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<DIV><FONT face=Arial size=2>Obrigado amigo, funcionou....</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Willians</FONT></DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=jose.leitao@oi.com.br href="mailto:jose.leitao@oi.com.br">Jose P.
Leitao</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A title=frederico@lufrenne.com.br
href="mailto:frederico@lufrenne.com.br">frederico@lufrenne.com.br</A> ; <A
title=asteriskbrasil@listas.asteriskbrasil.org
href="mailto:asteriskbrasil@listas.asteriskbrasil.org">asteriskbrasil@listas.asteriskbrasil.org</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, December 23, 2005 4:12
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [AsteriskBrasil] Sip´s não
falam entre si no <A href="mailto:asterisk@home">asterisk@home</A></DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>Sugiro verificarem quais os codecs estão
permitidos para os respectivos ramais e se os os mesmos estão habilitados nos
clientes, ative o debug do sip na interface CLI, que ele irá informar o motivo
da não conexão.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>SDS</FONT></DIV>
<DIV><FONT face=Arial size=2>José Leitão</FONT></DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=redskin_tupi@yahoo.com
href="mailto:redskin_tupi@yahoo.com">Frederico Simões</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asteriskbrasil@listas.asteriskbrasil.org
href="mailto:asteriskbrasil@listas.asteriskbrasil.org">asteriskbrasil@listas.asteriskbrasil.org</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, December 23, 2005 1:54
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [AsteriskBrasil] Sip´s não
falam entre si no <A href="mailto:asterisk@home">asterisk@home</A></DIV>
<DIV><BR></DIV>
<DIV id=RTEContent>Estou com um problema parecido...</DIV>
<DIV> </DIV>
<DIV>O meu fala de X-lite para X-lite... so que não fala de PAP para
PAP!!!</DIV>
<DIV> </DIV>
<DIV>Se alguem poder solucionar </DIV>
<DIV> </DIV>
<DIV>[ext-local]<BR>include => ext-local-custom<BR>exten =>
10001,1,Macro(exten-vm,novm,10001)<BR>exten =>
10002,1,Macro(exten-vm,novm,10002)<BR>exten =>
1001,1,Macro(exten-vm,1001@default,1001)<BR>exten =>
${VM_PREFIX}1001,1,Macro(vm,1001)<BR>exten =>
1002,1,Macro(exten-vm,1002@default,1002)<BR>exten =>
${VM_PREFIX}1002,1,Macro(vm,1002)<BR>exten =>
10021,1,Macro(exten-vm,10021@default,10021)<BR>exten =>
${VM_PREFIX}10021,1,Macro(vm,10021)<BR>exten =>
10022,1,Macro(exten-vm,10022@default,10022)<BR>exten =>
${VM_PREFIX}10022,1,Macro(vm,10022)<BR>exten =>
1003,1,Macro(exten-vm,1003@default,1003)<BR>exten =>
${VM_PREFIX}1003,1,Macro(vm,1003)<BR>exten =>
1004,1,Macro(exten-vm,1004@default,1004)<BR>exten =>
${VM_PREFIX}1004,1,Macro(vm,1004)<BR>exten =>
1005,1,Macro(exten-vm,1005@default,1005)<BR>exten =>
${VM_PREFIX}1005,1,Macro(vm,1005)<BR>exten =>
11001,1,Macro(exten-vm,novm,11001)<BR>exten =>
11002,1,Macro(exten-vm,novm,11002)<BR>exten =>
12001,1,Macro(exten-vm,novm,12001)<BR>exten =>
3001,1,Macro(exten-vm,novm,3001)<BR>exten =>
3002,1,Macro(exten-vm,novm,3002)<BR>exten =>
4001,1,Macro(exten-vm,novm,4001)<BR>exten =>
4002,1,Macro(exten-vm,novm,4002)<BR>exten =>
4011,1,Macro(exten-vm,novm,4011)<BR>exten =>
4012,1,Macro(exten-vm,novm,4012)<BR>exten =>
4021,1,Macro(exten-vm,novm,4021)<BR></DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>[macro-exten-vm]<BR>exten => s,1,Macro(user-callerid)<BR>exten =>
s,2,Setvar(FROMCONTEXT=exten-vm)<BR>exten =>
s,3,Macro(record-enable,${ARG2},IN)<BR>exten =>
s,4,Macro(dial,${RINGTIMER},${DIAL_OPTIONS},${ARG2})<BR>exten =>
s,5,GotoIf($[${CHANNEL:0:5} = Local]?s-${DIALSTATUS},1) ; if the channel is
Local, then do not go to voicemail. This is primarily to avoid vm for
call-forwarded extensions in ring groups<BR>exten => s,6,GotoIf($[${ARG1}
= novm]?s-${DIALSTATUS},1) ; no voicemail in use for this extension<BR>exten
=> s,7,NoOp(Sending to Voicemail box ${ARG2})<BR>exten =>
s,8,Macro(vm,${ARG1},${DIALSTATUS})<BR>exten => s-BUSY,1,NoOp(Extension
is reporting BUSY and has no Voicemail)<BR>exten =>
s-BUSY,2,Busy()<BR>exten => s-BUSY,3,Wait(60)<BR>exten =>
s-BUSY,4,NoOp()<BR>exten => _s-.,1,Congestion()<BR></DIV>
<DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV><FONT face=Arial size=2>Bom dia Asteriskers;</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sempre trabalhei com o Asterisk instalado ´no
braço´. Porém resolvi testar o <A
href="http://us.f309.mail.yahoo.com/ym/Compose?To=asterisk@home"
target=_blank><FONT color=#003399>asterisk@home</FONT></A> e estou com um
probleminha básico.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Configurei dois ramais sip com o x-lite. Eles
estão logando e o ecotest está ok.</FONT></DIV>
<DIV><FONT face=Arial size=2>Porém quando tento falar de um sip (1234)
para outro (2000) aparece a menságem no x-lite:(</FONT> <FONT
face=Arial size=2>Call failed 486 Busy Here ).</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>O <A
href="http://us.f309.mail.yahoo.com/ym/Compose?To=ast@home"
target=_blank><FONT color=#003399>ast@home</FONT></A> está em uma rede
interna juntamente com os outros softphones, exluindo-se a possibilidade
de ser uma dificuldade em passar pelo NAT.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Existe algo a mais que tem que ser feito,
além de se configurar os ramais no <A
href="http://us.f309.mail.yahoo.com/ym/Compose?To=ast@home"
target=_blank><FONT color=#003399>ast@home</FONT></A> para funcionar?
</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Abaixo o log do que acontece no momento que
ligo para outro sip:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Desde já agradeço a atenção de
todos.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Willians Dias</FONT></DIV>
<DIV><FONT face=Arial size=2>Vitória E.S.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>login as: root<BR><A
href="http://us.f309.mail.yahoo.com/ym/Compose?To=root@192.168.30.125's"
target=_blank><FONT color=#003399>root@192.168.30.125's</FONT></A>
password:<BR>Last login: Thu Jan 6 19:27:39 2005 from
192.168.30.161</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Welcome to <A
href="http://us.f309.mail.yahoo.com/ym/Compose?To=Asterisk@Home"
target=_blank><FONT
color=#003399>Asterisk@Home</FONT></A><BR>-------------------------------------------------</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>For access to the <A
href="http://us.f309.mail.yahoo.com/ym/Compose?To=Asterisk@Home"
target=_blank><FONT color=#003399>Asterisk@Home</FONT></A> web GUI use
this URL<BR><A onclick="return ShowLinkWarning()"
href="http://192.168.30.125/" target=_blank><FONT
color=#003399>http://192.168.30.125</FONT></A></FONT></DIV>
<DIV><FONT color=#003399></FONT> </DIV>
<DIV><FONT face=Arial size=2>For help on <A
href="http://us.f309.mail.yahoo.com/ym/Compose?To=Asterisk@Home"
target=_blank><FONT color=#003399>Asterisk@Home</FONT></A> commands you
can use from this<BR>command shell type help-aah.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>[root@asterisk1 ~]# asterisk -r<BR>Asterisk
1.2.1, Copyright (C) 1999 - 2005 Digium.<BR>Written by Mark Spencer <<A
href="http://us.f309.mail.yahoo.com/ym/Compose?To=markster@digium.com"
target=_blank><FONT
color=#003399>markster@digium.com</FONT></A>><BR>=========================================================================<BR>Connected
to Asterisk 1.2.1 currently running on asterisk1 (pid = 2381)<BR>Verbosity
is at least 3<BR> -- Executing Macro("SIP/2000-7e61",
"exten-vm|novm|1234") in new stack<BR> -- Executing
Macro("SIP/2000-7e61", "user-callerid") in new stack<BR>
-- Executing DBget("SIP/2000-7e61", "AMPUSER=DEVICE/2000/user") in new
stack<BR> -- DBget: varname=AMPUSER, family=DEVICE,
key=2000/user<BR> -- DBget: set variable AMPUSER to
2000<BR> -- Executing DBget("SIP/2000-7e61",
"AMPUSERCIDNAME=AMPUSER/2000/cidname") in new stack<BR>
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER,
key=2000/cidname<BR> -- DBget: set variable
AMPUSERCIDNAME to Willians<BR> -- Executing
GotoIf("SIP/2000-7e61", "0?5") in new stack<BR> --
Executing SetCallerID("SIP/2000-7e61", ""Willians" <2000>") in new
stack<BR> -- Executing NoOp("SIP/2000-7e61", "Using
CallerID "Willians" <2000>") in new stack<BR> --
Executing SetVar("SIP/2000-7e61", "FROMCONTEXT=exten-vm") in new
stack<BR> -- Executing Macro("SIP/2000-7e61",
"record-enable|1234|IN") in new stack<BR> -- Executing
GotoIf("SIP/2000-7e61", "0 > 0?2:4") in new stack<BR>
-- Goto (macro-record-enable,s,4)<BR> -- Executing
AGI("SIP/2000-7e61", "recordingcheck|20050106-193009|1105057809.8") in new
stack<BR> -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck<BR>
recordingcheck|20050106-193009|1105057809.8: Inbound recording not
enabled<BR> -- AGI Script recordingcheck completed,
returning 0<BR> -- Executing NoOp("SIP/2000-7e61", "No
recording needed") in new stack<BR> -- Executing
Macro("SIP/2000-7e61", "dial|49|tr|1234") in new
stack<BR> -- Executing GotoIf("SIP/2000-7e61", "0?4:2")
in new stack<BR> -- Goto
(macro-dial,s,2)<BR> -- Executing
GotoIf("SIP/2000-7e61", "0?5:4") in new stack<BR> --
Goto (macro-dial,s,4)<BR> -- Executing
AGI("SIP/2000-7e61", "dialparties.agi") in new stack<BR>
-- Launched AGI Script
/var/lib/asterisk/agi-bin/dialparties.agi<BR> -- AGI
Script dialparties.agi completed, returning 0<BR> --
Executing NoOp("SIP/2000-7e61", "Returned from dialparties with no
extensions to call") in new stack<BR> -- Executing
SetVar("SIP/2000-7e61", "DIALSTATUS=BUSY") in new
stack<BR> -- Executing GotoIf("SIP/2000-7e61",
"0?s-BUSY|1") in new stack<BR> -- Executing
GotoIf("SIP/2000-7e61", "1?s-BUSY|1") in new stack<BR>
-- Goto (macro-exten-vm,s-BUSY,1)<BR> -- Executing
NoOp("SIP/2000-7e61", "Extension is reporting BUSY and has no Voicemail")
in new stack<BR> -- Executing Busy("SIP/2000-7e61", "")
in new stack<BR> == Spawn extension (macro-exten-vm, s-BUSY, 2)
exited non-zero on 'SIP/2000-7e61' in macro 'exten-vm'<BR> == Spawn
extension (from-internal, 1234, 1) exited non-zero on
'SIP/2000-7e61'<BR> -- Executing Macro("SIP/2000-7e61",
"hangupcall") in new stack<BR> -- Executing
ResetCDR("SIP/2000-7e61", "w") in new stack<BR> --
Executing NoCDR("SIP/2000-7e61", "") in new stack<BR> --
Executing Wait("SIP/2000-7e61", "5") in new stack<BR> == Spawn
extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/2000-7e61' in
macro 'hangupcall'<BR> == Spawn extension (from-internal, h, 1)
exited non-zero on
'SIP/2000-7e61'<BR>asterisk1*CLI><BR></FONT></DIV></BLOCKQUOTE></XBODY><!-- type = text --></DIV>
<P>
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<HR>
<P></P>_______________________________________________<BR>LIsta de
discussões
AsteriskBrasil.org<BR>AsteriskBrasil@listas.asteriskbrasil.org<BR>http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil<BR><BR>_______________________________________________<BR>Acesse
o wiki AsteriskBrasil.org:<BR>http://www.asteriskbrasil.org</BLOCKQUOTE>
<P>
<HR>
<P></P>_______________________________________________<BR>LIsta de discussões
AsteriskBrasil.org<BR>AsteriskBrasil@listas.asteriskbrasil.org<BR>http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil<BR><BR>_______________________________________________<BR>Acesse
o wiki
AsteriskBrasil.org:<BR>http://www.asteriskbrasil.org</BLOCKQUOTE></BODY></HTML>