I have a really weird problem. when I call from my personal phone to sip number the sound works well. but when I call from internal phone (extension) lets say from 100 to 200 the sound not working until I press and release the hold. when I look in the asterisk log it looks that the rtp not start before the hold although the phones bridge correctly. I google the problem and see a lot of info about the problem but none of those fixed the problem.
the point is that the rtp not even start when someone answering but when I press hold or transfer the call to another ext then the rtp start and everything works correct
btw all the phones connect to asterisk from another network (from my home network to asterisk server)