTERNET EXPLORER 8 AJUDA VOCÊ A FICAR LONGE DOS VÍRUS. DESCUBRA > COMO.<http://www.microsoft.com/brasil/windows/internet-explorer/features/stay-safer-online.aspx?tabid=1&catid=1&WT.mc_id=1632> > > _______________________________________________ > 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 > AsteriskBrasil@listas.asteriskbrasil.org > http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil > -- ---- Roger Pitigliani rogerwin...@gmail.com msn: roger_pitigli...@hotmail.com Gravataí - RS
_______________________________________________ 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 AsteriskBrasil@listas.asteriskbrasil.org http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil