Filtered by vendor Digium
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Filtered by product Asterisk
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Total
114 CVE
| CVE | Vendors | Products | Updated | CVSS v3.1 |
|---|---|---|---|---|
| CVE-2017-16672 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-08-05 | N/A |
| 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-2017-14603 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-08-05 | N/A |
| 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-2017-14098 | 1 Digium | 1 Asterisk | 2024-08-05 | N/A |
| 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-2017-14100 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-08-05 | N/A |
| 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. | ||||
| CVE-2017-14099 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-08-05 | N/A |
| 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-2017-7617 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-08-05 | N/A |
| 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-2018-17281 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-08-05 | N/A |
| 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 | 2024-08-05 | N/A |
| 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 | 2024-08-05 | N/A |
| 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-2018-7287 | 1 Digium | 1 Asterisk | 2024-08-05 | N/A |
| 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-2018-7285 | 1 Digium | 1 Asterisk | 2024-08-05 | N/A |
| 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-2018-7286 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-08-05 | N/A |
| 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-2019-18976 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-08-05 | 7.5 High |
| An issue was discovered in res_pjsip_t38.c in Sangoma Asterisk through 13.x and Certified Asterisk through 13.21-x. If it receives a re-invite initiating T.38 faxing and has a port of 0 and no c line in the SDP, a NULL pointer dereference and crash will occur. This is different from CVE-2019-18940. | ||||
| CVE-2019-18790 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-08-05 | 6.5 Medium |
| An issue was discovered in channels/chan_sip.c in Sangoma Asterisk 13.x before 13.29.2, 16.x before 16.6.2, and 17.x before 17.0.1, and Certified Asterisk 13.21 before cert5. A SIP request can be sent to Asterisk that can change a SIP peer's IP address. A REGISTER does not need to occur, and calls can be hijacked as a result. The only thing that needs to be known is the peer's name; authentication details such as passwords do not need to be known. This vulnerability is only exploitable when the nat option is set to the default, or auto_force_rport. | ||||
| CVE-2019-18610 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-08-05 | 8.8 High |
| An issue was discovered in manager.c in Sangoma Asterisk through 13.x, 16.x, 17.x and Certified Asterisk 13.21 through 13.21-cert4. A remote authenticated Asterisk Manager Interface (AMI) user without system authorization could use a specially crafted Originate AMI request to execute arbitrary system commands. | ||||
| CVE-2019-15639 | 1 Digium | 1 Asterisk | 2024-08-05 | 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 | 2024-08-05 | 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 | 2024-08-04 | 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 | 2024-08-04 | N/A |
| 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-2019-7251 | 1 Digium | 1 Asterisk | 2024-08-04 | N/A |
| An Integer Signedness issue (for a return code) in the res_pjsip_sdp_rtp module in Digium Asterisk versions 15.7.1 and earlier and 16.1.1 and earlier allows remote authenticated users to crash Asterisk via a specially crafted SDP protocol violation. | ||||