Hi all, i'm new at this forum. i have install my first elastix freePBX system. everything going well expet one thing (for now). i'm tring to connect SIP trunk to elastix system, as at seems the system was register but i'm not able to get calls or dial out. i think that the problem are codec or RTP port. So, i would like to know how can i change the RTP ports range and how can i choice diffrents codec for each SIP trunk. i don't have any knowledge with linux commands or asterix. Please help. Thank you.