Filtered by vendor Digium
Subscriptions
Filtered by product Asterisk
Subscriptions
Total
114 CVE
| CVE | Vendors | Products | Updated | CVSS v3.1 |
|---|---|---|---|---|
| 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-2014-8413 | 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-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-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-2014-4048 | 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-2014-2286 | 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-2014-4045 | 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-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-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-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-2014-8417 | 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-2016-9937 | 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-2014-6610 | 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-2016-9938 | 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-2014-8414 | 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-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. | ||||
| 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-2010-1224 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| main/acl.c in Asterisk Open Source 1.6.0.x before 1.6.0.25, 1.6.1.x before 1.6.1.17, and 1.6.2.x before 1.6.2.5 does not properly enforce remote host access controls when CIDR notation "/0" is used in permit= and deny= configuration rules, which causes an improper arithmetic shift and might allow remote attackers to bypass ACL rules and access services from unauthorized hosts. | ||||
| CVE-2011-1175 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| tcptls.c in the TCP/TLS server in Asterisk Open Source 1.6.1.x before 1.6.1.23, 1.6.2.x before 1.6.2.17.1, and 1.8.x before 1.8.3.1 allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) by establishing many short TCP sessions to services that use a certain TLS API. | ||||
| 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. | ||||