Olá a todos, obrigado pela ajuda. Eu vou seguir os seus conselhos, mas alguém ja teve esse problema?
É estranho, pq eu consigo fazer ligação, mas não consigo receber ligação vindo do Asterisk. Abraços. Em 16/03/2017, Alex Freitas<aledefreitas...@gmail.com> escreveu: > segue tutorial > > > > 2017-03-16 14:13 GMT-03:00 Luciano Cavalcante Souza <lucin...@gmail.com>: > >> verifica os tutorias da loja mundi >> >> >> *Sds,* >> >> *Luciano Cavalcante Souza* >> *Tecnólogo em Gestão da Tecnologia da Informação* >> *Mobile: + 55 79 98814.5895 <(79)%2098814-5895>(vivo)* >> *e-mail: lucin...@gmail.com <lucin...@gmail.com> * >> *Perfil no Linkdin <https://www.linkedin.com/in/luciano-souza-28240035>* >> >> *Sobre o Google Apps: Google Apps <https://goo.gl/CngU34>* >> >> Concentre-se nos pontos FORTES, reconheça as FRAQUEZAS, agarre as >> OPORTUNIDADES e proteja-se contra as AMEAÇAS. >> >> 2017-03-16 12:00 GMT-03:00 Vitor Mazuco <vitor.maz...@gmail.com>: >> >>> Ola a todos! >>> >>> Estou com um problema de ligação, não estou conseguindo receber >>> ligações de meu ASTERISK para o meu FXO Grandstream. >>> >>> Ele dá erro de "Forbidden" from" conforme as msg abaixo. >>> >>> Já tentei de tudo, mas nao acho o problema. >>> >>> Lembrando que eu uso um LOAD BALANCE nesse Grandstream para fazer o >>> balanceador de rede. >>> >>> Seria esse um problema de NAT/Firewall? >>> >>> Obrigado quem puder me ajudar. >>> >>> Log na CLI: >>> >>> Using SIP RTP CoS mark 5 >>> -- Executing [27100@ramais:1] MixMonitor("SIP/2000-0000bd8b", >>> "/media/HDExterno/gravacoes/feitas/APTO/27100/1489675579.120546.wav") >>> in new stack >>> -- Executing [27100@ramais:2] Dial("SIP/2000-0000bd8b", >>> "SIP/136/100,60,tT") in new stack >>> == Begin MixMonitor Recording SIP/2000-0000bd8b >>> == Using SIP RTP CoS mark 5 >>> -- Called SIP/136/100 >>> [2017-03-16 11:46:19] WARNING[1554][C-000098b9]: chan_sip.c:23843 >>> handle_response_invite: Received response: "Forbidden" from >>> '<sip:2000@192.168.25.24:5089>;tag=as57804b2e' >>> == Everyone is busy/congested at this time (1:0/0/1) >>> -- Auto fallthrough, channel 'SIP/2000-0000bd8b' status is >>> 'CHANUNAVAIL' >>> == MixMonitor close filestream (mixed) >>> == End MixMonitor Recording SIP/2000-0000bd8b >>> >>> >>> SIP Debuug: >>> >>> Called SIP/136/100 >>> >>> <--- SIP read from UDP:192.168.25.169:3329 ---> >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP 192.168.25.24:5089;branch=z9hG4bK4433efa4;rport=5089 >>> From: <sip:2000@192.168.25.24:5089>;tag=as62bede9e >>> To: <sip:100@192.168.25.169> >>> Call-ID: 692c5d293d0fae0853872d6a3206af86@192.168.25.24:5089 >>> CSeq: 102 INVITE >>> Supported: replaces, path, timer, eventlist >>> User-Agent: Grandstream HT-503 V2.0A 1.0.14.1 chip V2.2 >>> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, >>> REFER, >>> UPDATE >>> Content-Length: 0 >>> >>> <-------------> >>> --- (10 headers 0 lines) --- >>> >>> <--- SIP read from UDP:192.168.25.169:3329 ---> >>> SIP/2.0 403 Forbidden >>> Via: SIP/2.0/UDP 192.168.25.24:5089;branch=z9hG4bK4433efa4;rport=5089 >>> From: <sip:2000@192.168.25.24:5089>;tag=as62bede9e >>> To: <sip:100@192.168.25.169>;tag=1820807938 >>> Call-ID: 692c5d293d0fae0853872d6a3206af86@192.168.25.24:5089 >>> CSeq: 102 INVITE >>> Supported: replaces, path, timer, eventlist >>> User-Agent: Grandstream HT-503 V2.0A 1.0.14.1 chip V2.2 >>> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, >>> REFER, >>> UPDATE >>> Content-Length: 0 >>> >>> <-------------> >>> --- (10 headers 0 lines) --- >>> Transmitting (NAT) to 192.168.25.169:3329: >>> ACK sip:100@192.168.25.169 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.25.24:5089;branch=z9hG4bK4433efa4;rport >>> Max-Forwards: 70 >>> From: <sip:2000@192.168.25.24:5089>;tag=as62bede9e >>> To: <sip:100@192.168.25.169>;tag=1820807938 >>> Contact: <sip:2000@192.168.25.24:5089> >>> Call-ID: 692c5d293d0fae0853872d6a3206af86@192.168.25.24:5089 >>> CSeq: 102 ACK >>> User-Agent: Asterisk PBX 13.10.0 >>> Content-Length: 0 >>> >>> >>> --- >>> [2017-03-16 11:34:53] WARNING[1554][C-000098af]: chan_sip.c:23843 >>> handle_response_invite: Received response: "Forbidden" from >>> '<sip:2000@192.168.25.24:5089>;tag=as62bede9e' >>> Scheduling destruction of SIP dialog >>> '692c5d293d0fae0853872d6a3206af86@192.168.25.24:5089' in 6400 ms >>> (Method: INVITE) >>> == Everyone is busy/congested at this time (1:0/0/1) >>> -- Auto fallthrough, channel 'SIP/2000-0000bd7a' status is >>> 'CHANUNAVAIL' >>> == MixMonitor close filestream (mixed) >>> == End MixMonitor Recording SIP/2000-0000bd7a >>> >>> See my sip.conf >>> >>> ;; >>> [136] >>> type=friend >>> defaultuser=136 >>> secret=XXXXX >>> qualify=yes >>> ;nat=no >>> nat=force_rport,comedia >>> context=ramais >>> ;insecure=invite,port >>> disallow=all >>> allow=ulaw,alaw,gsm >>> host=dynamic >>> canreinvite=no >>> regext=136 >>> callgroup=1 >>> pickupgroup=1 >>> _______________________________________________ >>> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1 >>> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7 >>> Intercomunicador e acesso remoto via rede IP e telefones IP >>> Conheça todo o portfólio em www.Khomp.com >>> _______________________________________________ >>> Para remover seu email desta lista, basta enviar um email em branco para >>> asteriskbrasil-unsubscr...@listas.asteriskbrasil.org >>> >> >> >> _______________________________________________ >> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1 >> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7 >> Intercomunicador e acesso remoto via rede IP e telefones IP >> Conheça todo o portfólio em www.Khomp.com >> _______________________________________________ >> Para remover seu email desta lista, basta enviar um email em branco para >> asteriskbrasil-unsubscr...@listas.asteriskbrasil.org >> > > > > -- > Atenciosamente > Alex de Freitas > Chamada via web 3CX https://afstisolution.ddns.net:5001/webrtc/116207 > _______________________________________________ KHOMP: completa linha de placas externas FXO, FXS, GSM e E1 Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7 Intercomunicador e acesso remoto via rede IP e telefones IP Conheça todo o portfólio em www.Khomp.com _______________________________________________ Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscr...@listas.asteriskbrasil.org