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.
https://issues.asterisk.org/jira/browse/ASTERISK-19992
http://www.securityfocus.com/bid/54327
http://www.debian.org/security/2012/dsa-2550
http://secunia.com/advisories/50756
http://secunia.com/advisories/50687
http://downloads.asterisk.org/pub/security/AST-2012-010.html