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.
References
Configurations
Configuration 1 (hide)
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Configuration 2 (hide)
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Configuration 3 (hide)
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Configuration 4 (hide)
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Configuration 5 (hide)
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Information
Published : 2012-07-09 03:20
Updated : 2013-10-10 11:24
NVD link : CVE-2012-3863
Mitre link : CVE-2012-3863
JSON object : View
CWE
CWE-399
Resource Management Errors
Products Affected
digium
- certified_asterisk
- asterisk_business_edition
- asteriske
- asterisk