One Way Audio - NAT Involved

Asterisk Asterisk Support
1

Okay. Here is the set-up:

Asterisk is in a DMZ behind a NAT. Clients are in DMZ’s behind other NAT’s, but are using STUN servers to find their public IP.

Clients can call to Asterisk and hear everything fine. Audio from their phones does not work, and Asterisk does not show any RTP packets received from the client.

Here is my sip.conf file:

[general]

port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
nat=yes
externip=69.252.188.101
localnet=192.168.2.0/255.255.255.0

[4747]
username=4747
type=friend
secret=*********
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
mailbox=4747@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=device <4747>

Anyone have any ideas? I know this has to be working for someone.

Soltys - If you read this, does this problem relate to the issues that were discussed in the previous topic :question:

2

NAT does not allways work. I was trying to get my server (which was behind NAT) to work and it didnt. I had to set the router to pass the packets rather than use NAT.

3

What is weird is that I do have the routed set up to forward the packets, and I have the computer set up on a DMZ. From what I can tell, I haven’t had any problems getting packets in to me.

It looks like the client (far end) does not send any RTP packets at all, based on Ethereal traces at the far end.