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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.
| Product | Vendor Advisory | Solution File | Added | Published |
|---|---|---|---|---|
| Alpine Linux | — | Upgrade asterisk | Aug 30, 2017 | Jul 9, 2012 |
| Debian | — | Upgrade asterisk | Jul 30, 2024 | Jul 9, 2012 |
| Gentoo Linux | — | Upgrade net-misc/asterisk. | Oct 30, 2017 | Jul 9, 2012 |
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