Before your latest reply, I was searching online trying to wrap my brain around this problem. I found many references to codec mismatches in FreePBX, but I still do not understand the source of the issue, especially since it’s happening so intermittently, only on transfers, and spontaneously began with nothing having been changed on the phones or system to my knowledge.
A call comes in and the operator picks it up - both parties can hear each other, so the PBX must be doing whatever transcoding is necessary (if any) behind the scenes. When the operator transfers a call to a ring group, the operator phone is basically calling that ring group and then connecting the already-established call to it. The operators have no problem calling the ring group as a new call - it’s only when connecting the established call to the ring group that there’s an infrequent issue. If it were a codec problem, wouldn’t it be an every-call problem, basically disabling the ring group, instead of only happening a few times a month? My knowledge of codecs comes mainly from the AV world, using ffmpeg to transcode audio and video into whatever formats the device I’m putting them on wants. (Exceptionally silly of the TV manufacturers to not standardize to support a given set of codecs, but whatever… I now have a library of scripts to transcode the movies I own to a format that will play on any TV in my home. 
) What am I missing in my understanding???
One post I found on reddit says to look for a hyphen outside of the diagonal line in the translation matrix of asterisk to find the missing codec. I did this & there do not appear to be any missing:
Enabled sip debugging in asterisk and almost immediately turned it off again - the log file would’ve filled the file system in less than 4 hours. I opened a new thread for this sub-topic as it’s about a separate, though related, issue. Was thinking that might help others if indexed with a more accurate title.
I’ve not tried migrating to pjsip yet, but plan to try doing so on a few phones. Remember fighting with pjsip for several days when first setting up our phones and was never able to get it to work. Tried chan_sip and things started working. Big difference between then & now is the endpoints - Polycom instead of Cisco phones. Unfortunately, I do not remember enough details to be of use should problems arise now. 