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CVE-2012-3863

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, Ce

Published
Last modified
published
Not listed in CISA KEV
None on Exploit-DB, Metasploit or GitHub

Summary

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.

Timeline

  1. CVE published
  2. Record last updated

Affected versions

  • n/a n/a

    Version range not reported

References 6

secunia.com

debian.org

securityfocus.com

downloads.asterisk.org

issues.asterisk.org