Vulnerabilities (CVE)

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Total 115 CVE
CVE Vendors Products Updated CVSS v2 CVSS v3
CVE-2011-1175 1 Digium 1 Asterisk 2017-08-16 5.0 MEDIUM N/A
tcptls.c in the TCP/TLS server in Asterisk Open Source 1.6.1.x before 1.6.1.23, 1.6.2.x before 1.6.2.17.1, and 1.8.x before 1.8.3.1 allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) by establishing many short TCP sessions to services that use a certain TLS API.
CVE-2011-1174 1 Digium 1 Asterisk 2017-08-16 5.0 MEDIUM N/A
manager.c in Asterisk Open Source 1.6.1.x before 1.6.1.24, 1.6.2.x before 1.6.2.17.2, and 1.8.x before 1.8.3.2 allows remote attackers to cause a denial of service (CPU and memory consumption) via a series of manager sessions involving invalid data.
CVE-2009-2651 1 Digium 1 Asterisk 2017-08-16 5.0 MEDIUM N/A
main/rtp.c in Asterisk Open Source 1.6.1 before 1.6.1.2 allows remote attackers to cause a denial of service (crash) via an RTP text frame without a certain delimiter, which triggers a NULL pointer dereference and the subsequent calculation of an invalid pointer.
CVE-2007-1306 1 Digium 1 Asterisk 2017-07-28 7.8 HIGH N/A
Asterisk 1.4 before 1.4.1 and 1.2 before 1.2.16 allows remote attackers to cause a denial of service (crash) by sending a Session Initiation Protocol (SIP) packet without a URI and SIP-version header, which results in a NULL pointer dereference.
CVE-2016-9937 1 Digium 1 Asterisk 2017-07-26 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-2016-9938 1 Digium 2 Asterisk, Certified Asterisk 2017-07-26 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-2005-2081 1 Digium 1 Asterisk 2017-07-10 5.0 MEDIUM N/A
Stack-based buffer overflow in the function that parses commands in Asterisk 1.0.7, when the 'write = command' option is enabled, allows remote attackers to execute arbitrary code via a command that has two double quotes followed by a tab character.
CVE-2016-7551 2 Debian, Digium 3 Debian Linux, Asterisk, Certified Asterisk 2017-04-24 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 2017-04-17 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-2014-8414 1 Digium 2 Asterisk, Certified Asterisk 2014-12-30 5.0 MEDIUM 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-2014-6609 1 Digium 1 Asterisk 2014-11-26 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-2014-6610 1 Digium 2 Asterisk, Certified Asterisk 2014-11-26 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-2014-2289 1 Digium 1 Asterisk 2014-04-21 3.5 LOW 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 2014-04-21 4.3 MEDIUM 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 2014-04-21 3.5 LOW 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-2014-2286 2 Digium, Fedoraproject 3 Asterisk, Certified Asterisk, Fedora 2014-04-21 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-2012-3863 1 Digium 4 Asterisk, Asterisk Business Edition, Asteriske and 1 more 2013-10-10 4.0 MEDIUM N/A
channels/chan_sip.c in Asterisk Open Source 1.8.x before 1.8.13.1 and 10.x before 10.5.2, Asterisk Business Edition C.3.x before C.3.7.5, Certified Asterisk 1.8.11-certx before 1.8.11-cert4, and Asterisk Digiumphones 10.x.x-digiumphones before 10.5.2-digiumphones does not properly handle a provisional response to a SIP reINVITE request, which allows remote authenticated users to cause a denial of service (RTP port exhaustion) via sessions that lack final responses.
CVE-2013-5642 1 Digium 3 Asterisk, Asterisk Digiumphones, Certified Asterisk 2013-09-11 5.0 MEDIUM N/A
The SIP channel driver (channels/chan_sip.c) in Asterisk Open Source 1.8.x before 1.8.23.1, 10.x before 10.12.3, and 11.x before 11.5.1; Certified Asterisk 1.8.15 before 1.8.15-cert3 and 11.2 before 11.2-cert2; and Asterisk Digiumphones 10.x-digiumphones before 10.12.3-digiumphones allows remote attackers to cause a denial of service (NULL pointer dereference, segmentation fault, and daemon crash) via an invalid SDP that defines a media description before the connection description in a SIP request.
CVE-2013-5641 1 Digium 2 Asterisk, Certified Asterisk 2013-09-11 5.0 MEDIUM N/A
The SIP channel driver (channels/chan_sip.c) in Asterisk Open Source 1.8.17.x through 1.8.22.x, 1.8.23.x before 1.8.23.1, and 11.x before 11.5.1 and Certified Asterisk 1.8.15 before 1.8.15-cert3 and 11.2 before 11.2-cert2 allows remote attackers to cause a denial of service (NULL pointer dereference, segmentation fault, and daemon crash) via an ACK with SDP to a previously terminated channel. NOTE: some of these details are obtained from third party information.
CVE-2012-4737 1 Digium 2 Asterisk, Certified Asterisk 2013-04-18 6.0 MEDIUM N/A
channels/chan_iax2.c in Asterisk Open Source 1.8.x before 1.8.15.1 and 10.x before 10.7.1, Certified Asterisk 1.8.11 before 1.8.11-cert7, Asterisk Digiumphones 10.x.x-digiumphones before 10.7.1-digiumphones, and Asterisk Business Edition C.3.x before C.3.7.6 does not enforce ACL rules during certain uses of peer credentials, which allows remote authenticated users to bypass intended outbound-call restrictions by leveraging the availability of these credentials.