DirectMedia and the Digium G100/G200 gateways

Hello,

In a configuration with an asterisk server (serving as the IPBX) and a digium G100/G200, is it possible to activate the directmedia mode in the asterisk server so that any sip phone calling a line “behind the E1/T1 trunk of the gateway” would exchange RTP packets directly with the gateway?

Thanks for the help,

No, if a call comes into the gateway, the gateway will handle the media. It doesn’t operate in a mode where it doesn’t terminate media.

using directmedia=yes should allow the packets to go directly from the SIP phone to the gateway and leave asterisk out of the loop… I do this with ADTRAN Total access gateways all the time… a SIP user makes a call to a PSTN destination which the trunk is a PRI / T1 connected to an adtran gateway… once the call is established the voice path goes from the SIP phone directly to the Adtran gateway…

because the gateway is basically an E2T (ethernet to TDM) there is NO WAY a call can go directly from the far end phone (even if it is SIP on a SIP trunk) directly to the local SIP set…

for True endpoint to endpoint communications both the caller and the Callee would need to SIP sets on SIP trunks and NOT PASS through any SBC, or other SIP servers transcoding… (unlikely to happen)…

-Christopher