Filtered by vendor Digium
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Total
119 CVE
| CVE | Vendors | Products | Updated | CVSS v2 | CVSS v3 |
|---|---|---|---|---|---|
| CVE-2019-15639 | 1 Digium | 1 Asterisk | 2024-11-21 | 5.0 MEDIUM | 7.5 HIGH |
| main/translate.c in Sangoma Asterisk 13.28.0 and 16.5.0 allows a remote attacker to send a specific RTP packet during a call and cause a crash in a specific scenario. | |||||
| CVE-2019-15297 | 1 Digium | 1 Asterisk | 2024-11-21 | 4.0 MEDIUM | 6.5 MEDIUM |
| res_pjsip_t38 in Sangoma Asterisk 15.x before 15.7.4 and 16.x before 16.5.1 allows an attacker to trigger a crash by sending a declined stream in a response to a T.38 re-invite initiated by Asterisk. The crash occurs because of a NULL session media object dereference. | |||||
| CVE-2019-13161 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 3.5 LOW | 5.3 MEDIUM |
| 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). | |||||
| CVE-2019-12827 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 4.0 MEDIUM | 6.5 MEDIUM |
| Buffer overflow in res_pjsip_messaging in Digium Asterisk versions 13.21-cert3, 13.27.0, 15.7.2, 16.4.0 and earlier allows remote authenticated users to crash Asterisk by sending a specially crafted SIP MESSAGE message. | |||||
| CVE-2018-7287 | 1 Digium | 1 Asterisk | 2024-11-21 | 4.3 MEDIUM | 5.9 MEDIUM |
| An issue was discovered in res_http_websocket.c in Asterisk 15.x through 15.2.1. If the HTTP server is enabled (default is disabled), WebSocket payloads of size 0 are mishandled (with a busy loop). | |||||
| CVE-2018-7286 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 4.0 MEDIUM | 6.5 MEDIUM |
| An 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. res_pjsip allows remote authenticated users to crash Asterisk (segmentation fault) by sending a number of SIP INVITE messages on a TCP or TLS connection and then suddenly closing the connection. | |||||
| CVE-2018-7285 | 1 Digium | 1 Asterisk | 2024-11-21 | 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-2018-7284 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 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-2018-19278 | 1 Digium | 1 Asterisk | 2024-11-21 | 5.0 MEDIUM | 7.5 HIGH |
| Buffer overflow in DNS SRV and NAPTR lookups in Digium Asterisk 15.x before 15.6.2 and 16.x before 16.0.1 allows remote attackers to crash Asterisk via a specially crafted DNS SRV or NAPTR response, because a buffer size is supposed to match an expanded length but actually matches a compressed length. | |||||
| CVE-2018-17281 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 5.0 MEDIUM | 7.5 HIGH |
| There is a stack consumption vulnerability in the res_http_websocket.so module of Asterisk through 13.23.0, 14.7.x through 14.7.7, and 15.x through 15.6.0 and Certified Asterisk through 13.21-cert2. It allows an attacker to crash Asterisk via a specially crafted HTTP request to upgrade the connection to a websocket. | |||||
| CVE-2018-12227 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 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-2017-9372 | 1 Digium | 2 Certified Asterisk, Open Source | 2024-11-21 | 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 | 2024-11-21 | 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-2017-7617 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 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. | |||||
| CVE-2017-17850 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 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-2017-17664 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 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-2017-17090 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 5.0 MEDIUM | 7.5 HIGH |
| An issue was discovered in chan_skinny.c in Asterisk Open Source 13.18.2 and older, 14.7.2 and older, and 15.1.2 and older, and Certified Asterisk 13.13-cert7 and older. If the chan_skinny (aka SCCP protocol) channel driver is flooded with certain requests, it can cause the asterisk process to use excessive amounts of virtual memory, eventually causing asterisk to stop processing requests of any kind. | |||||
| CVE-2017-16672 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 4.3 MEDIUM | 5.9 MEDIUM |
| An 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. A memory leak occurs when an Asterisk pjsip session object is created and that call gets rejected before the session itself is fully established. When this happens the session object never gets destroyed. Eventually Asterisk can run out of memory and crash. | |||||
| CVE-2017-16671 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 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-2017-14603 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 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. | |||||