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-2016-9938 | 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-2014-4046 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-12 | N/A |
| Asterisk Open Source 11.x before 11.10.1 and 12.x before 12.3.1 and Certified Asterisk 11.6 before 11.6-cert3 allows remote authenticated Manager users to execute arbitrary shell commands via a MixMonitor action. | ||||
| 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-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-2014-2286 | 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-2014-6609 | 1 Digium | 1 Asterisk | 2025-04-12 | N/A |
| The res_pjsip_pubsub module in Asterisk Open Source 12.x before 12.5.1 allows remote authenticated users to cause a denial of service (crash) via crafted headers in a SIP SUBSCRIBE request for an event package. | ||||
| CVE-2014-4047 | 1 Digium | 2 Asterisk, Certified Asterisk | 2025-04-12 | N/A |
| Asterisk Open Source 1.8.x before 1.8.28.1, 11.x before 11.10.1, and 12.x before 12.3.1 and Certified Asterisk 1.8.15 before 1.8.15-cert6 and 11.6 before 11.6-cert3 allows remote attackers to cause a denial of service (connection consumption) via a large number of (1) inactive or (2) incomplete HTTP connections. | ||||
| 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-2014-8414 | 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-2014-8416 | 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-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-2014-6610 | 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-2010-0685 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| The design of the dialplan functionality in Asterisk Open Source 1.2.x, 1.4.x, and 1.6.x; and Asterisk Business Edition B.x.x and C.x.x, when using the ${EXTEN} channel variable and wildcard pattern matches, allows context-dependent attackers to inject strings into the dialplan using metacharacters that are injected when the variable is expanded, as demonstrated using the Dial application to process a crafted SIP INVITE message that adds an unintended outgoing channel leg. NOTE: it could be argued that this is not a vulnerability in Asterisk, but a class of vulnerabilities that can occur in any program that uses this feature without the associated filtering functionality that is already available. | ||||
| 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-2012-3553 | 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-2012-5976 | 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-2012-1184 | 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. | ||||