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*Sds,* *Luciano Cavalcante Souza* *Tecnólogo em Gestão da Tecnologia da Informação* *Mobile: + 55 79 98814.5895(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
