Note: Running Elastix 2.0.3., and I have licensed g729 codec from Digium. Using Polycom 601 phones that support g729 natively. My goal is the following: ulaw for extension->extension (works) ulaw for extension->voicemail (works) ulaw for extension -> PSTN FXO (works) ulaw for PSTN FXO -> extensions/voicemail/IVR (works) g729 for extension->SIP Trunk (transcodes ) g729 for SIP Trunk->extensions/voicemail/IVR (works) So far everything is working, except outgoing calls from the extensions--they are showing being transcoded rather than going g729 end to end (pass-through). Right now in sip show channels they show up as ulaw at the extension and g729 on the trunk so they're being transcoded--which wastes my g729 licenses (and I can hear a difference...even in a blind test between g729->ulaw and g729->g729). How can I get outgoing calls on my SIP trunk to run native g729 from end to end while keeping ulaw for internal calls and calls oven the PSTN?