Filtered by vendor Digium
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Total
115 CVE
| CVE | Vendors | Products | Updated | CVSS v2 | CVSS v3 |
|---|---|---|---|---|---|
| CVE-2014-4045 | 1 Digium | 1 Asterisk | 2018-10-09 | 4.3 MEDIUM | N/A |
| The Publish/Subscribe Framework in the PJSIP channel driver in Asterisk Open Source 12.x before 12.3.1, when sub_min_expiry is set to zero, allows remote attackers to cause a denial of service (assertion failure and crash) via an unsubscribe request when not subscribed to the device. | |||||
| 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-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-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-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-2017-9372 | 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-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-2017-14099 | 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-2011-2529 | 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-2011-2535 | 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-2011-2666 | 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. | |||||