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.
<--- SIP read from TCP://10.200.132.97:29878 ---> INVITE sip:41...@avaya.com SIP/2.0 From: "Rafael Augusto - Telecom" <sip:38...@invalid.unknown.domain>;tag=8074e48ce16bdf112f24b3e81900 To: "41190" <sip:41...@avaya.com> Call-ID: 8074e48ce16bdf113f24b3e81900 CSeq: 1 INVITE Max-Forwards: 71 Route: <sip:10.200.50.133;lr;phase=terminating;transport=tcp> Record-Route: <sip:10.200.132.97;lr;transport=tcp> Via: SIP/2.0/TCP 10.200.132.97;branch=z9hG4bK8074e48ce16bdf114f24b3e81900 User-Agent: Avaya CM/R015x.02.0.947.3 Supported: timer, replaces, join, histinfo, 100rel Allow: INVITE, CANCEL, BYE, ACK, PRACK, SUBSCRIBE, NOTIFY, REFER, OPTIONS, INFO, PUBLISH Contact: "Rafael Augusto - Telecom" <sip:38...@10.200.132.97;transport=tcp> Session-Expires: 1200;refresher=uac Min-SE: 1200 P-Asserted-Identity: "Rafael Augusto - Telecom" <sip:38...@invalid.unknown.domain> Accept-Language: en Content-Type: application/sdp History-Info: <sip:41...@avaya.com>;index=1 History-Info: "41190" <sip:41...@avaya.com>;index=1.1 Alert-Info: <cid:internal@avaya.com>;avaya-cm-alert-type=internal Content-Length: 188 v=0 o=- 1 1 IN IP4 10.200.132.97 s=- c=IN IP4 10.200.96.145 b=AS:64 t=0 0 m=audio 2176 RTP/AVP 18 127 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 <-------------> --- (22 headers 10 lines) --- == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Sending to 10.200.132.97 : 5060 (no NAT) Using INVITE request as basis request - 8074e48ce16bdf113f24b3e81900 No user '38495' in SIP users list No matching peer for '38495' from '10.200.132.97:29878' Found RTP audio format 18 Found RTP audio format 127 Peer audio RTP is at port 10.200.96.145:2176 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 127 Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.200.96.145:2176 Looking for 41190 in default (domain avaya.com) <--- Reliably Transmitting (no NAT) to 10.200.132.97:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/TCP 10.200.132.97;branch=z9hG4bK8074e48ce16bdf114f24b3e81900;received=10.200.132.97 From: "Rafael Augusto - Telecom" <sip:38...@invalid.unknown.domain>;tag=8074e48ce16bdf112f24b3e81900 To: "41190" <sip:41...@avaya.com>;tag=as50d9a660 Call-ID: 8074e48ce16bdf113f24b3e81900 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> [May 14 09:56:15] NOTICE[30262]: chan_sip.c:17295 handle_request_invite: Call from '' to extension '41190' rejected because extension not found. Scheduling destruction of SIP dialog '8074e48ce16bdf113f24b3e81900' in 32000 ms (Method: INVITE) udppbxip01*CLI> <--- SIP read from TCP://10.200.132.97:29878 ---> ACK sip:41...@avaya.com SIP/2.0 From: "Rafael Augusto - Telecom" <sip:38...@invalid.unknown.domain>;tag=8074e48ce16bdf112f24b3e81900 To: "41190" <sip:41...@avaya.com>;tag=as50d9a660 Call-ID: 8074e48ce16bdf113f24b3e81900 Via: SIP/2.0/TCP 10.200.132.97;branch=z9hG4bK8074e48ce16bdf114f24b3e81900;received=10.200.132.97 CSeq: 1 ACK Max-Forwards: 70 Route: <sip:10.200.50.133;lr;phase=terminating;transport=tcp> User-Agent: Avaya CM/R015x.02.0.947.3 Content-Length: 0
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