Filtered by vendor Digium
Subscriptions
Filtered by product Asterisk
Subscriptions
Total
114 CVE
| CVE | Vendors | Products | Updated | CVSS v3.1 |
|---|---|---|---|---|
| CVE-2016-9937 | 1 Digium | 1 Asterisk | 2025-04-12 | N/A |
| An issue was discovered in Asterisk Open Source 13.12.x and 13.13.x before 13.13.1 and 14.x before 14.2.1. If an SDP offer or answer is received with the Opus codec and with the format parameters separated using a space the code responsible for parsing will recursively call itself until it crashes. This occurs as the code does not properly handle spaces separating the parameters. This does NOT require the endpoint to have Opus configured in Asterisk. This also does not require the endpoint to be authenticated. If guest is enabled for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still processed and the crash occurs. | ||||
| 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-2014-8412 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-12 | N/A |
| The (1) VoIP channel drivers, (2) DUNDi, and (3) Asterisk Manager Interface (AMI) in Asterisk Open Source 1.8.x before 1.8.32.1, 11.x before 11.14.1, 12.x before 12.7.1, and 13.x before 13.0.1 and Certified Asterisk 1.8.28 before 1.8.28-cert3 and 11.6 before 11.6-cert8 allows remote attackers to bypass the ACL restrictions via a packet with a source IP that does not share the address family as the first ACL entry. | ||||
| CVE-2014-2289 | 1 Digium | 1 Asterisk | 2025-04-12 | N/A |
| res/res_pjsip_exten_state.c in the PJSIP channel driver in Asterisk Open Source 12.x before 12.1.0 allows remote authenticated users to cause a denial of service (crash) via a SUBSCRIBE request without any Accept headers, which triggers an invalid pointer dereference. | ||||
| CVE-2014-2286 | 2 Digium, Fedoraproject | 3 Asterisk, Certified Asterisk, Fedora | 2025-04-12 | N/A |
| main/http.c in Asterisk Open Source 1.8.x before 1.8.26.1, 11.8.x before 11.8.1, and 12.1.x before 12.1.1, and Certified Asterisk 1.8.x before 1.8.15-cert5 and 11.6 before 11.6-cert2, allows remote attackers to cause a denial of service (stack consumption) and possibly execute arbitrary code via an HTTP request with a large number of Cookie headers. | ||||
| CVE-2014-2288 | 1 Digium | 1 Asterisk | 2025-04-12 | N/A |
| The PJSIP channel driver in Asterisk Open Source 12.x before 12.1.1, when qualify_frequency "is enabled on an AOR and the remote SIP server challenges for authentication of the resulting OPTIONS request," allows remote attackers to cause a denial of service (crash) via a PJSIP endpoint that does not have an associated outgoing request. | ||||
| CVE-2014-2287 | 2 Digium, Fedoraproject | 3 Asterisk, Certified Asterisk, Fedora | 2025-04-12 | N/A |
| channels/chan_sip.c in Asterisk Open Source 1.8.x before 1.8.26.1, 11.8.x before 11.8.1, and 12.1.x before 12.1.1, and Certified Asterisk 1.8.15 before 1.8.15-cert5 and 11.6 before 11.6-cert2, when chan_sip has a certain configuration, allows remote authenticated users to cause a denial of service (channel and file descriptor consumption) via an INVITE request with a (1) Session-Expires or (2) Min-SE header with a malformed or invalid value. | ||||
| CVE-2014-8416 | 1 Digium | 1 Asterisk | 2025-04-12 | N/A |
| Use-after-free vulnerability in the PJSIP channel driver in Asterisk Open Source 12.x before 12.7.1 and 13.x before 13.0.1, when using the res_pjsip_refer module, allows remote attackers to cause a denial of service (crash) via an in-dialog INVITE with Replaces message, which triggers the channel to be hung up. | ||||
| CVE-2014-8415 | 1 Digium | 1 Asterisk | 2025-04-12 | N/A |
| Race condition in the chan_pjsip channel driver in Asterisk Open Source 12.x before 12.7.1 and 13.x before 13.0.1 allows remote attackers to cause a denial of service (assertion failure and crash) via a cancel request for a SIP session with a queued action to (1) answer a session or (2) send ringing. | ||||
| CVE-2014-8413 | 1 Digium | 1 Asterisk | 2025-04-12 | N/A |
| The res_pjsip_acl module in Asterisk Open Source 12.x before 12.7.1 and 13.x before 13.0.1 does not properly create and load ACLs defined in pjsip.conf at startup, which allows remote attackers to bypass intended PJSIP ACL rules. | ||||
| CVE-2014-8414 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-12 | N/A |
| ConfBridge in Asterisk 11.x before 11.14.1 and Certified Asterisk 11.6 before 11.6-cert8 does not properly handle state changes, which allows remote attackers to cause a denial of service (channel hang and memory consumption) by causing transitions to be delayed, which triggers a state change from hung up to waiting for media. | ||||
| 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-2014-4045 | 1 Digium | 1 Asterisk | 2025-04-12 | N/A |
| The Publish/Subscribe Framework in the PJSIP channel driver in Asterisk Open Source 12.x before 12.3.1, when sub_min_expiry is set to zero, allows remote attackers to cause a denial of service (assertion failure and crash) via an unsubscribe request when not subscribed to the device. | ||||
| CVE-2015-1558 | 1 Digium | 1 Asterisk | 2025-04-12 | N/A |
| Asterisk Open Source 12.x before 12.8.1 and 13.x before 13.1.1, when using the PJSIP channel driver, does not properly reclaim RTP ports, which allows remote authenticated users to cause a denial of service (file descriptor consumption) via an SDP offer containing only incompatible codecs. | ||||
| 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-2011-4597 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| The SIP over UDP implementation in Asterisk Open Source 1.4.x before 1.4.43, 1.6.x before 1.6.2.21, and 1.8.x before 1.8.7.2 uses different port numbers for responses to invalid requests depending on whether a SIP username exists, which allows remote attackers to enumerate usernames via a series of requests. | ||||
| CVE-2012-3553 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| chan_skinny.c in the Skinny (aka SCCP) channel driver in Asterisk Open Source 10.x before 10.5.1 allows remote authenticated users to cause a denial of service (NULL pointer dereference and daemon crash) by sending a Station Key Pad Button message and closing a connection in off-hook mode, a related issue to CVE-2012-2948. | ||||
| 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-2012-1184 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| Stack-based buffer overflow in the ast_parse_digest function in main/utils.c in Asterisk 1.8.x before 1.8.10.1 and 10.x before 10.2.1 allows remote attackers to cause a denial of service (crash) or possibly execute arbitrary code via a long string in an HTTP Digest Authentication header. | ||||
| 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. | ||||