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<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></BODY></HTML>