[AsteriskBrasil] Received response: "Forbidden" em um Grandstream HT-503

[Luciano Cavalcante Souza]

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*Sds,*

*Luciano Cavalcante Souza*
*Tecnólogo em Gestão da Tecnologia da Informação*
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2017-03-16 12:00 GMT-03:00 Vitor Mazuco <vitor.mazuco at 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 at ramais:1] MixMonitor("SIP/2000-0000bd8b",
> "/media/HDExterno/gravacoes/feitas/APTO/27100/1489675579.120546.wav")
> in new stack
>     -- Executing [27100 at 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 at 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 at 192.168.25.24:5089>;tag=as62bede9e
> To: <sip:100 at 192.168.25.169>
> Call-ID: 692c5d293d0fae0853872d6a3206af86 at 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 at 192.168.25.24:5089>;tag=as62bede9e
> To: <sip:100 at 192.168.25.169>;tag=1820807938
> Call-ID: 692c5d293d0fae0853872d6a3206af86 at 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 at 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 at 192.168.25.24:5089>;tag=as62bede9e
> To: <sip:100 at 192.168.25.169>;tag=1820807938
> Contact: <sip:2000 at 192.168.25.24:5089>
> Call-ID: 692c5d293d0fae0853872d6a3206af86 at 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 at 192.168.25.24:5089>;tag=as62bede9e'
> Scheduling destruction of SIP dialog
> '692c5d293d0fae0853872d6a3206af86 at 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
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