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<DIV><FONT face=Arial size=2>Alterei a configuração ,mas continua com o mesmo
problema</FONT></DIV>
<DIV> </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=sebastiaorocha@interlinksistemas.com.br
href="mailto:sebastiaorocha@interlinksistemas.com.br">Sebastiao Rocha</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> Thursday, August 21, 2008 10:14
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [AsteriskBrasil] Problemas
com ocupado</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>Se usar nat=yes, o canreinvite=no deve ser
especificado, não se pode usar os dois juntos.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>CanReinvite é usado para que o trafego de voz não
passe pelo servidor, ele será feito via RTP entre os ramais, "Peer to
Peer" ou IP para IP, como preferir, o NAT não permite que seja estabelecida
uma conexão assim.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </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=eduardo@impactovoip.com.br
href="mailto:eduardo@impactovoip.com.br">Eduardo_Impacto</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> Thursday, August 21, 2008 9:43
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [AsteriskBrasil] Problemas com
ocupado</DIV>
<DIV><BR></DIV><FONT face=Arial size=2>
<DIV>Bom dia Lista, estou com úm ramal sip só dando ocupado, todas as
chamadas que eu realizo elas dão ocupado...abaixo vai o debug, se alguem
poder ajudar fico grato</DIV>
<DIV> </DIV>
<DIV>[2454]<BR>type=friend<BR>username=2454<BR>accountcode=2454<BR>regexten=2454<BR>callerid=2401<BR>amaflags=billing<BR>secret=xxxxxxxxxxx<BR>nat=yes<BR>dtmfmode=RFC2833<BR>qualify=yes<BR>canreinvite=yes<BR>disallow=all<BR>allow=ulaw<BR>allow=alaw<BR>allow=gsm<BR>allow=g729<BR>host=dynamic<BR>context=a2billing<BR>regseconds=0<BR>cancallforward=yes<BR><BR>---<BR>Destroying
call <A
href="mailto:'26198a1069cd6c66171b81860ebf9c7a@201.48.251.15'">'26198a1069cd6c66171b81860ebf9c7a@201.48.251.15'</A><BR>Retransmitting
#4 (NAT) to 201.22.164.167:5060:<BR>OPTIONS sip:201.22.164.167
SIP/2.0<BR>Via: SIP/2.0/UDP
201.48.251.15:5060;branch=z9hG4bK2232d31a;rport<BR>From: "Unknown"
<sip:Unknown@201.48.251.15>;tag=as1b92be04<BR>To:
<sip:201.22.164.167><BR>Contact:
<sip:Unknown@201.48.251.15><BR>Call-ID: <A
href="mailto:390a89934b46b54214b5943e6f33424f@201.48.251.15">390a89934b46b54214b5943e6f33424f@201.48.251.15</A><BR>CSeq:
102 OPTIONS<BR>User-Agent: Impacto Voip Pbx<BR>Max-Forwards: 70<BR>Date:
Thu, 21 Aug 2008 12:37:19 GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY<BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR>---<BR>Destroying call <A
href="mailto:'390a89934b46b54214b5943e6f33424f@201.48.251.15'">'390a89934b46b54214b5943e6f33424f@201.48.251.15'</A><BR>asterisk1*CLI><BR><--
SIP read from 201.22.164.167:59317:<BR>INVITE sip:06230911858@201.48.251.15
SIP/2.0<BR>Via: SIP/2.0/UDP
192.168.0.104:44598;branch=z9hG4bK1440404089;rport<BR>Route:
<sip:201.48.251.15:5060;lr><BR>From: "Claudio"
<sip:2454@201.48.251.15>;tag=1407274989<BR>To:
<sip:06230911858@201.48.251.15><BR>Call-ID: <A
href="mailto:1762016748-44598-10@192.168.0.104">1762016748-44598-10@192.168.0.104</A><BR>CSeq:
90 INVITE<BR>Contact: <sip:2454@192.168.0.104:44598><BR>Max-Forwards:
70<BR>Supported: replaces, path, timer<BR>User-Agent: Grandstream
GXW-4004 V1.1A 1.0.0.67<BR>Allow: INVITE, ACK, OPTIONS, CANCEL, BYE,
SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE<BR>Content-Type:
application/sdp<BR>Accept: application/sdp,
application/dtmf-relay<BR>Content-Length: 297</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=2454 8002 8000 IN IP4 192.168.0.104<BR>s=SIP Call<BR>c=IN IP4
192.168.0.104<BR>t=0 0<BR>m=audio 18038 RTP/AVP 18 4 0 8
101<BR>a=sendrecv<BR>a=rtpmap:18 G729/8000<BR>a=ptime:20<BR>a=rtpmap:4
G723/8000<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16,32-36,54</DIV>
<DIV> </DIV>
<DIV>--- (15 headers 14 lines)---<BR>Using INVITE request as basis request -
<A
href="mailto:1762016748-44598-10@192.168.0.104">1762016748-44598-10@192.168.0.104</A><BR>Sending
to 192.168.0.104 : 44598 (NAT)<BR>Reliably Transmitting (NAT) to
201.22.164.167:59317:<BR>SIP/2.0 407 Proxy Authentication Required<BR>Via:
SIP/2.0/UDP
192.168.0.104:44598;branch=z9hG4bK1440404089;received=201.22.164.167;rport=59317<BR>From:
"Claudio" <sip:2454@201.48.251.15>;tag=1407274989<BR>To:
<sip:06230911858@201.48.251.15>;tag=as75d3af08<BR>Call-ID: <A
href="mailto:1762016748-44598-10@192.168.0.104">1762016748-44598-10@192.168.0.104</A><BR>CSeq:
90 INVITE<BR>User-Agent: Impacto Voip Pbx<BR>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<BR>Contact:
<sip:06230911858@201.48.251.15><BR>Proxy-Authenticate: Digest
algorithm=MD5, realm="asterisk", nonce="72cfe697"<BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR>---<BR>Scheduling destruction of call <A
href="mailto:'1762016748-44598-10@192.168.0.104'">'1762016748-44598-10@192.168.0.104'</A>
in 15000 ms<BR>Found user '2454'<BR>asterisk1*CLI><BR><-- SIP read
from 201.22.164.167:59317:<BR>ACK sip:06230911858@201.48.251.15
SIP/2.0<BR>Via: SIP/2.0/UDP
192.168.0.104:44598;branch=z9hG4bK1440404089;rport<BR>Route:
<sip:201.48.251.15:5060;lr><BR>From: "Claudio"
<sip:2454@201.48.251.15>;tag=1407274989<BR>To:
<sip:06230911858@201.48.251.15>;tag=as75d3af08<BR>Call-ID: <A
href="mailto:1762016748-44598-10@192.168.0.104">1762016748-44598-10@192.168.0.104</A><BR>CSeq:
90 ACK<BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR>--- (8 headers 0 lines)---<BR>asterisk1*CLI><BR><-- SIP read
from 201.22.164.167:59317:<BR>INVITE sip:06230911858@201.48.251.15
SIP/2.0<BR>Via: SIP/2.0/UDP
192.168.0.104:44598;branch=z9hG4bK1021032546;rport<BR>Route:
<sip:201.48.251.15:5060;lr><BR>From: "Claudio"
<sip:2454@201.48.251.15>;tag=1407274989<BR>To:
<sip:06230911858@201.48.251.15><BR>Call-ID: <A
href="mailto:1762016748-44598-10@192.168.0.104">1762016748-44598-10@192.168.0.104</A><BR>CSeq:
91 INVITE<BR>Contact:
<sip:2454@192.168.0.104:44598><BR>Proxy-Authorization: Digest
username="2454", realm="asterisk", nonce="72cfe697",
uri="sip:06230911858@201.48.251.15",
response="d223043cc27813ce35691920977491c0", algorithm=MD5<BR>Max-Forwards:
70<BR>Supported: replaces, path, timer<BR>User-Agent: Grandstream
GXW-4004 V1.1A 1.0.0.67<BR>Allow: INVITE, ACK, OPTIONS, CANCEL, BYE,
SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE<BR>Content-Type:
application/sdp<BR>Accept: application/sdp,
application/dtmf-relay<BR>Content-Length: 297</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=2454 8002 8000 IN IP4 192.168.0.104<BR>s=SIP Call<BR>c=IN IP4
192.168.0.104<BR>t=0 0<BR>m=audio 18038 RTP/AVP 18 4 0 8
101<BR>a=sendrecv<BR>a=rtpmap:18 G729/8000<BR>a=ptime:20<BR>a=rtpmap:4
G723/8000<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16,32-36,54</DIV>
<DIV> </DIV>
<DIV>--- (16 headers 14 lines)---<BR>Using INVITE request as basis request -
<A
href="mailto:1762016748-44598-10@192.168.0.104">1762016748-44598-10@192.168.0.104</A><BR>Sending
to 192.168.0.104 : 44598 (NAT)<BR>Found user '2454'<BR>Found RTP audio
format 18<BR>Found RTP audio format 4<BR>Found RTP audio format 0<BR>Found
RTP audio format 8<BR>Found RTP audio format 101<BR>Peer audio RTP is at
port 192.168.0.104:18038<BR>Found description format G729<BR>Found
description format G723<BR>Found description format PCMU<BR>Found
description format PCMA<BR>Found description format
telephone-event<BR>Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer -
audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c
(ulaw|alaw|g729)<BR>Non-codec capabilities: us - 0x1 (telephone-event), peer
- 0x1 (telephone-event), combined - 0x1 (telephone-event)<BR>Looking for
06230911858 in a2billing (domain 201.48.251.15)<BR>Reliably Transmitting
(NAT) to 201.22.164.167:59317:<BR>SIP/2.0 404 Not Found<BR>Via: SIP/2.0/UDP
192.168.0.104:44598;branch=z9hG4bK1021032546;received=201.22.164.167;rport=59317<BR>From:
"Claudio" <sip:2454@201.48.251.15>;tag=1407274989<BR>To:
<sip:06230911858@201.48.251.15>;tag=as75d3af08<BR>Call-ID: <A
href="mailto:1762016748-44598-10@192.168.0.104">1762016748-44598-10@192.168.0.104</A><BR>CSeq:
91 INVITE<BR>User-Agent: Impacto Voip Pbx<BR>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<BR>Contact:
<sip:06230911858@201.48.251.15><BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR>---<BR>asterisk1*CLI><BR><-- SIP read from
201.22.164.167:59317:<BR>ACK sip:06230911858@201.48.251.15 SIP/2.0<BR>Via:
SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1021032546;rport<BR>Route:
<sip:201.48.251.15:5060;lr><BR>From: "Claudio"
<sip:2454@201.48.251.15>;tag=1407274989<BR>To:
<sip:06230911858@201.48.251.15>;tag=as75d3af08<BR>Call-ID: <A
href="mailto:1762016748-44598-10@192.168.0.104">1762016748-44598-10@192.168.0.104</A><BR>CSeq:
91 ACK<BR>Content-Length: 0<BR></FONT></DIV>
<DIV><FONT face=Arial size=2>Eduardo de Sousa <BR>Departamento
Comercial</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>MSN:atendimento@impactovoip.com.br<BR>Impacto
Voip Tecnologia e Teleinformática <BR><A
href="http://www.impactovoip.com.br">www.impactovoip.com.br</A><BR>Fone:
(62) 4053-8840 - 9651-4660<BR> <BR></FONT></DIV>
<P>
<HR>
<P></P>_______________________________________________<BR>Compre uma
camiseta da AsteriskBrasil.org!<BR>http://www.voipmania.com.br<BR><BR>Acesse
o canal IRC de discussão sobre Asterisk em Português Brasileiro na rede
Freenode.net:
#asterisk-br<BR>_______________________________________________<BR>Lista de
discussões
AsteriskBrasil.org<BR>AsteriskBrasil@listas.asteriskbrasil.org<BR>http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil</BLOCKQUOTE>
<P>
<HR>
<P></P>_______________________________________________<BR>Compre uma camiseta
da AsteriskBrasil.org!<BR>http://www.voipmania.com.br<BR><BR>Acesse o canal
IRC de discussão sobre Asterisk em Português Brasileiro na rede Freenode.net:
#asterisk-br<BR>_______________________________________________<BR>Lista de
discussões
AsteriskBrasil.org<BR>AsteriskBrasil@listas.asteriskbrasil.org<BR>http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil<BR><BR>__________
Informação do NOD32 IMON 3371 (20080820) __________<BR><BR>Esta mensagem foi
verificada pelo NOD32 sistema
antivírus<BR>http://www.eset.com.br<BR><BR></BLOCKQUOTE></BODY></HTML>