Rafael, Para podermos entender melhor: -Qual o IP do Asterisk -Qual o IP da Avaya -Quem esta ligando pra quem?
Att, Guilherme Loch Góes 2010/5/14 Rafael Augusto <rafael_...@yahoo.com.br> > > 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 > > > > _______________________________________________ > KHOMP: qualidade em placas de E1, GSM, FXS e FXO para Asterisk. > - Hardware com alta disponibilidade de recursos e qualidade KHOMP > - Suporte técnico local qualificado e gratuito > Conheça a linha completa de produtos KHOMP em www.khomp.com.br > _______________________________________________ > Participe do I Encontro VoIPCenter, 08 a 10 de junho – Rio de Janeiro. > Área de exposição, palestras e cursos de VoIP, Asterisk e Convergência de > Redes. > http://www.encontrovoipcenter.com.br > ______________________________________________ > Lista de discussões AsteriskBrasil.org > AsteriskBrasil@listas.asteriskbrasil.org > http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
_______________________________________________ KHOMP: qualidade em placas de E1, GSM, FXS e FXO para Asterisk. - Hardware com alta disponibilidade de recursos e qualidade KHOMP - Suporte técnico local qualificado e gratuito Conheça a linha completa de produtos KHOMP em www.khomp.com.br _______________________________________________ Participe do I Encontro VoIPCenter, 08 a 10 de junho Rio de Janeiro. Área de exposição, palestras e cursos de VoIP, Asterisk e Convergência de Redes. http://www.encontrovoipcenter.com.br ______________________________________________ Lista de discussões AsteriskBrasil.org AsteriskBrasil@listas.asteriskbrasil.org http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil