Heh, should have guessed it would be you that replied Gareth ;)
Sorry yes, this box is on public IP with no NAT as is the upstream
providers box (or so they say).
So we have had audio cease outbound towards the provider. We have a
couple of volunteer customers who are being routed via this new
Hi All,
I have some questions regarding RTP and Asterisk;
I am trialling a new SIP upstream provider. We connect to them over
the Internet at present which I know is not ideal, but we are just
testing at present. During the trials we have had an issue where we
have had one way audio between us
On 27/11/13 14:12, James Bensley wrote:
What is the maximum delay RTP will tolerate one way (Does Asterisk
have a limit too)?
Can this be tuned (increased or decreased) within Asterisk (I'm
thinking of DSL customers where we may have this issue between our
PBXs and the customer)?
There isnt
