Filtered by vendor Digium
Subscriptions
Filtered by product Asterisk
Subscriptions
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-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-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-2014-8416 | 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-2014-2289 | 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-2014-8413 | 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-2014-6610 | 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-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-2014-8417 | 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-2014-4047 | 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-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-2014-8415 | 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-2014-2288 | 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-2011-2666 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| The default configuration of the SIP channel driver in Asterisk Open Source 1.4.x through 1.4.41.2 and 1.6.2.x through 1.6.2.18.2 does not enable the alwaysauthreject option, which allows remote attackers to enumerate account names by making a series of invalid SIP requests and observing the differences in the responses for different usernames, a different vulnerability than CVE-2011-2536. | ||||
| CVE-2011-2665 | 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 1.8.4.3 allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) via a SIP packet with a Contact header that lacks a < (less than) character. | ||||
| CVE-2011-2536 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| chan_sip.c in the SIP channel driver in Asterisk Open Source 1.4.x before 1.4.41.2, 1.6.2.x before 1.6.2.18.2, and 1.8.x before 1.8.4.4, and Asterisk Business Edition C.3.x before C.3.7.3, disregards the alwaysauthreject option and generates different responses for invalid SIP requests depending on whether the user account exists, which allows remote attackers to enumerate account names via a series of requests. | ||||
| CVE-2011-2535 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| chan_iax2.c in the IAX2 channel driver in Asterisk Open Source 1.4.x before 1.4.41.1, 1.6.2.x before 1.6.2.18.1, and 1.8.x before 1.8.4.3, and Asterisk Business Edition C.3 before C.3.7.3, accesses a memory address contained in an option control frame, which allows remote attackers to cause a denial of service (daemon crash) or possibly have unspecified other impact via a crafted frame. | ||||
| CVE-2011-2529 | 1 Digium | 1 Asterisk | 2025-04-11 | N/A |
| chan_sip.c in the SIP channel driver in Asterisk Open Source 1.6.x before 1.6.2.18.1 and 1.8.x before 1.8.4.3 does not properly handle '\0' characters in SIP packets, which allows remote attackers to cause a denial of service (memory corruption) or possibly have unspecified other impact via a crafted packet. | ||||