Filtered by vendor Digium
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Total
119 CVE
| CVE | Vendors | Products | Updated | CVSS v2 | CVSS v3 |
|---|---|---|---|---|---|
| CVE-2017-14100 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-02-28 | 7.5 HIGH | 9.8 CRITICAL |
| In Asterisk 11.x before 11.25.2, 13.x before 13.17.1, and 14.x before 14.6.1 and Certified Asterisk 11.x before 11.6-cert17 and 13.x before 13.13-cert5, unauthorized command execution is possible. The app_minivm module has an "externnotify" program configuration option that is executed by the MinivmNotify dialplan application. The application uses the caller-id name and number as part of a built string passed to the OS shell for interpretation and execution. Since the caller-id name and number can come from an untrusted source, a crafted caller-id name or number allows an arbitrary shell command injection. | |||||
| CVE-2017-14603 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-02-28 | 5.0 MEDIUM | 7.5 HIGH |
| In Asterisk 11.x before 11.25.3, 13.x before 13.17.2, and 14.x before 14.6.2 and Certified Asterisk 11.x before 11.6-cert18 and 13.x before 13.13-cert6, insufficient RTCP packet validation could allow reading stale buffer contents and when combined with the "nat" and "symmetric_rtp" options allow redirecting where Asterisk sends the next RTCP report. | |||||
| CVE-2017-14099 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-02-28 | 5.0 MEDIUM | 7.5 HIGH |
| In res/res_rtp_asterisk.c in Asterisk 11.x before 11.25.2, 13.x before 13.17.1, and 14.x before 14.6.1 and Certified Asterisk 11.x before 11.6-cert17 and 13.x before 13.13-cert5, unauthorized data disclosure (media takeover in the RTP stack) is possible with careful timing by an attacker. The "strictrtp" option in rtp.conf enables a feature of the RTP stack that learns the source address of media for a session and drops any packets that do not originate from the expected address. This option is enabled by default in Asterisk 11 and above. The "nat" and "rtp_symmetric" options (for chan_sip and chan_pjsip, respectively) enable symmetric RTP support in the RTP stack. This uses the source address of incoming media as the target address of any sent media. This option is not enabled by default, but is commonly enabled to handle devices behind NAT. A change was made to the strict RTP support in the RTP stack to better tolerate late media when a reinvite occurs. When combined with the symmetric RTP support, this introduced an avenue where media could be hijacked. Instead of only learning a new address when expected, the new code allowed a new source address to be learned at all times. If a flood of RTP traffic was received, the strict RTP support would allow the new address to provide media, and (with symmetric RTP enabled) outgoing traffic would be sent to this new address, allowing the media to be hijacked. Provided the attacker continued to send traffic, they would continue to receive traffic as well. | |||||
| CVE-2017-16672 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-02-28 | 4.3 MEDIUM | 5.9 MEDIUM |
| An issue was discovered in Asterisk Open Source 13 before 13.18.1, 14 before 14.7.1, and 15 before 15.1.1 and Certified Asterisk 13.13 before 13.13-cert7. A memory leak occurs when an Asterisk pjsip session object is created and that call gets rejected before the session itself is fully established. When this happens the session object never gets destroyed. Eventually Asterisk can run out of memory and crash. | |||||
| CVE-2017-14001 | 1 Digium | 1 Asterisk Gui | 2024-02-28 | 9.0 HIGH | 8.8 HIGH |
| An Improper Neutralization of Special Elements used in an OS Command issue was discovered in Digium Asterisk GUI 2.1.0 and prior. An OS command injection vulnerability has been identified that may allow the execution of arbitrary code on the system through the inclusion of OS commands in the URL request of the program. | |||||
| CVE-2016-7551 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-02-28 | 5.0 MEDIUM | 7.5 HIGH |
| 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-2017-7617 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-02-28 | 6.5 MEDIUM | 8.8 HIGH |
| Remote code execution can occur in Asterisk Open Source 13.x before 13.14.1 and 14.x before 14.3.1 and Certified Asterisk 13.13 before 13.13-cert3 because of a buffer overflow in a CDR user field, related to X-ClientCode in chan_sip, the CDR dialplan function, and the AMI Monitor action. | |||||
| CVE-2016-9938 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-02-28 | 5.0 MEDIUM | 5.3 MEDIUM |
| 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-2016-9937 | 1 Digium | 1 Asterisk | 2024-02-28 | 5.0 MEDIUM | 7.5 HIGH |
| 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-2016-2232 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-02-28 | 4.0 MEDIUM | 6.5 MEDIUM |
| 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 | 2024-02-28 | 7.1 HIGH | 5.9 MEDIUM |
| 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 | 2024-02-28 | 5.0 MEDIUM | 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-2014-6610 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-02-28 | 4.0 MEDIUM | 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 | 2024-02-28 | 4.0 MEDIUM | 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-2014-8413 | 1 Digium | 1 Asterisk | 2024-02-28 | 7.5 HIGH | 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-2014-2286 | 2 Digium, Fedoraproject | 3 Asterisk, Certified Asterisk, Fedora | 2024-02-28 | 7.5 HIGH | 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-2015-3008 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-02-28 | 4.3 MEDIUM | 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-2014-4048 | 1 Digium | 1 Asterisk | 2024-02-28 | 4.3 MEDIUM | 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-2014-8412 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-02-28 | 5.0 MEDIUM | 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-2014-8417 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-02-28 | 6.5 MEDIUM | 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. | |||||