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-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-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-2014-2289 | 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-2014-4048 | 1 Digium | 1 Asterisk | 2025-04-12 | N/A |
| The PJSIP Channel Driver in Asterisk Open Source before 12.3.1 allows remote attackers to cause a denial of service (deadlock) by terminating a subscription request before it is complete, which triggers a SIP transaction timeout. | ||||
| 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-2014-8416 | 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-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-2014-6610 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-12 | N/A |
| Asterisk Open Source 11.x before 11.12.1 and 12.x before 12.5.1 and Certified Asterisk 11.6 before 11.6-cert6, when using the res_fax_spandsp module, allows remote authenticated users to cause a denial of service (crash) via an out of call message, which is not properly handled in the ReceiveFax dialplan application. | ||||
| 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-2014-8414 | 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-2014-2287 | 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-2014-8412 | 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-2015-3008 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-12 | N/A |
| Asterisk Open Source 1.8 before 1.8.32.3, 11.x before 11.17.1, 12.x before 12.8.2, and 13.x before 13.3.2 and Certified Asterisk 1.8.28 before 1.8.28-cert5, 11.6 before 11.6-cert11, and 13.1 before 13.1-cert2, when registering a SIP TLS device, does not properly handle a null byte in a domain name in the subject's Common Name (CN) field of an X.509 certificate, which allows man-in-the-middle attackers to spoof arbitrary SSL servers via a crafted certificate issued by a legitimate Certification Authority. | ||||
| 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-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-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-2011-2216 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| reqresp_parser.c in the SIP channel driver in Asterisk Open Source 1.8.x before 1.8.4.2 does not initialize certain strings, which allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) via a malformed Contact header. | ||||
| CVE-2011-1507 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| Asterisk Open Source 1.4.x before 1.4.40.1, 1.6.1.x before 1.6.1.25, 1.6.2.x before 1.6.2.17.3, and 1.8.x before 1.8.3.3 and Asterisk Business Edition C.x.x before C.3.6.4 do not restrict the number of unauthenticated sessions to certain interfaces, which allows remote attackers to cause a denial of service (file descriptor exhaustion and disk space exhaustion) via a series of TCP connections. | ||||
| 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-2011-1174 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| manager.c in Asterisk Open Source 1.6.1.x before 1.6.1.24, 1.6.2.x before 1.6.2.17.2, and 1.8.x before 1.8.3.2 allows remote attackers to cause a denial of service (CPU and memory consumption) via a series of manager sessions involving invalid data. | ||||