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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Amigos, por favor, me desculpe eu interromper a discussão, mas
porque em assuntos complicados vocês trocam mil mensagens, e as minhas duvidas
que envio que são muito simples pois estou começando e ninguém responde?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Grato & Atenciosamente<o:p></o:p></span></b></p>
<p class=MsoNormal><span style='font-family:"Calibri","sans-serif";color:#1F497D'>Marllon
R. P. Castro<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>www.TuningParts.com.br<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>11 4063-9263 | 21 4063-9252 | 31 4063-9243 | 41 4063-9276<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>De:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asteriskbrasil-bounces@listas.asteriskbrasil.org
[mailto:asteriskbrasil-bounces@listas.asteriskbrasil.org] <b>Em nome de </b>Rafael
Augusto<br>
<b>Enviada em:</b> sexta-feira, 14 de maio de 2010 11:54<br>
<b>Para:</b> asteriskbrasil@listas.asteriskbrasil.org<br>
<b>Assunto:</b> [SPAM] [AsteriskBrasil] 7. Re: Integrando 1.6 com Avaya
Definity via SIP<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal>Caro Guilherme, desde de já agradeço pela ajuda. Segue o
debug, realmente está dando o 404, mas o contexto e o plano de discagem estão
corretos. Está estranho. Obrigado.<o:p></o:p></p>
</div>
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margin-left:3.75pt;margin-top:5.0pt;margin-bottom:5.0pt'>
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<p class=MsoNormal><--- SIP read from TCP://10.200.132.97:29878 ---><br>
INVITE sip:41190@avaya.com SIP/2.0<br>
From: "Rafael Augusto - Telecom"
<sip:38495@invalid.unknown.domain>;tag=8074e48ce16bdf112f24b3e81900<br>
To: "41190" <sip:41190@avaya.com><br>
Call-ID: 8074e48ce16bdf113f24b3e81900<br>
CSeq: 1 INVITE<br>
Max-Forwards: 71<br>
Route: <sip:10.200.50.133;lr;phase=terminating;transport=tcp><br>
Record-Route: <sip:10.200.132.97;lr;transport=tcp><br>
Via: SIP/2.0/TCP 10.200.132.97;branch=z9hG4bK8074e48ce16bdf114f24b3e81900<br>
User-Agent: Avaya CM/R015x.02.0.947.3<br>
Supported: timer, replaces, join, histinfo, 100rel<br>
Allow: INVITE, CANCEL, BYE, ACK, PRACK, SUBSCRIBE, NOTIFY, REFER, OPTIONS,
INFO, PUBLISH<br>
Contact: "Rafael Augusto - Telecom"
<sip:38495@10.200.132.97;transport=tcp><br>
Session-Expires: 1200;refresher=uac<br>
Min-SE: 1200<br>
P-Asserted-Identity: "Rafael Augusto - Telecom"
<sip:38495@invalid.unknown.domain><br>
Accept-Language: en<br>
Content-Type: application/sdp<br>
History-Info: <sip:41190@avaya.com>;index=1<br>
History-Info: "41190" <sip:41190@avaya.com>;index=1.1<br>
Alert-Info: <cid:internal@avaya.com>;avaya-cm-alert-type=internal<br>
Content-Length: 188<o:p></o:p></p>
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<p class=MsoNormal>v=0<br>
o=- 1 1 IN IP4 10.200.132.97<br>
s=-<br>
c=IN IP4 10.200.96.145<br>
b=AS:64<br>
t=0 0<br>
m=audio 2176 RTP/AVP 18 127<br>
a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>
a=rtpmap:127 telephone-event/8000<o:p></o:p></p>
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<p class=MsoNormal><-------------><br>
--- (22 headers 10 lines) ---<br>
== Using SIP RTP CoS mark 5<br>
== Using UDPTL CoS mark 5<br>
Sending to 10.200.132.97 : 5060 (no NAT)<br>
Using INVITE request as basis request - 8074e48ce16bdf113f24b3e81900<br>
No user '38495' in SIP users list<br>
No matching peer for '38495' from '10.200.132.97:29878'<br>
Found RTP audio format 18<br>
Found RTP audio format 127<br>
Peer audio RTP is at port 10.200.96.145:2176<br>
Found audio description format G729 for ID 18<br>
Found audio description format telephone-event for ID 127<br>
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100
(g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)<br>
Peer audio RTP is at port 10.200.96.145:2176<br>
Looking for 41190 in default (domain avaya.com)<o:p></o:p></p>
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<div>
<p class=MsoNormal><--- Reliably Transmitting (no NAT) to
10.200.132.97:5060 ---><br>
SIP/2.0 404 Not Found<br>
Via: SIP/2.0/TCP
10.200.132.97;branch=z9hG4bK8074e48ce16bdf114f24b3e81900;received=10.200.132.97<br>
From: "Rafael Augusto - Telecom"
<sip:38495@invalid.unknown.domain>;tag=8074e48ce16bdf112f24b3e81900<br>
To: "41190" <sip:41190@avaya.com>;tag=as50d9a660<br>
Call-ID: 8074e48ce16bdf113f24b3e81900<br>
CSeq: 1 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces, timer<br>
Content-Length: 0<o:p></o:p></p>
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<p class=MsoNormal><br>
<------------><br>
[May 14 09:56:15] NOTICE[30262]: chan_sip.c:17295 handle_request_invite: Call
from '' to extension '41190' rejected because extension not found.<br>
Scheduling destruction of SIP dialog '8074e48ce16bdf113f24b3e81900' in 32000
ms (Method: INVITE)<br>
udppbxip01*CLI><br>
<--- SIP read from TCP://10.200.132.97:29878 ---><br>
ACK sip:41190@avaya.com SIP/2.0<br>
From: "Rafael Augusto - Telecom"
<sip:38495@invalid.unknown.domain>;tag=8074e48ce16bdf112f24b3e81900<br>
To: "41190" <sip:41190@avaya.com>;tag=as50d9a660<br>
Call-ID: 8074e48ce16bdf113f24b3e81900<br>
Via: SIP/2.0/TCP 10.200.132.97;branch=z9hG4bK8074e48ce16bdf114f24b3e81900;received=10.200.132.97<br>
CSeq: 1 ACK<br>
Max-Forwards: 70<br>
Route: <sip:10.200.50.133;lr;phase=terminating;transport=tcp><br>
User-Agent: Avaya CM/R015x.02.0.947.3<br>
Content-Length: 0<o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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</blockquote>
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