Total
51 CVE
| CVE | Vendors | Products | Updated | CVSS v2 | CVSS v3 |
|---|---|---|---|---|---|
| CVE-2022-26651 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 7.5 HIGH | 9.8 CRITICAL |
| An issue was discovered in Asterisk through 19.x and Certified Asterisk through 16.8-cert13. The func_odbc module provides possibly inadequate escaping functionality for backslash characters in SQL queries, resulting in user-provided data creating a broken SQL query or possibly a SQL injection. This is fixed in 16.25.2, 18.11.2, and 19.3.2, and 16.8-cert14. | |||||
| CVE-2021-32558 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 5.0 MEDIUM | 7.5 HIGH |
| An issue was discovered in Sangoma Asterisk 13.x before 13.38.3, 16.x before 16.19.1, 17.x before 17.9.4, and 18.x before 18.5.1, and Certified Asterisk before 16.8-cert10. If the IAX2 channel driver receives a packet that contains an unsupported media format, a crash can occur. | |||||
| CVE-2021-26906 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 4.3 MEDIUM | 5.9 MEDIUM |
| An issue was discovered in res_pjsip_session.c in Digium Asterisk through 13.38.1; 14.x, 15.x, and 16.x through 16.16.0; 17.x through 17.9.1; and 18.x through 18.2.0, and Certified Asterisk through 16.8-cert5. An SDP negotiation vulnerability in PJSIP allows a remote server to potentially crash Asterisk by sending specific SIP responses that cause an SDP negotiation failure. | |||||
| CVE-2021-26717 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 5.0 MEDIUM | 7.5 HIGH |
| An issue was discovered in Sangoma Asterisk 16.x before 16.16.1, 17.x before 17.9.2, and 18.x before 18.2.1 and Certified Asterisk before 16.8-cert6. When re-negotiating for T.38, if the initial remote response was delayed just enough, Asterisk would send both audio and T.38 in the SDP. If this happened, and the remote responded with a declined T.38 stream, then Asterisk would crash. | |||||
| CVE-2021-26713 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 4.0 MEDIUM | 6.5 MEDIUM |
| A stack-based buffer overflow in res_rtp_asterisk.c in Sangoma Asterisk before 16.16.1, 17.x before 17.9.2, and 18.x before 18.2.1 and Certified Asterisk before 16.8-cert6 allows an authenticated WebRTC client to cause an Asterisk crash by sending multiple hold/unhold requests in quick succession. This is caused by a signedness comparison mismatch. | |||||
| CVE-2021-26712 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 5.0 MEDIUM | 7.5 HIGH |
| Incorrect access controls in res_srtp.c in Sangoma Asterisk 13.38.1, 16.16.0, 17.9.1, and 18.2.0 and Certified Asterisk 16.8-cert5 allow a remote unauthenticated attacker to prematurely terminate secure calls by replaying SRTP packets. | |||||
| CVE-2020-28327 | 2 Digium, Sangoma | 2 Certified Asterisk, Asterisk | 2024-11-21 | 2.1 LOW | 5.3 MEDIUM |
| A res_pjsip_session crash was discovered in Asterisk Open Source 13.x before 13.37.1, 16.x before 16.14.1, 17.x before 17.8.1, and 18.x before 18.0.1. and Certified Asterisk before 16.8-cert5. Upon receiving a new SIP Invite, Asterisk did not return the created dialog locked or referenced. This caused a gap between the creation of the dialog object, and its next use by the thread that created it. Depending on some off-nominal circumstances and timing, it was possible for another thread to free said dialog in this gap. Asterisk could then crash when the dialog object, or any of its dependent objects, were dereferenced or accessed next by the initial-creation thread. Note, however, that this crash can only occur when using a connection-oriented protocol (e.g., TCP or TLS, but not UDP) for SIP transport. Also, the remote client must be authenticated, or Asterisk must be configured for anonymous calling. | |||||
| CVE-2019-18976 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 5.0 MEDIUM | 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 | 2024-11-21 | 5.8 MEDIUM | 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 | 2024-11-21 | 9.0 HIGH | 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-2019-13161 | 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-2018-7286 | 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-2018-7284 | 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-2018-17281 | 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. | |||||