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<DIV><FONT face=Arial size=2>nao deu pra entender por ai</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>pode ser "imcompatibilidade" de codec ou algo
parecido ..</FONT></DIV>
<DIV><FONT face=Arial size=2>talvez algum paremetro que esta errado ou faltando
no extension ou no softfone</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>como sao ramais internos</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>adicione as linhas </FONT></DIV>
<DIV><FONT face=Arial size=2>allow=ulaw</FONT></DIV>
<DIV><FONT face=Arial size=2>disallow=all</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>na configuraçao do seu ramal.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>se ainda nao funcionar</FONT></DIV>
<DIV><FONT face=Arial size=2>verifique se nao esta com DND ( do not disturb
)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>asterisk1*CLI> database show<BR>verifique a
lista das opcoes/ramal.</DIV></FONT>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>vc pode ver tambem a negociacao SIP atraves
do debug </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>asterisk1*CLI> sip debug<BR></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>ah detalhe .. pelo que eu sei o <A
href="mailto:asterisk@home">asterisk@home</A> se vc ligar pro seu proprio numero
( mesmo que o call waiting estiver ligado ) ele acusa o ramal como ocupado e
joga pro voicemail.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>espero ter ajudado,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Att,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Allan.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2></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=willians@ebr.com.br href="mailto:willians@ebr.com.br">Willians</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 11:17
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [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></FONT> </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="mailto:asterisk@home">asterisk@home</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="mailto:ast@home">ast@home</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="mailto:ast@home">ast@home</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="mailto:root@192.168.30.125's">root@192.168.30.125's</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="mailto:Asterisk@Home">Asterisk@Home</A><BR>-------------------------------------------------</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>For access to the <A
href="mailto:Asterisk@Home">Asterisk@Home</A> web GUI use this URL<BR><A
href="http://192.168.30.125">http://192.168.30.125</A></FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>For help on <A
href="mailto:Asterisk@Home">Asterisk@Home</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="mailto:markster@digium.com">markster@digium.com</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>
<P>
<HR>
E-mail classificado pelo Identificador de Spam Inteligente.<BR>Para alterar a
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href="http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=zyryz&_l=1,1135343933.84930.16058.cabue.terra.com.br,17293,Des15,Des15">Terra
Mail</A>
<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
<P>
<HR>
<P></P>No virus found in this incoming message.<BR>Checked by AVG Free
Edition.<BR>Version: 7.1.371 / Virus Database: 267.14.5/212 - Release Date:
23/12/2005<BR></BLOCKQUOTE></BODY></HTML>