svn commit: r428237 - head/security/vuxml
Guido Falsi
madpilot at FreeBSD.org
Fri Dec 9 19:44:13 UTC 2016
Author: madpilot
Date: Fri Dec 9 19:44:11 2016
New Revision: 428237
URL: https://svnweb.freebsd.org/changeset/ports/428237
Log:
Document vulnerabilities in net/asterisk11 and net/asterisk13.
Modified:
head/security/vuxml/vuln.xml
Modified: head/security/vuxml/vuln.xml
==============================================================================
--- head/security/vuxml/vuln.xml Fri Dec 9 19:30:19 2016 (r428236)
+++ head/security/vuxml/vuln.xml Fri Dec 9 19:44:11 2016 (r428237)
@@ -58,6 +58,92 @@ Notes:
* Do not forget port variants (linux-f10-libxml2, libxml2, etc.)
-->
<vuxml xmlns="http://www.vuxml.org/apps/vuxml-1">
+ <vuln vid="c0b13887-be44-11e6-b04f-001999f8d30b">
+ <topic>asterisk -- Authentication Bypass</topic>
+ <affects>
+ <package>
+ <name>asterisk11</name>
+ <range><lt>11.25.1</lt></range>
+ </package>
+ <package>
+ <name>asterisk13</name>
+ <range><lt>13.13.1</lt></range>
+ </package>
+ </affects>
+ <description>
+ <body xmlns="http://www.w3.org/1999/xhtml">
+ <p>The Asterisk project reports:</p>
+ <blockquote cite="http://www.asterisk.org/downloads/security-advisories">
+ <p>The chan_sip channel driver has a liberal definition
+ for whitespace when attempting to strip the content between
+ a SIP header name and a colon character. Rather than
+ following RFC 3261 and stripping only spaces and horizontal
+ tabs, Asterisk treats any non-printable ASCII character
+ as if it were whitespace.</p>
+ <p>This mostly does not pose a problem until Asterisk is
+ placed in tandem with an authenticating SIP proxy. In
+ such a case, a crafty combination of valid and invalid
+ To headers can cause a proxy to allow an INVITE request
+ into Asterisk without authentication since it believes
+ the request is an in-dialog request. However, because of
+ the bug described above, the request will look like an
+ out-of-dialog request to Asterisk. Asterisk will then
+ process the request as a new call. The result is that
+ Asterisk can process calls from unvetted sources without
+ any authentication.</p>
+ <p>If you do not use a proxy for authentication, then
+ this issue does not affect you.</p>
+ <p>If your proxy is dialog-aware (meaning that the proxy
+ keeps track of what dialogs are currently valid), then
+ this issue does not affect you.</p>
+ <p>If you use chan_pjsip instead of chan_sip, then this
+ issue does not affect you.</p>
+ </blockquote>
+ </body>
+ </description>
+ <references>
+ <url>http://downloads.digium.com/pub/security/ASTERISK-2016-009.html</url>
+ </references>
+ <dates>
+ <discovery>2016-11-28</discovery>
+ <entry>2016-12-09</entry>
+ </dates>
+ </vuln>
+
+ <vuln vid="9e6640fe-be3a-11e6-b04f-001999f8d30b">
+ <topic>asterisk -- Crash on SDP offer or answer from endpoint using Opus</topic>
+ <affects>
+ <package>
+ <name>asterisk13</name>
+ <range><ge>13.12.0</ge><lt>13.13.1</lt></range>
+ </package>
+ </affects>
+ <description>
+ <body xmlns="http://www.w3.org/1999/xhtml">
+ <p>The Asterisk project reports:</p>
+ <blockquote cite="http://www.asterisk.org/downloads/security-advisories">
+ <p>If an SDP offer or answer is received with the Opus
+ codec and with the format parameters separated using a
+ space the code responsible for parsing will recursively
+ call itself until it crashes. This occurs as the code
+ does not properly handle spaces separating the parameters.
+ This does NOT require the endpoint to have Opus configured
+ in Asterisk. This also does not require the endpoint to
+ be authenticated. If guest is enabled for chan_sip or
+ anonymous in chan_pjsip an SDP offer or answer is still
+ processed and the crash occurs.</p>
+ </blockquote>
+ </body>
+ </description>
+ <references>
+ <url>http://downloads.asterisk.org/pub/security/AST-2016-008.html</url>
+ </references>
+ <dates>
+ <discovery>2016-11-11</discovery>
+ <entry>2016-12-09</entry>
+ </dates>
+ </vuln>
+
<vuln vid="eab68cff-bc0c-11e6-b2ca-001b3856973b">
<topic>cryptopp -- multiple vulnerabilities</topic>
<affects>
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