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Vuln ID | Summary | CVSS Severity |
---|---|---|
CVE-2017-16671 |
A Buffer Overflow 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. No size checking is done when setting the user field for Party B on a CDR. Thus, it is possible for someone to use an arbitrarily large string and write past the end of the user field storage buffer. NOTE: this is different from CVE-2017-7617, which was only about the Party A buffer. Published: November 08, 2017; 7:29:00 PM -0500 |
V4.0:(not available) V3.0: 8.8 HIGH V2.0: 6.5 MEDIUM |
CVE-2016-7551 |
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). Published: April 17, 2017; 12:59:00 PM -0400 |
V4.0:(not available) V3.0: 7.5 HIGH V2.0: 5.0 MEDIUM |
CVE-2016-9938 |
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. Published: December 12, 2016; 4:59:01 PM -0500 |
V4.0:(not available) V3.0: 5.3 MEDIUM V2.0: 5.0 MEDIUM |
CVE-2015-1558 |
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. Published: February 09, 2015; 6:59:00 AM -0500 |
V4.0:(not available) V3.x:(not available) V2.0: 3.5 LOW |
CVE-2014-9374 |
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. Published: December 12, 2014; 10:59:14 AM -0500 |
V4.0:(not available) V3.x:(not available) V2.0: 5.0 MEDIUM |
CVE-2014-8417 |
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. Published: November 24, 2014; 10:59:09 AM -0500 |
V4.0:(not available) V3.x:(not available) V2.0: 6.5 MEDIUM |
CVE-2014-8416 |
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. Published: November 24, 2014; 10:59:08 AM -0500 |
V4.0:(not available) V3.x:(not available) V2.0: 5.0 MEDIUM |
CVE-2014-8415 |
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. Published: November 24, 2014; 10:59:07 AM -0500 |
V4.0:(not available) V3.x:(not available) V2.0: 5.0 MEDIUM |
CVE-2014-8413 |
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. Published: November 24, 2014; 10:59:05 AM -0500 |
V4.0:(not available) V3.x:(not available) V2.0: 7.5 HIGH |
CVE-2014-8412 |
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. Published: November 24, 2014; 10:59:04 AM -0500 |
V4.0:(not available) V3.x:(not available) V2.0: 5.0 MEDIUM |