I set up asterisk 16 and jigasi using a pjsip channel following the example in Working configuration for Jitsi/Jigasi with Asterisk/SIP. I can successfully call into a room and can call out from the room, but there is no audio. I see no messages in the log file that seems to help. I suspected a codec mismatch. My SIP provider supports G711 (alaw) G723 and G739. So configuring my endpoint for alaw successfully allowed me to go through the whole dialplan including the prompts.
Jigasi apparently doesn’t support G711a, G723 and G739. Can I setup the dialplan to do transcoding from alaw to G722 or speex or gsm for example for pjsip snd how basically can I do it? I think chan-sip had channel variables like SIP_CODEC and SIP_CODEC_OUTBOUND to do this, but I prefer to use chan_pjsip.
I noticed that Asterisk 18 has Advanced Codec Negotiation, https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation. Must I use 18?
Any suggestions appreciated.