Description
channels/chan_sip.c in Asterisk Open Source 1.8.x before 1.8.13.1 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.
Metrics
AV:N/AC:L/Au:S/C:N/I:N/A:PWeakness enumeration
Affected software
No affected versions have been published.
Timeline
- Published
- Jul 9, 2012
- Last modified
- Jun 16, 2026
- NVD status
- Modified