Filtered by vendor Digium
Subscriptions
Total
119 CVE
| CVE | Vendors | Products | Updated | CVSS v3.1 |
|---|---|---|---|---|
| CVE-2014-6609 | 1 Digium | 1 Asterisk | 2024-08-06 | 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-2014-4048 | 1 Digium | 1 Asterisk | 2024-08-06 | 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-2014-4045 | 1 Digium | 1 Asterisk | 2024-08-06 | 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-2014-4047 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-08-06 | 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-2014-4046 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-08-06 | 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-2014-2286 | 2 Digium, Fedoraproject | 3 Asterisk, Certified Asterisk, Fedora | 2024-08-06 | 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-2014-2289 | 1 Digium | 1 Asterisk | 2024-08-06 | 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-2014-2288 | 1 Digium | 1 Asterisk | 2024-08-06 | 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 | 2024-08-06 | 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-2015-3008 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-08-06 | 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-2015-2690 | 1 Digium | 1 Addons Module | 2024-08-06 | N/A |
| Multiple cross-site scripting (XSS) vulnerabilities in views/add-license-form.php in the Digium Addons module (digiumaddoninstaller) before 2.11.0.7 for FreePBX allow remote attackers to inject arbitrary web script or HTML via the (1) add_license_key, (2) add_license_first_name, (3) add_license_last_name, (4) add_license_company, (5) add_license_address1, (6) add_license_address2, (7) add_license_city, (8) add_license_state, (9) add_license_post_code, (10) add_license_country, (11) add_license_phone, or (12) add_license_email parameter in an add-license-form page to admin/config.php. | ||||
| CVE-2015-1558 | 1 Digium | 1 Asterisk | 2024-08-06 | 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-2016-9937 | 1 Digium | 1 Asterisk | 2024-08-06 | 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-2016-9938 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-08-06 | 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-2016-7551 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-08-06 | N/A |
| chain_sip in Asterisk Open Source 11.x before 11.23.1 and 13.x 13.11.1 and Certified Asterisk 11.6 before 11.6-cert15 and 13.8 before 13.8-cert3 allows remote attackers to cause a denial of service (port exhaustion). | ||||
| CVE-2016-7550 | 1 Digium | 1 Asterisk | 2024-08-06 | N/A |
| asterisk 13.10.0 is affected by: denial of service issues in asterisk. The impact is: cause a denial of service (remote). | ||||
| CVE-2016-2232 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-08-05 | 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-2016-2316 | 2 Digium, Fedoraproject | 3 Asterisk, Certified Asterisk, Fedora | 2024-08-05 | 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-2017-17850 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-08-05 | N/A |
| An issue was discovered in Asterisk 13.18.4 and older, 14.7.4 and older, 15.1.4 and older, and 13.18-cert1 and older. A select set of SIP messages create a dialog in Asterisk. Those SIP messages must contain a contact header. For those messages, if the header was not present and the PJSIP channel driver was used, Asterisk would crash. The severity of this vulnerability is somewhat mitigated if authentication is enabled. If authentication is enabled, a user would have to first be authorized before reaching the crash point. | ||||
| CVE-2017-17664 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-08-05 | N/A |
| A Remote Crash issue was discovered in Asterisk Open Source 13.x before 13.18.4, 14.x before 14.7.4, and 15.x before 15.1.4 and Certified Asterisk before 13.13-cert9. Certain compound RTCP packets cause a crash in the RTCP Stack. | ||||