The CVE-2017-14099 advisory stated that Asterisk Open Source version 11.x series is affected by the RTP bleed.
1) I am having FreePBX 188.8.131.52 with these modules Asterisk CLI 184.108.40.206, Asterisk Info 12.0.2, Asterisk Logfiles 12.0.6, Asterisk API 12.0.2, Asterisk IAX Settings 220.127.116.11, Asterisk SIP Settings 12.0.16.
2) The Asterisk Info page is showing "Asterisk (Ver. 11.15.0)"
3) The Asterisk SIP setting is showing that my "Strict RTP" setting is turned on.
Am I affected by this RTP bleed bug?
If yes, is there any bug fix for this?