<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;"><DIV>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.</DIV>
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<DIV class=plainMail><--- 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</DIV>
<DIV class=plainMail>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</DIV>
<DIV class=plainMail><-------------><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)</DIV>
<DIV class=plainMail><--- 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</DIV>
<DIV class=plainMail><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</DIV>
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