| CVE |
Vendors |
Products |
Updated |
CVSS v3.1 |
| Asterisk is an open source private branch exchange and telephony toolkit. Prior to Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3, the LDAP realtime driver `res_config_ldap` may permit `LDAP filter injection` through a `SIP username`, allowing an unauthenticated attacker to make arbitrary LDAP queries using carefully crafted SIP packets and disclose information. The `res_config_ldap` module must be loaded and configured to access an LDAP server, which is not the default. This issue is fixed in Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3. |
| Asterisk is an open source private branch exchange and telephony toolkit. Prior to Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3, insufficient bounds checking in the `chan_ooh323` `Q.931` `information-element parser` permits an `out-of-bounds read` that may allow an unauthenticated attacker to crash Asterisk using carefully crafted `setup packets`. The `chan_ooh323` addon channel driver must be explicitly compiled and installed, which is not the default, and its port must be publicly reachable. This issue is fixed in Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3. |
| Asterisk is an open source private branch exchange and telephony toolkit. Prior to Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3, an unauthenticated attacker may use reflected cross-site scripting in `phone provisioning HTTP error pages` to retrieve an authorized user's `AMI session cookie` and impersonate that user when `res_phoneprov` is enabled. The Asterisk HTTP web server, Asterisk Manager Interface (AMI), and AMI access via HTTP must all be enabled; each is disabled by default. The attacker must be able to connect to the HTTP server, which binds only to `127.0.0.1` by default even when enabled, and must convince the authorized user to click a malicious link. This issue is fixed in Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3. |
| Asterisk is an open source private branch exchange and telephony toolkit. Prior to Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3, an authenticated attacker may cause a heap-use-after-free in `PJSIP TCP/SDP` handling by sending a `SIP INVITE` over a connection-oriented transport and disconnecting before Asterisk responds with the `200 OK`. So far, the crash has only been reproduced when Asterisk is built with `Address Sanitizer`, which detects the issue. `Address Sanitizer` is a development tool with significant performance overhead and, according to the maintainer, is not used in production systems. This issue is fixed in Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3. |
| Asterisk Open Source 1.0.x and 1.2.x before 1.2.29 and Business Edition A.x.x and B.x.x before B.2.5.3, when pedantic parsing (aka pedanticsipchecking) is enabled, allows remote attackers to cause a denial of service (daemon crash) via a SIP INVITE message that lacks a From header, related to invocations of the ast_uri_decode function, and improper handling of (1) an empty const string and (2) a NULL pointer. |
| The channel driver in Asterisk before 1.2.17 and 1.4.x before 1.4.2 allows remote attackers to cause a denial of service (crash) via a SIP INVITE message with an SDP containing one valid and one invalid IP address. |
| The SIP channel driver (chan_sip) in Asterisk Open Source 1.4.x before 1.4.11, AsteriskNOW before beta7, Asterisk Appliance Developer Kit 0.x before 0.8.0, and s800i (Asterisk Appliance) 1.x before 1.0.3 allows remote attackers to cause a denial of service (memory exhaustion) via a SIP dialog that causes a large number of history entries to be created. |
| Asterisk Open Source 1.2.x before 1.2.32, 1.4.x before 1.4.24.1, and 1.6.0.x before 1.6.0.8; Asterisk Business Edition A.x.x, B.x.x before B.2.5.8, C.1.x.x before C.1.10.5, and C.2.x.x before C.2.3.3; s800i 1.3.x before 1.3.0.2; and Trixbox PBX 2.6.1, when Digest authentication and authalwaysreject are enabled, generates different responses depending on whether a SIP username is valid, which allows remote attackers to enumerate valid usernames. |
| The ooh323 channel driver in Asterisk Addons 1.2.x before 1.2.9 and Asterisk-Addons 1.4.x before 1.4.7 creates a remotely accessible TCP port that is intended solely for localhost communication, and interprets some TCP application-data fields as addresses of memory to free, which allows remote attackers to cause a denial of service (daemon crash) via crafted TCP packets. |
| Buffer overflow in sethdlc.c in the Asterisk Zaptel 1.4.5.1 might allow local users to gain privileges via a long device name (interface name) in the ifr_name field. NOTE: the vendor disputes this issue, stating that the application requires root access, so privilege boundaries are not crossed |
| Array index error in the dahdi/tor2.c driver in Zaptel (aka DAHDI) 1.4.11 and earlier allows local users in the dialout group to overwrite an integer value in kernel memory by writing to /dev/zap/ctl, related to an incorrect tor2 patch for CVE-2008-5396 that uses the wrong variable in a range check against the value of lc->sync. |
| Multiple SQL injection vulnerabilities in cdr_addon_mysql in Asterisk-Addons before 1.2.8, and 1.4.x before 1.4.4, allow remote attackers to execute arbitrary SQL commands via the (1) source and (2) destination numbers, and probably (3) SIP URI, when inserting a record. |
| Unspecified vulnerability in Asterisk Open Source 1.2.x before 1.2.27, 1.4.x before 1.4.18.1 and 1.4.19-rc3; Business Edition A.x.x, B.x.x before B.2.5.1, and C.x.x before C.1.6.2; AsteriskNOW 1.0.x before 1.0.2; Appliance Developer Kit before 1.4 revision 109393; and s800i 1.0.x before 1.1.0.2; allows remote attackers to access the SIP channel driver via a crafted From header. |
| The AsteriskGUI HTTP server in Asterisk Open Source 1.4.x before 1.4.19-rc3 and 1.6.x before 1.6.0-beta6, Business Edition C.x.x before C.1.6, AsteriskNOW before 1.0.2, Appliance Developer Kit before revision 104704, and s800i 1.0.x before 1.1.0.2 generates insufficiently random manager ID values, which makes it easier for remote attackers to hijack a manager session via a series of ID guesses. |
| The Asterisk Extension Language (AEL) in pbx/pbx_ael.c in Asterisk does not properly generate extensions, which allows remote attackers to execute arbitrary extensions and have an unknown impact by specifying an invalid extension in a certain form. |
| Multiple buffer overflows in Asterisk Open Source 1.4.x before 1.4.18.1 and 1.4.19-rc3, Open Source 1.6.x before 1.6.0-beta6, Business Edition C.x.x before C.1.6.1, AsteriskNOW 1.0.x before 1.0.2, Appliance Developer Kit before 1.4 revision 109386, and s800i 1.1.x before 1.1.0.2 allow remote attackers to (1) write a zero to an arbitrary memory location via a large RTP payload number, related to the ast_rtp_unset_m_type function in main/rtp.c; or (2) write certain integers to an arbitrary memory location via a large number of RTP payloads, related to the process_sdp function in channels/chan_sip.c. |
| Format string vulnerability in Asterisk Open Source 1.6.x before 1.6.0-beta6 might allow remote attackers to execute arbitrary code via logging messages that are not properly handled by (1) the ast_verbose logging API call, or (2) the astman_append function. |
| The Skinny channel driver (chan_skinny) in Asterisk Open Source before 1.4.10, AsteriskNOW before beta7, Appliance Developer Kit before 0.7.0, and Appliance s800i before 1.0.3 allows remote authenticated users to cause a denial of service (application crash) via a CAPABILITIES_RES_MESSAGE packet with a capabilities count larger than the capabilities_res_message array population. |
| Asterisk Open Source 1.4.5 through 1.4.11, when configured to use an IMAP voicemail storage backend, allows remote attackers to cause a denial of service via an e-mail with an "invalid/corrupted" MIME body, which triggers a crash when the recipient listens to voicemail. |
| The handle_response function in chan_sip.c in Asterisk before 1.2.17 and 1.4.x before 1.4.2 allows remote attackers to cause a denial of service (crash) via a SIP Response code 0 in a SIP packet. |
