channels/chan_sip.c in Asterisk Open Source 1.8.x before 220.127.116.11 and
10.x before 10.5.2, Asterisk Business Edition C.3.x before C.3.7.5,
Certified Asterisk 1.8.11-certx before 1.8.11-cert4, and Asterisk
Digiumphones 10.x.x-digiumphones before 10.5.2-digiumphones does not
properly handle a provisional response to a SIP reINVITE request,
which allows remote authenticated users to cause a denial of service
(RTP port exhaustion) via sessions that lack final responses.
Note:References are provided for the convenience of the reader to help distinguish between vulnerabilities. The list is not intended to be complete.
Disclaimer: The entry creation date may reflect when
the CVE-ID was allocated or reserved, and does not
necessarily indicate when this vulnerability was
discovered, shared with the affected vendor, publicly
disclosed, or updated in CVE.
This is an entry on the CVE
list, which standardizes names for security