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
Affected Vendors & Products
Advisories
Source | ID | Title |
---|---|---|
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DSA-2550-1 | asterisk security update |
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DSA-2550-2 | asterisk regression update |
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EUVD-2012-3810 | 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. |
Fixes
Solution
No solution given by the vendor.
Workaround
No workaround given by the vendor.
References
History
No history.

Status: PUBLISHED
Assigner: mitre
Published:
Updated: 2024-08-06T20:21:03.613Z
Reserved: 2012-07-06T00:00:00
Link: CVE-2012-3863

No data.

Status : Deferred
Published: 2012-07-09T10:20:44.823
Modified: 2025-04-11T00:51:21.963
Link: CVE-2012-3863

No data.

No data.