Total
51 CVE
| CVE | Vendors | Products | Updated | CVSS v2 | CVSS v3 |
|---|---|---|---|---|---|
| CVE-2018-12227 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2019-03-29 | 5.0 MEDIUM | 5.3 MEDIUM |
| An issue was discovered in Asterisk Open Source 13.x before 13.21.1, 14.x before 14.7.7, and 15.x before 15.4.1 and Certified Asterisk 13.18-cert before 13.18-cert4 and 13.21-cert before 13.21-cert2. When endpoint specific ACL rules block a SIP request, they respond with a 403 forbidden. However, if an endpoint is not identified, then a 401 unauthorized response is sent. This vulnerability just discloses which requests hit a defined endpoint. The ACL rules cannot be bypassed to gain access to the disclosed endpoints. | |||||
| CVE-2018-7284 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2019-03-01 | 5.0 MEDIUM | 7.5 HIGH |
| A Buffer Overflow issue was discovered in Asterisk through 13.19.1, 14.x through 14.7.5, and 15.x through 15.2.1, and Certified Asterisk through 13.18-cert2. When processing a SUBSCRIBE request, the res_pjsip_pubsub module stores the accepted formats present in the Accept headers of the request. This code did not limit the number of headers it processed, despite having a fixed limit of 32. If more than 32 Accept headers were present, the code would write outside of its memory and cause a crash. | |||||
| CVE-2017-17850 | 1 Digium | 2 Asterisk, Certified Asterisk | 2018-11-25 | 5.0 MEDIUM | 7.5 HIGH |
| An issue was discovered in Asterisk 13.18.4 and older, 14.7.4 and older, 15.1.4 and older, and 13.18-cert1 and older. A select set of SIP messages create a dialog in Asterisk. Those SIP messages must contain a contact header. For those messages, if the header was not present and the PJSIP channel driver was used, Asterisk would crash. The severity of this vulnerability is somewhat mitigated if authentication is enabled. If authentication is enabled, a user would have to first be authorized before reaching the crash point. | |||||
| CVE-2017-16671 | 1 Digium | 2 Asterisk, Certified Asterisk | 2018-11-25 | 6.5 MEDIUM | 8.8 HIGH |
| A Buffer Overflow issue was discovered in Asterisk Open Source 13 before 13.18.1, 14 before 14.7.1, and 15 before 15.1.1 and Certified Asterisk 13.13 before 13.13-cert7. No size checking is done when setting the user field for Party B on a CDR. Thus, it is possible for someone to use an arbitrarily large string and write past the end of the user field storage buffer. NOTE: this is different from CVE-2017-7617, which was only about the Party A buffer. | |||||
| CVE-2015-3008 | 1 Digium | 2 Asterisk, Certified Asterisk | 2018-10-09 | 4.3 MEDIUM | N/A |
| 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. | |||||
| CVE-2014-9374 | 1 Digium | 2 Asterisk, Certified Asterisk | 2018-10-09 | 5.0 MEDIUM | N/A |
| Double free vulnerability in the WebSocket Server (res_http_websocket module) in Asterisk Open Source 11.x before 11.14.2, 12.x before 12.7.2, and 13.x before 13.0.2 and Certified Asterisk 11.6 before 11.6-cert9 allows remote attackers to cause a denial of service (crash) by sending a zero length frame after a non-zero length frame. | |||||
| CVE-2014-4047 | 1 Digium | 2 Asterisk, Certified Asterisk | 2018-10-09 | 5.0 MEDIUM | N/A |
| 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. | |||||
| CVE-2014-4046 | 1 Digium | 2 Asterisk, Certified Asterisk | 2018-10-09 | 6.5 MEDIUM | N/A |
| 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. | |||||
| CVE-2017-17664 | 1 Digium | 2 Asterisk, Certified Asterisk | 2018-01-02 | 4.3 MEDIUM | 5.9 MEDIUM |
| A Remote Crash issue was discovered in Asterisk Open Source 13.x before 13.18.4, 14.x before 14.7.4, and 15.x before 15.1.4 and Certified Asterisk before 13.13-cert9. Certain compound RTCP packets cause a crash in the RTCP Stack. | |||||
| CVE-2012-2947 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2017-11-13 | 2.6 LOW | N/A |
| chan_iax2.c in the IAX2 channel driver in Certified Asterisk 1.8.11-cert before 1.8.11-cert2 and Asterisk Open Source 1.8.x before 1.8.12.1 and 10.x before 10.4.1, when a certain mohinterpret setting is enabled, allows remote attackers to cause a denial of service (daemon crash) by placing a call on hold. | |||||
| CVE-2017-14603 | 1 Digium | 2 Asterisk, Certified Asterisk | 2017-11-05 | 5.0 MEDIUM | 7.5 HIGH |
| 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. | |||||
| CVE-2017-9359 | 1 Digium | 2 Certified Asterisk, Open Source | 2017-11-04 | 5.0 MEDIUM | 7.5 HIGH |
| The multi-part body parser in PJSIP, as used in Asterisk Open Source 13.x before 13.15.1 and 14.x before 14.4.1, Certified Asterisk 13.13 before 13.13-cert4, and other products, allows remote attackers to cause a denial of service (out-of-bounds read and application crash) via a crafted packet. | |||||
| CVE-2017-9372 | 1 Digium | 2 Certified Asterisk, Open Source | 2017-11-04 | 5.0 MEDIUM | 7.5 HIGH |
| PJSIP, as used in Asterisk Open Source 13.x before 13.15.1 and 14.x before 14.4.1, Certified Asterisk 13.13 before 13.13-cert4, and other products, allows remote attackers to cause a denial of service (buffer overflow and application crash) via a SIP packet with a crafted CSeq header in conjunction with a Via header that lacks a branch parameter. | |||||
| CVE-2016-2316 | 2 Digium, Fedoraproject | 3 Asterisk, Certified Asterisk, Fedora | 2017-11-03 | 7.1 HIGH | 5.9 MEDIUM |
| 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. | |||||
| CVE-2016-2232 | 1 Digium | 2 Asterisk, Certified Asterisk | 2017-11-03 | 4.0 MEDIUM | 6.5 MEDIUM |
| 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. | |||||
| CVE-2017-14099 | 1 Digium | 2 Asterisk, Certified Asterisk | 2017-11-03 | 5.0 MEDIUM | 7.5 HIGH |
| 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. | |||||
| CVE-2013-7100 | 1 Digium | 3 Asterisk, Asterisk Digiumphones, Certified Asterisk | 2017-08-28 | 5.0 MEDIUM | N/A |
| Buffer overflow in the unpacksms16 function in apps/app_sms.c in Asterisk Open Source 1.8.x before 1.8.24.1, 10.x before 10.12.4, and 11.x before 11.6.1; Asterisk with Digiumphones 10.x-digiumphones before 10.12.4-digiumphones; and Certified Asterisk 1.8.x before 1.8.15-cert4 and 11.x before 11.2-cert3 allows remote attackers to cause a denial of service (daemon crash) via a 16-bit SMS message with an odd number of bytes, which triggers an infinite loop. | |||||
| CVE-2016-9938 | 1 Digium | 2 Asterisk, Certified Asterisk | 2017-07-26 | 5.0 MEDIUM | 5.3 MEDIUM |
| 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. | |||||
| CVE-2016-7551 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2017-04-24 | 5.0 MEDIUM | 7.5 HIGH |
| 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). | |||||
| CVE-2017-7617 | 1 Digium | 2 Asterisk, Certified Asterisk | 2017-04-17 | 6.5 MEDIUM | 8.8 HIGH |
| Remote code execution can occur in Asterisk Open Source 13.x before 13.14.1 and 14.x before 14.3.1 and Certified Asterisk 13.13 before 13.13-cert3 because of a buffer overflow in a CDR user field, related to X-ClientCode in chan_sip, the CDR dialplan function, and the AMI Monitor action. | |||||