Verifica o seu SIP.CONF para saber quais portas RTP e porta SIP ele está usando e cria um redirecionamento fixo dessas portas para o IP do seu grandstream! Em 16 de mar de 2017 3:30 PM, "Luciano Cavalcante Souza" <lucin...@gmail.com> escreveu:
> 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 >
_______________________________________________ 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