Name | CVE-2012-3863 |
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. |
Source | CVE (at NVD; CERT, LWN, oss-sec, fulldisc, Red Hat, Ubuntu, Gentoo, SUSE bugzilla/CVE, GitHub advisories/code/issues, web search, more) |
References | DSA-2550-1 |
The table below lists information on source packages.
The information below is based on the following data on fixed versions.