Hi there, I’m trying to get SpeeX 16 and SpeeX 32 to work with Asterisk 13.1.
Both codecs are listed and allowed in sip.conf and the codecs are choosen and used by the phones if connected. Its the same using Playback for soundfiles encoded in sln32 to the phone.
My only problem is that I cannot seem to change the quality settings of the codec in codecs.conf. The RTP payload size stays at 20 bytes per packet if using speex32, which sounds pretty crappy.
Is codecs.conf still the right place for configuration? I noticed that even if I deliberately use wrong keywords in the codecs.conf, asterisk does not complain if I reload the config using config reload codecs.conf from CLI.
Any pointers to solving this are appreciated.