Hi list, i'm a brand new freeswitch user (without previous asterisk/voip experience), after reading all wiki pages, google searchs, etc i need some help to solve a problem. configuration: Freeswitch -> Firewall (nat) -> internet -> Sip Provider In my current configuration the gateway is REGED and inbound calls (from provider to freeswitch) works ok with good audio quality. However outbound calls don't. When i call through the gateway the destination phone rings, and when is answered there is no audio. I've check with "show channels" in fs_cli and i cant see any codec in the read_codec write_codec part, they are blank. I've reviewed all sip profiles configuration, but obviously i'm missing something. I will really appreciate any comment,guidance,help,etc. (if someone is in Buenos Aires/Argentina i can also offer a free beer!) Thanks in advance. Eduardo.