I’m having trouble with asterisk trying to use L16/44100 codec on my SIP-client. If I put allow=slin44 in pjsip.conf, call does not connect because there are no common codecs. If I put allow=slin, the call connects, but asterisk treats the audio as 8 kHz, which is wrong.
I was wondering if this is the reason why allow=slin44 does not work: https://github.com/asterisk/asterisk/blob/master/main/rtp_engine.c
See line 3650:
set_next_mime_type(ast_format_slin44, 0, “audio”, “L16”, 44000); // Shouldn’t this be 44100, not 44000
I actually tried to to edit the source file and changed the 44000 to 44100 and recompile and after that the calls were connected ok with allow=slin44.