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Search Parameters:
  • CPE Product Version: cpe:/a:digium:certified_asterisk:11.6:cert1_rc2
There are 13 matching records.
Displaying matches 1 through 13.
Vuln ID Summary CVSS Severity
CVE-2019-13161

An issue was discovered in Asterisk Open Source through 13.27.0, 14.x and 15.x through 15.7.2, and 16.x through 16.4.0, and Certified Asterisk through 13.21-cert3. A pointer dereference in chan_sip while handling SDP negotiation allows an attacker to crash Asterisk when handling an SDP answer to an outgoing T.38 re-invite. To exploit this vulnerability an attacker must cause the chan_sip module to send a T.38 re-invite request to them. Upon receipt, the attacker must send an SDP answer containing both a T.38 UDPTL stream and another media stream containing only a codec (which is not permitted according to the chan_sip configuration).

Published: July 12, 2019; 4:15:11 PM -0400
V3.1: 5.3 MEDIUM
V2.0: 3.5 LOW
CVE-2017-14603

In Asterisk 11.x before 11.25.3, 13.x before 13.17.2, and 14.x before 14.6.2 and Certified Asterisk 11.x before 11.6-cert18 and 13.x before 13.13-cert6, insufficient RTCP packet validation could allow reading stale buffer contents and when combined with the "nat" and "symmetric_rtp" options allow redirecting where Asterisk sends the next RTCP report.

Published: October 09, 2017; 9:30:21 PM -0400
V3.0: 7.5 HIGH
V2.0: 5.0 MEDIUM
CVE-2017-14100

In Asterisk 11.x before 11.25.2, 13.x before 13.17.1, and 14.x before 14.6.1 and Certified Asterisk 11.x before 11.6-cert17 and 13.x before 13.13-cert5, unauthorized command execution is possible. The app_minivm module has an "externnotify" program configuration option that is executed by the MinivmNotify dialplan application. The application uses the caller-id name and number as part of a built string passed to the OS shell for interpretation and execution. Since the caller-id name and number can come from an untrusted source, a crafted caller-id name or number allows an arbitrary shell command injection.

Published: September 02, 2017; 12:29:00 PM -0400
V3.0: 9.8 CRITICAL
V2.0: 7.5 HIGH
CVE-2017-14099

In res/res_rtp_asterisk.c in Asterisk 11.x before 11.25.2, 13.x before 13.17.1, and 14.x before 14.6.1 and Certified Asterisk 11.x before 11.6-cert17 and 13.x before 13.13-cert5, unauthorized data disclosure (media takeover in the RTP stack) is possible with careful timing by an attacker. The "strictrtp" option in rtp.conf enables a feature of the RTP stack that learns the source address of media for a session and drops any packets that do not originate from the expected address. This option is enabled by default in Asterisk 11 and above. The "nat" and "rtp_symmetric" options (for chan_sip and chan_pjsip, respectively) enable symmetric RTP support in the RTP stack. This uses the source address of incoming media as the target address of any sent media. This option is not enabled by default, but is commonly enabled to handle devices behind NAT. A change was made to the strict RTP support in the RTP stack to better tolerate late media when a reinvite occurs. When combined with the symmetric RTP support, this introduced an avenue where media could be hijacked. Instead of only learning a new address when expected, the new code allowed a new source address to be learned at all times. If a flood of RTP traffic was received, the strict RTP support would allow the new address to provide media, and (with symmetric RTP enabled) outgoing traffic would be sent to this new address, allowing the media to be hijacked. Provided the attacker continued to send traffic, they would continue to receive traffic as well.

Published: September 02, 2017; 12:29:00 PM -0400
V3.0: 7.5 HIGH
V2.0: 5.0 MEDIUM
CVE-2016-7551

chain_sip in Asterisk Open Source 11.x before 11.23.1 and 13.x 13.11.1 and Certified Asterisk 11.6 before 11.6-cert15 and 13.8 before 13.8-cert3 allows remote attackers to cause a denial of service (port exhaustion).

Published: April 17, 2017; 12:59:00 PM -0400
V3.0: 7.5 HIGH
V2.0: 5.0 MEDIUM
CVE-2016-9938

An issue was discovered in Asterisk Open Source 11.x before 11.25.1, 13.x before 13.13.1, and 14.x before 14.2.1 and Certified Asterisk 11.x before 11.6-cert16 and 13.x before 13.8-cert4. 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.

Published: December 12, 2016; 4:59:01 PM -0500
V3.0: 5.3 MEDIUM
V2.0: 5.0 MEDIUM
CVE-2016-2316

chan_sip in Asterisk Open Source 1.8.x, 11.x before 11.21.1, 12.x, and 13.x before 13.7.1 and Certified Asterisk 1.8.28, 11.6 before 11.6-cert12, and 13.1 before 13.1-cert3, when the timert1 sip.conf configuration is set to a value greater than 1245, allows remote attackers to cause a denial of service (file descriptor consumption) via vectors related to large retransmit timeout values.

Published: February 22, 2016; 10:59:02 AM -0500
V3.0: 5.9 MEDIUM
V2.0: 7.1 HIGH
CVE-2016-2232

Asterisk Open Source 1.8.x, 11.x before 11.21.1, 12.x, and 13.x before 13.7.1 and Certified Asterisk 1.8.28, 11.6 before 11.6-cert12, and 13.1 before 13.1-cert3 allow remote authenticated users to cause a denial of service (uninitialized pointer dereference and crash) via a zero length error correcting redundancy packet for a UDPTL FAX packet that is lost.

Published: February 22, 2016; 10:59:01 AM -0500
V3.0: 6.5 MEDIUM
V2.0: 4.0 MEDIUM
CVE-2015-3008

Asterisk Open Source 1.8 before 1.8.32.3, 11.x before 11.17.1, 12.x before 12.8.2, and 13.x before 13.3.2 and Certified Asterisk 1.8.28 before 1.8.28-cert5, 11.6 before 11.6-cert11, and 13.1 before 13.1-cert2, when registering a SIP TLS device, does not properly handle a null byte in a domain name in the subject's Common Name (CN) field of an X.509 certificate, which allows man-in-the-middle attackers to spoof arbitrary SSL servers via a crafted certificate issued by a legitimate Certification Authority.

Published: April 10, 2015; 11:00:10 AM -0400
V3.x:(not available)
V2.0: 4.3 MEDIUM
CVE-2014-4047

Asterisk Open Source 1.8.x before 1.8.28.1, 11.x before 11.10.1, and 12.x before 12.3.1 and Certified Asterisk 1.8.15 before 1.8.15-cert6 and 11.6 before 11.6-cert3 allows remote attackers to cause a denial of service (connection consumption) via a large number of (1) inactive or (2) incomplete HTTP connections.

Published: June 17, 2014; 10:55:07 AM -0400
V3.x:(not available)
V2.0: 5.0 MEDIUM
CVE-2014-4046

Asterisk Open Source 11.x before 11.10.1 and 12.x before 12.3.1 and Certified Asterisk 11.6 before 11.6-cert3 allows remote authenticated Manager users to execute arbitrary shell commands via a MixMonitor action.

Published: June 17, 2014; 10:55:07 AM -0400
V3.x:(not available)
V2.0: 6.5 MEDIUM
CVE-2014-2287

channels/chan_sip.c in Asterisk Open Source 1.8.x before 1.8.26.1, 11.8.x before 11.8.1, and 12.1.x before 12.1.1, and Certified Asterisk 1.8.15 before 1.8.15-cert5 and 11.6 before 11.6-cert2, when chan_sip has a certain configuration, allows remote authenticated users to cause a denial of service (channel and file descriptor consumption) via an INVITE request with a (1) Session-Expires or (2) Min-SE header with a malformed or invalid value.

Published: April 18, 2014; 6:14:38 PM -0400
V3.x:(not available)
V2.0: 3.5 LOW
CVE-2014-2286

main/http.c in Asterisk Open Source 1.8.x before 1.8.26.1, 11.8.x before 11.8.1, and 12.1.x before 12.1.1, and Certified Asterisk 1.8.x before 1.8.15-cert5 and 11.6 before 11.6-cert2, allows remote attackers to cause a denial of service (stack consumption) and possibly execute arbitrary code via an HTTP request with a large number of Cookie headers.

Published: April 18, 2014; 6:14:37 PM -0400
V3.x:(not available)
V2.0: 7.5 HIGH