[Asterisk-Users] What route do diverted SIP calls travel?

[Andy Burns]

Sam Bashton wrote:

 > The data-heavy portion of the traffic is RTP, and that should be a
 > direct connection using your providers gateway.

Thanks, that was what I hoped for, no sense in all the traffic passing 
up and down my ADSL to get back to where it came from, I suppose the 
clue about SIP is in the name, if it only *initiates* the call the 
payload doesn't have to travel the same route as the call setup, nice ;-)

 > Make sure you have
 > 'canreinvite=yes' set in the appropriate section of your sip.conf.

I'll look into that, I'm just getting past the udev issues of asterisk 
on FC3 to get an X100P installed.