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Total
19 CVE
| CVE | Vendors | Products | Updated | CVSS v2 | CVSS v3 |
|---|---|---|---|---|---|
| CVE-2023-49786 | 2 Digium, Sangoma | 2 Asterisk, Certified Asterisk | 2023-12-29 | N/A | 5.9 MEDIUM |
| Asterisk is an open source private branch exchange and telephony toolkit. In Asterisk prior to versions 18.20.1, 20.5.1, and 21.0.1; as well as certified-asterisk prior to 18.9-cert6; Asterisk is susceptible to a DoS due to a race condition in the hello handshake phase of the DTLS protocol when handling DTLS-SRTP for media setup. This attack can be done continuously, thus denying new DTLS-SRTP encrypted calls during the attack. Abuse of this vulnerability may lead to a massive Denial of Service on vulnerable Asterisk servers for calls that rely on DTLS-SRTP. Commit d7d7764cb07c8a1872804321302ef93bf62cba05 contains a fix, which is part of versions 18.20.1, 20.5.1, 21.0.1, amd 18.9-cert6. | |||||
| CVE-2019-13161 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2022-06-01 | 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-2019-18790 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2022-05-13 | 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-2021-31878 | 1 Digium | 1 Asterisk | 2021-08-07 | 4.0 MEDIUM | 6.5 MEDIUM |
| An issue was discovered in PJSIP in Asterisk before 16.19.1 and before 18.5.1. To exploit, a re-INVITE without SDP must be received after Asterisk has sent a BYE request. | |||||
| CVE-2019-12827 | 1 Digium | 2 Asterisk, Certified Asterisk | 2021-07-21 | 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-2019-15297 | 1 Digium | 1 Asterisk | 2021-03-05 | 4.0 MEDIUM | 6.5 MEDIUM |
| res_pjsip_t38 in Sangoma Asterisk 13.21-cert4, 15.7.3, and 16.5.0 allows an attacker to trigger a crash by sending a declined stream in a response to a T.38 re-invite initiated by Asterisk. | |||||
| CVE-2021-26713 | 1 Digium | 2 Asterisk, Certified Asterisk | 2021-02-26 | 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-2020-35776 | 1 Digium | 1 Asterisk | 2021-02-24 | 4.3 MEDIUM | 6.5 MEDIUM |
| A buffer overflow in res_pjsip_diversion.c in Sangoma Asterisk versions 13.38.1, 16.15.1, 17.9.1, and 18.1.1 allows remote attacker to crash Asterisk by deliberately misusing SIP 181 responses. | |||||
| CVE-2021-26906 | 1 Digium | 2 Asterisk, Certified Asterisk | 2021-02-24 | 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-2020-35652 | 1 Digium | 1 Asterisk | 2021-02-04 | 4.0 MEDIUM | 6.5 MEDIUM |
| An issue was discovered in res_pjsip_diversion.c in Sangoma Asterisk before 13.38.0, 14.x through 16.x before 16.15.0, 17.x before 17.9.0, and 18.x before 18.1.0. A crash can occur when a SIP message is received with a History-Info header that contains a tel-uri, or when a SIP 181 response is received that contains a tel-uri in the Diversion header. | |||||
| CVE-2018-7287 | 1 Digium | 1 Asterisk | 2019-10-03 | 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 | 2019-10-03 | 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-2017-16672 | 1 Digium | 2 Asterisk, Certified Asterisk | 2019-10-03 | 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-2019-7251 | 1 Digium | 1 Asterisk | 2019-04-01 | 4.0 MEDIUM | 6.5 MEDIUM |
| 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. | |||||
| CVE-2018-12227 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2019-03-29 | 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-2017-17664 | 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-2016-2232 | 1 Digium | 2 Asterisk, Certified Asterisk | 2017-11-04 | 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 | 2017-11-04 | 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-2016-9938 | 1 Digium | 2 Asterisk, Certified Asterisk | 2017-07-27 | 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. | |||||