On test we have found that the maximum extension length for the main extension is 7 digits, This will then create a 9 digit extension for the webrtc phone, The sip user is created and importantly the certman_mapping table in the asterisk(freepbx) database is created as
| 997654321 | 1 | fingerprint | actpass | 0 |
this then means when the sip extension is created the lines
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/default.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
dtlsrekey=0
are added to the sip_additional for that extension.
for an 8 digit extension the certman_mapping isnt updated and the lines above arnt added to the extension in sip_additional.conf eventhough the 10 digit extension starting 99 is. this means when you try to call from the UCP the call will fail with “Rejecting secure audio stream without encryption details: audio 2363 RTP/SAVPF”
Also the logic on removing a UCP user is a little broken as you have to first disable the webrtc option then remove the user. if you jsut remove the user the sip setting remain in sip_additional.conf after reload, as well as this even if you do disable the webrtc phone and reload the sip settings are removed correctly but the certman_mapping is left in the database.
Seems my login to Jira is broken so cant raise bug yet,