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)
|
Configuration 2 (hide)
|
Configuration 3 (hide)
|
Configuration 4 (hide)
|
Configuration 5 (hide)
|
History
21 Nov 2024, 01:41
Type | Values Removed | Values Added |
---|---|---|
References | () http://downloads.asterisk.org/pub/security/AST-2012-010.html - Patch, Vendor Advisory | |
References | () http://secunia.com/advisories/50687 - | |
References | () http://secunia.com/advisories/50756 - | |
References | () http://www.debian.org/security/2012/dsa-2550 - | |
References | () http://www.securityfocus.com/bid/54327 - | |
References | () https://issues.asterisk.org/jira/browse/ASTERISK-19992 - |
Information
Published : 2012-07-09 10:20
Updated : 2024-11-21 01:41
NVD link : CVE-2012-3863
Mitre link : CVE-2012-3863
CVE.ORG link : CVE-2012-3863
JSON object : View
Products Affected
digium
- certified_asterisk
- asteriske
- asterisk_business_edition
- asterisk
CWE
CWE-399
Resource Management Errors