Ele ta mandando pro tronco com 0 asntes do numero..<br>Tenta colocal <br>0|. na rota de saida, dae ele vai cortar o zero..<br><br><br><br><br><div class="gmail_quote">Em 1 de julho de 2010 09:24, Gleidison Sampaio <span dir="ltr"><<a href="mailto:gleidison.sampaio@hotmail.com" target="_blank">gleidison.sampaio@hotmail.com</a>></span> escreveu:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div>
<div>Senhores</div><div><br></div><div>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.</div>
<div><br></div><div>Obs. os xxxxxx destacados em vermelho é o nome do meu trunk ou ip do Elastix</div><div><br></div><div>-- Executing [099215415@from-internal:1] Macro("SIP/12-088606a8", "user-callerid|SKIPTTL|") in new stack</div>
<div> -- Executing [s@macro-user-callerid:1] Set("SIP/12-088606a8", "AMPUSER=12") in new stack</div><div> -- Executing [s@macro-user-callerid:2] GotoIf("SIP/12-088606a8", "0?report") in new stack</div>
<div> -- Executing [s@macro-user-callerid:3] ExecIf("SIP/12-088606a8", "1|Set|REALCALLERIDNUM=12") in new stack</div><div> -- Executing [s@macro-user-callerid:4] Set("SIP/12-088606a8", "AMPUSER=12") in new stack</div>
<div> -- Executing [s@macro-user-callerid:5] Set("SIP/12-088606a8", "AMPUSERCIDNAME=Atendente") in new stack</div><div> -- Executing [s@macro-user-callerid:6] GotoIf("SIP/12-088606a8", "0?report") in new stack</div>
<div> -- Executing [s@macro-user-callerid:7] Set("SIP/12-088606a8", "AMPUSERCID=12") in new stack</div><div> -- Executing [s@macro-user-callerid:8] Set("SIP/12-088606a8", "CALLERID(all)="Atendente" <12>") in new stack</div>
<div> -- Executing [s@macro-user-callerid:9] ExecIf("SIP/12-088606a8", "0|Set|CHANNEL(language)=") in new stack</div><div> -- Executing [s@macro-user-callerid:10] GotoIf("SIP/12-088606a8", "1?continue") in new stack</div>
<div> -- Goto (macro-user-callerid,s,19)</div><div> -- Executing [s@macro-user-callerid:19] NoOp("SIP/12-088606a8", "Using CallerID "Atendente" <12>") in new stack</div><div> -- Executing [099215415@from-internal:2] Set("SIP/12-088606a8", "_NODEST=") in new stack</div>
<div> -- Executing [099215415@from-internal:3] Macro("SIP/12-088606a8", "record-enable|12|OUT|") in new stack</div><div> -- Executing [s@macro-record-enable:1] GotoIf("SIP/12-088606a8", "1?check") in new stack</div>
<div> -- Goto (macro-record-enable,s,4)</div><div> -- Executing [s@macro-record-enable:4] AGI("SIP/12-088606a8", "recordingcheck|20100701-085010|1277985010.14") in new stack</div><div> -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck</div>
<div> recordingcheck|20100701-085010|1277985010.14: Outbound recording enabled.</div><div> recordingcheck|20100701-085010|1277985010.14: CALLFILENAME=OUT12-20100701-085010-1277985010.14</div><div> -- AGI Script recordingcheck completed, returning 0</div>
<div> -- Executing [s@macro-record-enable:999] MixMonitor("SIP/12-088606a8", "OUT12-20100701-085010-1277985010.14.wav||") in new stack</div><div> -- Executing [099215415@from-internal:4] Macro("SIP/12-088606a8", "dialout-trunk|1|099215415||") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:1] Set("SIP/12-088606a8", "DIAL_TRUNK=1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/12-088606a8", "0?sub-pincheck|s|1") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/12-088606a8", "0?disabletrunk|1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:4] Set("SIP/12-088606a8", "DIAL_NUMBER=099215415") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:5] Set("SIP/12-088606a8", "DIAL_TRUNK_OPTIONS=tr") in new stack</div><div> -- Executing [s@macro-dialout-trunk:6] Set("SIP/12-088606a8", "OUTBOUND_GROUP=OUT_1") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/12-088606a8", "0?nomax") in new stack</div><div> -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/12-088606a8", "0?chanfull") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/12-088606a8", "0?skipoutcid") in new stack</div><div> -- Executing [s@macro-dialout-trunk:10] Set("SIP/12-088606a8", "DIAL_TRUNK_OPTIONS=") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/12-088606a8", "outbound-callerid|1") in new stack</div><div> -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/12-088606a8", "0|SetCallerPres|") in new stack</div>
<div> -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/12-088606a8", "0|Set|REALCALLERIDNUM=12") in new stack</div><div> -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/12-088606a8", "1?normcid") in new stack</div>
<div> -- Goto (macro-outbound-callerid,s,6)</div><div> -- Executing [s@macro-outbound-callerid:6] Set("SIP/12-088606a8", "USEROUTCID=") in new stack</div><div> -- Executing [s@macro-outbound-callerid:7] Set("SIP/12-088606a8", "EMERGENCYCID=") in new stack</div>
<div> -- Executing [s@macro-outbound-callerid:8] Set("SIP/12-088606a8", "TRUNKOUTCID=<<font color="#ff0000">xxxxxxxxxx</font>>") in new stack</div><div> -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/12-088606a8", "1?trunkcid") in new stack</div>
<div> -- Goto (macro-outbound-callerid,s,12)</div><div> -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/12-088606a8", "1|Set|CALLERID(all)=<<font color="#ff0000">xxxxxxxx</font>>") in new stack</div>
<div> -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/12-088606a8", "0|Set|CALLERID(all)=") in new stack</div><div> -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/12-088606a8", "0|SetCallerPres|prohib_passed_screen") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/12-088606a8", "0|AGI|fixlocalprefix") in new stack</div><div> -- Executing [s@macro-dialout-trunk:13] Set("SIP/12-088606a8", "OUTNUM=099215415") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:14] Set("SIP/12-088606a8", "custom=SIP/<font color="#ff0000">xxxxxxxxx</font>") in new stack</div><div> -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/12-088606a8", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:16] Macro("SIP/12-088606a8", "dialout-trunk-predial-hook|") in new stack</div><div> -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/12-088606a8", "") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/12-088606a8", "0?bypass|1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/12-088606a8", "0?customtrunk") in new stack</div>
<div> -- Executing [s@macro-dialout-trunk:19] Dial("SIP/12-088606a8", "SIP/<font color="#ff0000">xxxxxxxx</font>/099215415|300|") in new stack</div><div> -- Called <font color="#ff0000">xxxxxxxxx</font>/099215415</div>
<div> == Begin MixMonitor Recording SIP/12-088606a8</div><div> -- SIP/<font color="#ff0000">xxxxxxxxx</font>-088646c8 is ringing</div><div> -- SIP/<font color="#ff0000">xxxxxxxxx</font>-088646c8 answered SIP/12-088606a8</div>
<div> -- Executing [h@macro-dialout-trunk:1] Macro("SIP/12-088606a8", "hangupcall|") in new stack</div><div> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/12-088606a8", "1?skiprg") in new stack</div>
<div> -- Goto (macro-hangupcall,s,4)</div><div> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/12-088606a8", "1?skipblkvm") in new stack</div><div> -- Goto (macro-hangupcall,s,7)</div><div> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/12-088606a8", "1?theend") in new stack</div>
<div> -- Goto (macro-hangupcall,s,9)</div><div> -- Executing [s@macro-hangupcall:9] Hangup("SIP/12-088606a8", "") in new stack</div><div> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/12-088606a8' in macro 'hangupcall'</div>
<div> == Spawn h extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/12-088606a8'</div><div> == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/12-088606a8' in macro 'dialout-trunk'</div>
<div> == Spawn extension (from-internal, 099215415, 4) exited non-zero on 'SIP/12-088606a8'</div><div> == MixMonitor close filestream</div><div> == End MixMonitor Recording SIP/12-088606a8</div><div><br></div>
<br>
<hr>From: <a href="mailto:gleidison.sampaio@hotmail.com" target="_blank">gleidison.sampaio@hotmail.com</a><div><br>To: <a href="mailto:asteriskbrasil@listas.asteriskbrasil.org" target="_blank">asteriskbrasil@listas.asteriskbrasil.org</a><br>
</div>Date: Thu, 1 Jul 2010 07:57:06 -0400<div><div></div><div><br>Subject: Re: [AsteriskBrasil] Elastix 1.6 + Linksys SPA 3102 MSG: The number you have dialed is not in service please try again<br><br>
Roger, bom dia<div><br></div><div>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...</div>
<div><br></div><div>PEER DETAILS:</div><div><br></div></div></div><div><div><div></div><div><div>disallow=all</div><div>allow=ulaw</div><div>canreinvite=no</div><div>context=from-trunk</div><div>dtmfmode=rfc2833</div>
<div>host=dynamic</div><div>incominglimit=1</div><div>nat=never</div><div>port=5061</div><div>qualify=yes</div><div>secret=xxxxxx</div><div>type=friend</div><div>username=xxxx</div><div><br></div><div>REGISTER STRING:</div>
<div><br></div><div>xxxx:xxxxx@10.x.x.x:5061/xxxx</div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><br><hr>Date: Thu, 1 Jul 2010 00:50:18 -0300<br>From: <a href="mailto:rogerwinter@gmail.com" target="_blank">rogerwinter@gmail.com</a><br>
To: <a href="mailto:asteriskbrasil@listas.asteriskbrasil.org" target="_blank">asteriskbrasil@listas.asteriskbrasil.org</a><br>Subject: Re: [AsteriskBrasil] Elastix 1.6 + Linksys SPA 3102 MSG: The number you have dialed is not in service please try again<br>
<br>Seu ramal "parece" estar com o context setado como "from-trunk"...<div>Deveria ser from-internal.. Dá uma conferida ae</div><div><br></div><div><br></div></div></div><div><br><br><div><div><div></div>
<div>Em 30 de junho de 2010 14:00, Gleidison Sampaio <span dir="ltr"><<a href="mailto:gleidison.sampaio@hotmail.com" target="_blank">gleidison.sampaio@hotmail.com</a>></span> escreveu:<br>
</div></div><blockquote style="border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div>
Boa tarde Srs,<div><br></div><div><div><div></div><div>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.<br>
<br><div>-- Executing [98201590@from-trunk:1] Set("SIP/12-b72087d0", "__FROM_DID=98201590") in new stack</div><div> -- Executing [98201590@from-trunk:2] NoOp("SIP/12-b72087d0", "Received an unknown call with DID set to 98201590") in new stack</div>
<div> -- Executing [98201590@from-trunk:3] Goto("SIP/12-b72087d0", "s|a2") in new stack</div><div> -- Goto (from-trunk,s,2)</div><div> -- Executing [s@from-trunk:2] Answer("SIP/12-b72087d0", "") in new stack</div>
<div> -- Executing [s@from-trunk:3] Wait("SIP/12-b72087d0", "2") in new stack</div></div></div><div>Really destroying SIP dialog '<a href="mailto:0cb41fb06d71b2e0385c4f3b2642409f@192.168.2.131" target="_blank">0cb41fb06d71b2e0385c4f3b2642409f@x</a>.x.x.x' Method: OPTIONS</div>
<div>
<div> -- Executing [s@from-trunk:4] Playback("SIP/12-b72087d0", "ss-noservice") in new stack</div><div> -- <SIP/12-b72087d0> Playing 'ss-noservice' (language 'en')</div></div>
<div>
REGISTER attempt 29 to <a href="mailto:4430170686@192.168.2.131" target="_blank">xxxxxxxxx@</a>x.x.x.x</div><div><div>Really destroying SIP dialog '<a href="mailto:6e03058055b63ec6034244496845dc41@127.0.0.1" target="_blank">6e03058055b63ec6034244496845dc41@127.0.0.1</a>' Method: REGISTER</div>
<div> -- Executing [s@from-trunk:5] SayAlpha("SIP/12-b72087d0", "98201590") in new stack</div><div> -- <SIP/12-b72087d0> Playing 'digits/9' (language 'en')</div><div> -- <SIP/12-b72087d0> Playing 'digits/8' (language 'en')</div>
<div> -- <SIP/12-b72087d0> Playing 'digits/2' (language 'en')</div><div> == Spawn extension (from-trunk, s, 5) exited non-zero on 'SIP/12-b72087d0'</div><div> -- Executing [h@from-trunk:1] Hangup("SIP/12-b72087d0", "") in new stack</div>
<div> == Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/12-b72087d0'</div></div><div>Really destroying SIP dialog '<a href="mailto:74f94492-a71b9e3c@192.168.2.222" target="_blank">74f94492-a71b9e3c@x</a>.x.x.x' Method: BYE</div>
<div><br></div><div><br></div><div><div><div> -- Executing [98201590@from-trunk:1] Set("SIP/12-b7209df8", "__FROM_DID=98201590") in new stack</div><div> -- Executing [98201590@from-trunk:2] NoOp("SIP/12-b7209df8", "Received an unknown call with DID set to 98201590") in new stack</div>
<div> -- Executing [98201590@from-trunk:3] Goto("SIP/12-b7209df8", "s|a2") in new stack</div><div> -- Goto (from-trunk,s,2)</div><div> -- Executing [s@from-trunk:2] Answer("SIP/12-b7209df8", "") in new stack</div>
<div> -- Executing [s@from-trunk:3] Wait("SIP/12-b7209df8", "2") in new stack</div><div>Really destroying SIP dialog '794742104b0f4274' Method: REGISTER</div><div> -- Executing [s@from-trunk:4] Playback("SIP/12-b7209df8", "ss-noservice") in new stack</div>
<div> -- <SIP/12-b7209df8> Playing 'ss-noservice' (language 'en')</div><div> == Spawn extension (from-trunk, s, 4) exited non-zero on 'SIP/12-b7209df8'</div><div> -- Executing [h@from-trunk:1] Hangup("SIP/12-b7209df8", "") in new stack</div>
<div> == Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/12-b7209df8'</div></div><div>Really destroying SIP dialog '<a href="mailto:2c3a477-d9bc2bc1@192.168.2.222" target="_blank">2c3a477-d9bc2bc1@x</a>.x.x.x' Method: BYE</div>
<div>
<div>Really destroying SIP dialog '<a href="mailto:6e03058055b63ec6034244496845dc41@127.0.0.1" target="_blank">6e03058055b63ec6034244496845dc41@127.0.0.1</a>' Method: REGISTER</div><div><br></div></div></div><div>
<br></div>
<div><br></div><div><br></div></div><div> <br><hr>VEJA TODOS OS SEUS EMAILS DE VÁRIAS CONTAS COM UM SÓ LOGIN. <a href="http://www.windowslive.com.br/public/tip.aspx/view/16?product=1&ocid=Hotmail:Live:Hotmail:Tagline:1x1:VEJATODOSO84:-" target="_blank">CLIQUE AQUI E VEJA COMO.</a></div>
</div><div>
<br>_______________________________________________<br>
KHOMP: qualidade em placas de E1, GSM, FXS e FXO para Asterisk.<br>
- Hardware com alta disponibilidade de recursos e qualidade KHOMP<br>
- Suporte técnico local qualificado e gratuito<br>
Conheça a linha completa de produtos KHOMP em <a href="http://www.khomp.com.br" target="_blank">www.khomp.com.br</a><br>
_______________________________________________<br>
Temos tudo para seu projeto VoIP com Asterisk!<br>
Descontos especiais para assinantes da AsteriskBrasil.org.<br>
Registre-se e receba um cupom exclusivo de desconto!<br>
Acesse agora <a href="http://www.voipmania.com.br" target="_blank">www.voipmania.com.br</a><br>
______________________________________________<br>
Lista de discussões AsteriskBrasil.org<br>
<a href="mailto:AsteriskBrasil@listas.asteriskbrasil.org" target="_blank">AsteriskBrasil@listas.asteriskbrasil.org</a><br>
<a href="http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil" target="_blank">http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil</a><br></div></blockquote></div><div><br><br clear="all">
<br>-- <br>----<br>
Roger Pitigliani<br><a href="mailto:rogerwinter@gmail.com" target="_blank">rogerwinter@gmail.com</a><br>msn: <a href="mailto:roger_pitigliani@hotmail.com" target="_blank">roger_pitigliani@hotmail.com</a><br>Gravataí - RS<br>
</div></div></div><div> <br><hr>PARA NAVEGAR COM MAIS PRIVACIDADE USE O INTERNET EXPLORER 8. <a href="http://www.microsoft.com/brasil/windows/internet-explorer/default.aspx?WT.mc_id=1633" target="_blank">INSTALE GRÁTIS.</a> <br>
</div><hr>O INTERNET EXPLORER 8 AJUDA VOCÊ A FICAR LONGE DOS VÍRUS. <a href="http://www.microsoft.com/brasil/windows/internet-explorer/features/stay-safer-online.aspx?tabid=1&catid=1&WT.mc_id=1632" target="_blank">DESCUBRA COMO.</a></div>
<br>_______________________________________________<br>
KHOMP: qualidade em placas de E1, GSM, FXS e FXO para Asterisk.<br>
- Hardware com alta disponibilidade de recursos e qualidade KHOMP<br>
- Suporte técnico local qualificado e gratuito<br>
Conheça a linha completa de produtos KHOMP em <a href="http://www.khomp.com.br" target="_blank">www.khomp.com.br</a><br>
_______________________________________________<br>
Temos tudo para seu projeto VoIP com Asterisk!<br>
Descontos especiais para assinantes da AsteriskBrasil.org.<br>
Registre-se e receba um cupom exclusivo de desconto!<br>
Acesse agora <a href="http://www.voipmania.com.br" target="_blank">www.voipmania.com.br</a><br>
______________________________________________<br>
Lista de discussões AsteriskBrasil.org<br>
<a href="mailto:AsteriskBrasil@listas.asteriskbrasil.org" target="_blank">AsteriskBrasil@listas.asteriskbrasil.org</a><br>
<a href="http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil" target="_blank">http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil</a><br></blockquote></div><br><br clear="all"><br>-- <br>----<br>
Roger Pitigliani<br><a href="mailto:rogerwinter@gmail.com" target="_blank">rogerwinter@gmail.com</a><br>msn: <a href="mailto:roger_pitigliani@hotmail.com" target="_blank">roger_pitigliani@hotmail.com</a><br>Gravataí - RS<br>