[A*UG] Nat problem again - lost ability to hear callee

[Larry Alkoff]

Larry Alkoff wrote:
> I fixed my problem with NAT a long time ago but it has come back
> and I forgot what I did.
> 
> The person I called hears me ok but I cannot hear her.
> 
> I'm behind a firewall router with a 192.168 address.
> 
> Please look at my sip.conf settings or tell me what else to look at:
> 
> ;------------- NAT SUPPORT ----------------
> ; The externip, externhost and localnet settings are used if you use
> ; Asterisk behind a NAT device to communicate with services on the
> ; outside.
> 
> externip=larryalk.dyndns.org    ; Address that we're going to put in
> ;  outbound SIP messages if we're behind a NAT
> 
> ;externhost=foo.dyndns.net      ; Alternatively you can specify an
>                                 ; external host, and Asterisk will
>                                 ; perform DNS queries periodically.  Not
>                                 ; recommended for production
>                                 ; environments!  Use externip instead
> ;externrefresh=10               ; How often to refresh externhost if
>                                 ; used
>                                 ; You may add multiple local networks. 
>  ; A reasonable set of defaults are:
> localnet=192.168.0.0/255.255.255.0      ; lba
> 
> ;localnet=10.0.0.0/255.0.0.0    ; Also RFC1918
> ;localnet=172.16.0.0/12         ; Another RFC1918 with CIDR notation
> ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
> 
> ; The nat= setting is used when Asterisk is on a public IP,
> ; communicating with devices hidden behind a NAT device (broadband
> ; router).  If you have one-way
> ; audio problems, you usually have problems with your NAT configuration 
> ; or your firewall's support of SIP+RTP ports.
> ; You configure Asterisk choice of RTP ports for incoming audio in
> ; rtp.conf
> ;
> nat=yes                         ; lba set at A*UG meeting
> 
> -------------------------------------------------------------
> cat rtp.conf
> ;
> ; RTP Configuration
> [general]
> ;
> ; RTP start and RTP end configure start and end addresses
> ; Defaults are rtpstart=5000 and rtpend=31000
> ;
> rtpstart=10000
> rtpend=20000
> ;
> ; Whether to enable or disable UDP checksums on RTP traffic
> ;
> ;rtpchecksums=no
> ;
> ; The amount of time a DTMF digit with no 'end' marker should be
> ; allowed to continue (in 'samples', 1/8000 of a second)
> ;
> ;dtmftimeout=3000
> 
> -------------------------------------------------------------------
> 
> This once worked ok but it's been a while since I've worked on voip.
> 
In my router/firewall I forward UDP 5060-5082 and 10000-20000 to  the 
Asterisk machine.

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux