Filtered by vendor Digium
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Filtered by product Certified Asterisk
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Total
51 CVE
| CVE | Vendors | Products | Updated | CVSS v3.1 |
|---|---|---|---|---|
| CVE-2014-8418 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-12 | N/A |
| The DB dialplan function in Asterisk Open Source 1.8.x before 1.8.32, 11.x before 11.1.4.1, 12.x before 12.7.1, and 13.x before 13.0.1 and Certified Asterisk 1.8 before 1.8.28-cert8 and 11.6 before 11.6-cert8 allows remote authenticated users to gain privileges via a call from an external protocol, as demonstrated by the AMI protocol. | ||||
| CVE-2014-9374 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-12 | N/A |
| Double free vulnerability in the WebSocket Server (res_http_websocket module) in Asterisk Open Source 11.x before 11.14.2, 12.x before 12.7.2, and 13.x before 13.0.2 and Certified Asterisk 11.6 before 11.6-cert9 allows remote attackers to cause a denial of service (crash) by sending a zero length frame after a non-zero length frame. | ||||
| CVE-2016-2232 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-12 | N/A |
| 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-2016-9938 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-12 | N/A |
| 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. | ||||
| CVE-2014-8417 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-12 | N/A |
| ConfBridge in Asterisk 11.x before 11.14.1, 12.x before 12.7.1, and 13.x before 13.0.1 and Certified Asterisk 11.6 before 11.6-cert8 allows remote authenticated users to (1) gain privileges via vectors related to an external protocol to the CONFBRIDGE dialplan function or (2) execute arbitrary system commands via a crafted ConfbridgeStartRecord AMI action. | ||||
| CVE-2016-2316 | 2 Digium, Fedoraproject | 3 Asterisk, Certified Asterisk, Fedora | 2025-04-12 | N/A |
| 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-2012-5977 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-11 | N/A |
| Asterisk Open Source 1.8.x before 1.8.19.1, 10.x before 10.11.1, and 11.x before 11.1.2; Certified Asterisk 1.8.11 before 1.8.11-cert10; and Asterisk Digiumphones 10.x-digiumphones before 10.11.1-digiumphones, when anonymous calls are enabled, allow remote attackers to cause a denial of service (resource consumption) by making anonymous calls from multiple sources and consequently adding many entries to the device state cache. | ||||
| CVE-2012-5976 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-11 | N/A |
| Multiple stack consumption vulnerabilities in Asterisk Open Source 1.8.x before 1.8.19.1, 10.x before 10.11.1, and 11.x before 11.1.2; Certified Asterisk 1.8.11 before 1.8.11-cert10; and Asterisk Digiumphones 10.x-digiumphones before 10.11.1-digiumphones allow remote attackers to cause a denial of service (daemon crash) via TCP data using the (1) SIP, (2) HTTP, or (3) XMPP protocol. | ||||
| CVE-2012-2947 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2025-04-11 | 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-2012-3812 | 1 Digium | 3 Asterisk, Asteriske, Certified Asterisk | 2025-04-11 | N/A |
| Double free vulnerability in apps/app_voicemail.c in Asterisk Open Source 1.8.x before 1.8.13.1 and 10.x before 10.5.2, Certified Asterisk 1.8.11-certx before 1.8.11-cert4, and Asterisk Digiumphones 10.x.x-digiumphones before 10.5.2-digiumphones allows remote authenticated users to cause a denial of service (daemon crash) by establishing multiple voicemail sessions and accessing both the Urgent mailbox and the INBOX mailbox. | ||||
| CVE-2012-3863 | 1 Digium | 4 Asterisk, Asterisk Business Edition, Asteriske and 1 more | 2025-04-11 | N/A |
| channels/chan_sip.c in Asterisk Open Source 1.8.x before 1.8.13.1 and 10.x before 10.5.2, Asterisk Business Edition C.3.x before C.3.7.5, Certified Asterisk 1.8.11-certx before 1.8.11-cert4, and Asterisk Digiumphones 10.x.x-digiumphones before 10.5.2-digiumphones does not properly handle a provisional response to a SIP reINVITE request, which allows remote authenticated users to cause a denial of service (RTP port exhaustion) via sessions that lack final responses. | ||||
| CVE-2012-4737 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-11 | N/A |
| channels/chan_iax2.c in Asterisk Open Source 1.8.x before 1.8.15.1 and 10.x before 10.7.1, Certified Asterisk 1.8.11 before 1.8.11-cert7, Asterisk Digiumphones 10.x.x-digiumphones before 10.7.1-digiumphones, and Asterisk Business Edition C.3.x before C.3.7.6 does not enforce ACL rules during certain uses of peer credentials, which allows remote authenticated users to bypass intended outbound-call restrictions by leveraging the availability of these credentials. | ||||
| CVE-2013-5641 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-11 | N/A |
| The SIP channel driver (channels/chan_sip.c) in Asterisk Open Source 1.8.17.x through 1.8.22.x, 1.8.23.x before 1.8.23.1, and 11.x before 11.5.1 and Certified Asterisk 1.8.15 before 1.8.15-cert3 and 11.2 before 11.2-cert2 allows remote attackers to cause a denial of service (NULL pointer dereference, segmentation fault, and daemon crash) via an ACK with SDP to a previously terminated channel. NOTE: some of these details are obtained from third party information. | ||||
| CVE-2013-7100 | 1 Digium | 3 Asterisk, Asterisk Digiumphones, Certified Asterisk | 2025-04-11 | 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-2013-5642 | 1 Digium | 3 Asterisk, Asterisk Digiumphones, Certified Asterisk | 2025-04-11 | N/A |
| The SIP channel driver (channels/chan_sip.c) in Asterisk Open Source 1.8.x before 1.8.23.1, 10.x before 10.12.3, and 11.x before 11.5.1; Certified Asterisk 1.8.15 before 1.8.15-cert3 and 11.2 before 11.2-cert2; and Asterisk Digiumphones 10.x-digiumphones before 10.12.3-digiumphones allows remote attackers to cause a denial of service (NULL pointer dereference, segmentation fault, and daemon crash) via an invalid SDP that defines a media description before the connection description in a SIP request. | ||||
| CVE-2022-26651 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 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 | 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 | 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 | 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 | 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. | ||||