Total
110 CVE
| CVE | Vendors | Products | Updated | CVSS v2 | CVSS v3 |
|---|---|---|---|---|---|
| CVE-2014-4048 | 1 Digium | 1 Asterisk | 2018-10-09 | 4.3 MEDIUM | N/A |
| The PJSIP Channel Driver in Asterisk Open Source before 12.3.1 allows remote attackers to cause a denial of service (deadlock) by terminating a subscription request before it is complete, which triggers a SIP transaction timeout. | |||||
| CVE-2011-2216 | 1 Digium | 1 Asterisk | 2018-10-09 | 5.0 MEDIUM | N/A |
| reqresp_parser.c in the SIP channel driver in Asterisk Open Source 1.8.x before 1.8.4.2 does not initialize certain strings, which allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) via a malformed Contact header. | |||||
| CVE-2018-7285 | 1 Digium | 1 Asterisk | 2018-03-21 | 5.0 MEDIUM | 7.5 HIGH |
| A NULL pointer access issue was discovered in Asterisk 15.x through 15.2.1. The RTP support in Asterisk maintains its own registry of dynamic codecs and desired payload numbers. While an SDP negotiation may result in a codec using a different payload number, these desired ones are still stored internally. When an RTP packet was received, this registry would be consulted if the payload number was not found in the negotiated SDP. This registry was incorrectly consulted for all packets, even those which are dynamic. If the payload number resulted in a codec of a different type than the RTP stream (for example, the payload number resulted in a video codec but the stream carried audio), a crash could occur if no stream of that type had been negotiated. This was due to the code incorrectly assuming that a stream of that type would always exist. | |||||
| 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-2017-14099 | 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-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-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-2017-14098 | 1 Digium | 1 Asterisk | 2017-09-14 | 5.0 MEDIUM | 7.5 HIGH |
| In the pjsip channel driver (res_pjsip) in Asterisk 13.x before 13.17.1 and 14.x before 14.6.1, a carefully crafted tel URI in a From, To, or Contact header could cause Asterisk to crash. | |||||
| 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-2012-1184 | 1 Digium | 1 Asterisk | 2017-08-28 | 7.5 HIGH | N/A |
| Stack-based buffer overflow in the ast_parse_digest function in main/utils.c in Asterisk 1.8.x before 1.8.10.1 and 10.x before 10.2.1 allows remote attackers to cause a denial of service (crash) or possibly execute arbitrary code via a long string in an HTTP Digest Authentication header. | |||||
| CVE-2011-2666 | 1 Digium | 1 Asterisk | 2017-08-28 | 5.0 MEDIUM | N/A |
| The default configuration of the SIP channel driver in Asterisk Open Source 1.4.x through 1.4.41.2 and 1.6.2.x through 1.6.2.18.2 does not enable the alwaysauthreject option, which allows remote attackers to enumerate account names by making a series of invalid SIP requests and observing the differences in the responses for different usernames, a different vulnerability than CVE-2011-2536. | |||||
| CVE-2011-2535 | 1 Digium | 1 Asterisk | 2017-08-28 | 5.0 MEDIUM | N/A |
| chan_iax2.c in the IAX2 channel driver in Asterisk Open Source 1.4.x before 1.4.41.1, 1.6.2.x before 1.6.2.18.1, and 1.8.x before 1.8.4.3, and Asterisk Business Edition C.3 before C.3.7.3, accesses a memory address contained in an option control frame, which allows remote attackers to cause a denial of service (daemon crash) or possibly have unspecified other impact via a crafted frame. | |||||
| CVE-2011-2529 | 1 Digium | 1 Asterisk | 2017-08-28 | 5.0 MEDIUM | N/A |
| chan_sip.c in the SIP channel driver in Asterisk Open Source 1.6.x before 1.6.2.18.1 and 1.8.x before 1.8.4.3 does not properly handle '\0' characters in SIP packets, which allows remote attackers to cause a denial of service (memory corruption) or possibly have unspecified other impact via a crafted packet. | |||||
| CVE-2011-1174 | 1 Digium | 1 Asterisk | 2017-08-16 | 5.0 MEDIUM | N/A |
| manager.c in Asterisk Open Source 1.6.1.x before 1.6.1.24, 1.6.2.x before 1.6.2.17.2, and 1.8.x before 1.8.3.2 allows remote attackers to cause a denial of service (CPU and memory consumption) via a series of manager sessions involving invalid data. | |||||
| CVE-2011-1175 | 1 Digium | 1 Asterisk | 2017-08-16 | 5.0 MEDIUM | N/A |
| tcptls.c in the TCP/TLS server in Asterisk Open Source 1.6.1.x before 1.6.1.23, 1.6.2.x before 1.6.2.17.1, and 1.8.x before 1.8.3.1 allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) by establishing many short TCP sessions to services that use a certain TLS API. | |||||
| CVE-2009-2651 | 1 Digium | 1 Asterisk | 2017-08-16 | 5.0 MEDIUM | N/A |
| main/rtp.c in Asterisk Open Source 1.6.1 before 1.6.1.2 allows remote attackers to cause a denial of service (crash) via an RTP text frame without a certain delimiter, which triggers a NULL pointer dereference and the subsequent calculation of an invalid pointer. | |||||
| CVE-2007-1306 | 1 Digium | 1 Asterisk | 2017-07-28 | 7.8 HIGH | N/A |
| Asterisk 1.4 before 1.4.1 and 1.2 before 1.2.16 allows remote attackers to cause a denial of service (crash) by sending a Session Initiation Protocol (SIP) packet without a URI and SIP-version header, which results in a NULL pointer dereference. | |||||
| 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. | |||||