Ken is creating a bump-in-the-wire box for legacy IP phones and might
find this script useful.

mod_oreka: use zstr instead of only checking for null as some caller profile data may be an empty string

FS-5682 --resolve there was one other change to allow them to copy over by setting them on the A leg.  Exporting the sip_multipart variable or setting it in {} on the dial string should already work.  This patch should make it work with setting it too.

FS-5687 this patch adds a debouncer to that info message.  if you still have problems try altering switch_rtp.c:5469 to add 0 condition so it skips that whole block and retest eg... if (0 && switch_rtp_test_flag(rtp_session, SWITCH_RTP_FLAG_VIDEO)) {

FS-5686 --resolve mod_rayo: allow re-join to mixers so that audio direction can be changed.  Added support for audio direction in mixers.  Allow unjoin without mixer-name or call-id

FS-5682 --resolve new syntax <action application="set" data="sip_multipart=application/pidf+xml:1234"/> use push instead of set to add more to the same channel

Chrome send "Connection: Upgrade" while FF has "Connection: keep-alive, Upgrade"

like the quirks we used to get using those with C++ have gone away.