[asterisk-users] Asterisk 11.25.1, 13.13.1, 14.2.1, 11.6-cert16, and 13.8-cert4 Now Available (Security Release)
Asterisk Development Team
asteriskteam at digium.com
Thu Dec 8 16:19:49 CST 2016
The Asterisk Development Team has announced security releases for Asterisk
11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available
security releases are released as versions 11.25.1, 13.13.1, 14.2.1,
11.6-cert16, and 13.8-cert4.
These releases are available for immediate download at:
The release of versions 13.13.1 and 14.2.1 resolve the following security
* AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
If an SDP offer or answer is received with the Opus codec and with the
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.
also does not require the endpoint to be authenticated. If guest is
for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
processed and the crash occurs.
The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
resolve the following security vulnerability:
* AST-2016-009: Remote unauthenticated sessions in chan_sip
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
were whitespace. This means that headers such as
will be seen as a valid Contact header.
This mostly does not pose a problem until Asterisk is placed in tandem
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
request. However, because of the bug described above, the request will
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.
If you do not use a proxy for authentication, then this issue does not
you. If your proxy is dialog-aware (meaning that the proxy keeps track of
dialogs are currently valid), then this issue does not affect you. If you
chan_pjsip instead of chan_sip, then this issue does not affect you.
For a full list of changes in the current releases, please see the
The security advisories are available at:
Thank you for your continued support of Asterisk!
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