Does Asterisk allow direct voice between SIP endpoints?

Does Asterisk let endpoints converse directly? Is there any way to tell using show sip channels, etc to determine if two terminals are talking directly or via Asterisk?

I need to know this as I want to be sure when two terminals sitting on single LAN at a site remote to the Asterisk server they use Asterisk to setup the call (obviously), but the RTP traffic between the endpoints stays on their local LAN.

Any help on how to establish this and/or how to control it would be appreciated.


tcpdump is your friend (ethereal might be easier)

Got it thanks.

Yes I knew it was passing data through Asterisk, just couldn’t work out how to get it to hand over. The trick is to remove the t option in the sip_additional.conf file (when using AMP) so the dial command is not sent the t option.

Works a treat now.