Filtered by vendor Digium
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115 CVE
| CVE | Vendors | Products | Updated | CVSS v2 | CVSS v3 |
|---|---|---|---|---|---|
| CVE-2007-5358 | 1 Digium | 1 Asterisk | 2018-10-15 | 6.8 MEDIUM | N/A |
| Multiple buffer overflows in the voicemail functionality in Asterisk 1.4.x before 1.4.13, when using IMAP storage, might allow (1) remote attackers to execute arbitrary code via a long combination of Content-type and Content-description headers, or (2) local users to execute arbitrary code via a long combination of astspooldir, voicemail context, and voicemail mailbox fields. NOTE: vector 2 requires write access to Asterisk configuration files. | |||||
| CVE-2007-4103 | 1 Digium | 3 Asterisk, Asterisk Appliance Developer Kit, Asterisknow Pre-release | 2018-10-15 | 7.8 HIGH | N/A |
| The IAX2 channel driver (chan_iax2) in Asterisk Open 1.2.x before 1.2.23, 1.4.x before 1.4.9, and Asterisk Appliance Developer Kit before 0.6.0, when configured to allow unauthenticated calls, allows remote attackers to cause a denial of service (resource exhaustion) via a flood of calls that do not complete a 3-way handshake, which causes an ast_channel to be allocated but not released. | |||||
| CVE-2010-1224 | 1 Digium | 1 Asterisk | 2018-10-10 | 4.3 MEDIUM | N/A |
| main/acl.c in Asterisk Open Source 1.6.0.x before 1.6.0.25, 1.6.1.x before 1.6.1.17, and 1.6.2.x before 1.6.2.5 does not properly enforce remote host access controls when CIDR notation "/0" is used in permit= and deny= configuration rules, which causes an improper arithmetic shift and might allow remote attackers to bypass ACL rules and access services from unauthorized hosts. | |||||
| CVE-2010-0685 | 1 Digium | 1 Asterisk | 2018-10-10 | 5.0 MEDIUM | N/A |
| The design of the dialplan functionality in Asterisk Open Source 1.2.x, 1.4.x, and 1.6.x; and Asterisk Business Edition B.x.x and C.x.x, when using the ${EXTEN} channel variable and wildcard pattern matches, allows context-dependent attackers to inject strings into the dialplan using metacharacters that are injected when the variable is expanded, as demonstrated using the Dial application to process a crafted SIP INVITE message that adds an unintended outgoing channel leg. NOTE: it could be argued that this is not a vulnerability in Asterisk, but a class of vulnerabilities that can occur in any program that uses this feature without the associated filtering functionality that is already available. | |||||
| CVE-2009-4055 | 1 Digium | 2 Asterisk, S800i | 2018-10-10 | 5.0 MEDIUM | N/A |
| rtp.c in Asterisk Open Source 1.2.x before 1.2.37, 1.4.x before 1.4.27.1, 1.6.0.x before 1.6.0.19, and 1.6.1.x before 1.6.1.11; Business Edition B.x.x before B.2.5.13, C.2.x.x before C.2.4.6, and C.3.x.x before C.3.2.3; and s800i 1.3.x before 1.3.0.6 allows remote attackers to cause a denial of service (daemon crash) via an RTP comfort noise payload with a long data length. | |||||
| CVE-2009-0871 | 1 Digium | 1 Asterisk | 2018-10-10 | 3.5 LOW | N/A |
| The SIP channel driver in Asterisk Open Source 1.4.22, 1.4.23, and 1.4.23.1; 1.6.0 before 1.6.0.6; 1.6.1 before 1.6.1.0-rc2; and Asterisk Business Edition C.2.3, with the pedantic option enabled, allows remote authenticated users to cause a denial of service (crash) via a SIP INVITE request without any headers, which triggers a NULL pointer dereference in the (1) sip_uri_headers_cmp and (2) sip_uri_params_cmp functions. | |||||
| CVE-2015-3008 | 1 Digium | 2 Asterisk, Certified Asterisk | 2018-10-09 | 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-2015-2690 | 1 Digium | 1 Addons Module | 2018-10-09 | 4.3 MEDIUM | 6.1 MEDIUM |
| 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-2014-9374 | 1 Digium | 2 Asterisk, Certified Asterisk | 2018-10-09 | 5.0 MEDIUM | 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-2015-1558 | 1 Digium | 1 Asterisk | 2018-10-09 | 3.5 LOW | 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-2014-4045 | 1 Digium | 1 Asterisk | 2018-10-09 | 4.3 MEDIUM | 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 | 2018-10-09 | 5.0 MEDIUM | 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 | 2018-10-09 | 6.5 MEDIUM | 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-2014-4048 | 1 Digium | 1 Asterisk | 2018-10-09 | 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-2011-2216 | 1 Digium | 1 Asterisk | 2018-10-09 | 5.0 MEDIUM | N/A |
| reqresp_parser.c in the SIP channel driver in Asterisk Open Source 1.8.x before 1.8.4.2 does not initialize certain strings, which allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) via a malformed Contact header. | |||||
| CVE-2018-7285 | 1 Digium | 1 Asterisk | 2018-03-21 | 5.0 MEDIUM | 7.5 HIGH |
| A NULL pointer access issue was discovered in Asterisk 15.x through 15.2.1. The RTP support in Asterisk maintains its own registry of dynamic codecs and desired payload numbers. While an SDP negotiation may result in a codec using a different payload number, these desired ones are still stored internally. When an RTP packet was received, this registry would be consulted if the payload number was not found in the negotiated SDP. This registry was incorrectly consulted for all packets, even those which are dynamic. If the payload number resulted in a codec of a different type than the RTP stream (for example, the payload number resulted in a video codec but the stream carried audio), a crash could occur if no stream of that type had been negotiated. This was due to the code incorrectly assuming that a stream of that type would always exist. | |||||
| CVE-2017-17664 | 1 Digium | 2 Asterisk, Certified Asterisk | 2018-01-02 | 4.3 MEDIUM | 5.9 MEDIUM |
| 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. | |||||
| CVE-2012-2947 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2017-11-13 | 2.6 LOW | N/A |
| chan_iax2.c in the IAX2 channel driver in Certified Asterisk 1.8.11-cert before 1.8.11-cert2 and Asterisk Open Source 1.8.x before 1.8.12.1 and 10.x before 10.4.1, when a certain mohinterpret setting is enabled, allows remote attackers to cause a denial of service (daemon crash) by placing a call on hold. | |||||
| CVE-2017-14603 | 1 Digium | 2 Asterisk, Certified Asterisk | 2017-11-05 | 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-2017-9372 | 1 Digium | 2 Certified Asterisk, Open Source | 2017-11-05 | 5.0 MEDIUM | 7.5 HIGH |
| PJSIP, as used in Asterisk Open Source 13.x before 13.15.1 and 14.x before 14.4.1, Certified Asterisk 13.13 before 13.13-cert4, and other products, allows remote attackers to cause a denial of service (buffer overflow and application crash) via a SIP packet with a crafted CSeq header in conjunction with a Via header that lacks a branch parameter. | |||||
| CVE-2017-9359 | 1 Digium | 2 Certified Asterisk, Open Source | 2017-11-05 | 5.0 MEDIUM | 7.5 HIGH |
| The multi-part body parser in PJSIP, as used in Asterisk Open Source 13.x before 13.15.1 and 14.x before 14.4.1, Certified Asterisk 13.13 before 13.13-cert4, and other products, allows remote attackers to cause a denial of service (out-of-bounds read and application crash) via a crafted packet. | |||||
| CVE-2016-2232 | 1 Digium | 2 Asterisk, Certified Asterisk | 2017-11-04 | 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-2017-14099 | 1 Digium | 2 Asterisk, Certified Asterisk | 2017-11-04 | 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-2016-2316 | 2 Digium, Fedoraproject | 3 Asterisk, Certified Asterisk, Fedora | 2017-11-04 | 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-2017-14098 | 1 Digium | 1 Asterisk | 2017-09-14 | 5.0 MEDIUM | 7.5 HIGH |
| In the pjsip channel driver (res_pjsip) in Asterisk 13.x before 13.17.1 and 14.x before 14.6.1, a carefully crafted tel URI in a From, To, or Contact header could cause Asterisk to crash. | |||||
| CVE-2013-7100 | 1 Digium | 3 Asterisk, Asterisk Digiumphones, Certified Asterisk | 2017-08-29 | 5.0 MEDIUM | N/A |
| Buffer overflow in the unpacksms16 function in apps/app_sms.c in Asterisk Open Source 1.8.x before 1.8.24.1, 10.x before 10.12.4, and 11.x before 11.6.1; Asterisk with Digiumphones 10.x-digiumphones before 10.12.4-digiumphones; and Certified Asterisk 1.8.x before 1.8.15-cert4 and 11.x before 11.2-cert3 allows remote attackers to cause a denial of service (daemon crash) via a 16-bit SMS message with an odd number of bytes, which triggers an infinite loop. | |||||
| CVE-2012-1184 | 1 Digium | 1 Asterisk | 2017-08-29 | 7.5 HIGH | N/A |
| Stack-based buffer overflow in the ast_parse_digest function in main/utils.c in Asterisk 1.8.x before 1.8.10.1 and 10.x before 10.2.1 allows remote attackers to cause a denial of service (crash) or possibly execute arbitrary code via a long string in an HTTP Digest Authentication header. | |||||
| CVE-2011-2529 | 1 Digium | 1 Asterisk | 2017-08-29 | 5.0 MEDIUM | 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. | |||||
| CVE-2011-2666 | 1 Digium | 1 Asterisk | 2017-08-29 | 5.0 MEDIUM | 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-2011-2535 | 1 Digium | 1 Asterisk | 2017-08-29 | 5.0 MEDIUM | 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-2011-1174 | 1 Digium | 1 Asterisk | 2017-08-17 | 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-2011-1175 | 1 Digium | 1 Asterisk | 2017-08-17 | 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-2009-2651 | 1 Digium | 1 Asterisk | 2017-08-17 | 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 | 2017-07-29 | 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 | 2017-07-27 | 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 | 2017-07-27 | 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 | 2017-07-11 | 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 | 2017-04-25 | 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 | 2013-09-12 | 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 | 2013-09-12 | 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 | 2013-04-19 | 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. | |||||