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

[Larry Alkoff]

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.

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