No sound on calls into Asterisk subnet, but OK on outgoing

I’ve purchased some Grandstream 286 single channel pstn converters and setup Asterisk on a Debian Sarge box to test our WAN for a VOIP solution for internal calls. Sound quality is good on the 256k ADSL links BUT I can only get sound to work in calls from the subnet where the Asterisk server is to WAN subnets. When I make calls in the reverse direction the phones ring but no sound.

Routed private network - no NAT.

Sip.conf
[general]
context=atp-unauthenticated ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls

videosupport=yes ; Turn on support for SIP video

dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
disallow = all
allow = gsm
allow = alaw ; Other options:

sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
subscribecontext=atp-hint ; Set a specific context for SUBSCRIBE requests

nat=never ; Global NAT settings (Affects all peers and users)

[2000]
callgroup=
pickupgroup=
type=friend
username=2000
secret=secret2000
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=2000
nat=no
qualify=3000
call-limit=2
context=atp-extensions
accountcode=main
callerid=""<2000>
disallow=all
allow=alaw
allow=ulaw

[2030]
callgroup=
pickupgroup=
type=friend
username=2030
secret=secret2030
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=2030
nat=no
qualify=3000
call-limit=1
context=atp-extensions
accountcode=main
callerid=""<2030>
disallow=all
allow=alaw
allow=ulaw
allow=ilbc

[2049]
type=friend
username=2049
secret=secret2049
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=2049
nat=no
qualify=3000
call-limit=1
context=atp-extensions
accountcode=main
callerid=<2049>
disallow=all
allow=alaw
allow=ilbc
allow=ulaw

After resetting devices and causing the problem the logs on the 286 show:

External box Zero RTP packets sent or received
Internal box 712 sent Zero received

Asterisk RTP debugging shows GOT packets from internal extension IP & SENT to external external extension IP.

There is NO NAT in use and there are NO ports being blocked between the two subnets.

I’m stymied as to why it works fine one way but not the other. Any advice would be greatly appreciated.

Thanks.

PS When making the inwards calls, the ring tone isn’t heard on the headset of the calling phone and the call is disconnected after about 15 seconds although it has been answered.

possibly a firewall issue? If some but not all of your RTP range is blocked you can get some very strange issues like that (calls work one way but not the other)…