Download Asterisk 14.2.1 / 13.13.1 / 11.25.1 / 13.8-cert4 / 11.6-cert16

Asterisk is an extensive pbx for BSD, Linux and Mac OS X. The program offers all the functions you would expect from a telephone exchange. For example, it has options for voicemail, conferencing and call queuing. In addition, support for caller id services, adsi, sip and h323 is present. For a complete overview of all possibilities, we refer to this page. The developers have released versions 14.2.1, 13.13.1, 11.25.1, 13.8-cert4, and 11.6-cert16 with the following announcement:

Asterisk 11.25.1, 13.13.1, 14.2.1, 11.6-cert16, and 13.8-cert4 Now Available (Security Release)

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:
http://downloads.asterisk.org/pub/telephony/asterisk/releases
http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/

The release of versions 13.13.1 and 14.2.1 resolve the following security vulnerability:

• 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 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.

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 it were whitespace. This means that headers such as
  Contact\x01:
  will be seen as a valid Contact header. 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.
  If you do not use a proxy for authentication, then this issue does not affect you. 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. If you use 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 ChangeLogs:
ChangeLog-11.25.1
ChangeLog-13.13.1
ChangeLog-14.2.1
ChangeLog-certified-11.6-cert16
ChangeLog-certified-13.8-cert4

The security advisories are available at:
AST-2016-008.pdf
AST-2016-009.pdf

Thank you for your continued support of Asterisk!