The CVE-2017-14099 advisory stated that Asterisk Open Source version 11.x series is affected by the RTP bleed.
- I am having FreePBX 126.96.36.199 with these modules Asterisk CLI 188.8.131.52, Asterisk Info 12.0.2, Asterisk Logfiles 12.0.6, Asterisk API 12.0.2, Asterisk IAX Settings 184.108.40.206, Asterisk SIP Settings 12.0.16.
- The Asterisk Info page is showing “Asterisk (Ver. 11.15.0)”
- 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?