| 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, an out-of-bounds write in `chan_unistim` `DIALPAGE` digit handling can overflow the `phone_number` field and crash Asterisk when an attacker sends a carefully crafted incoming request packet. The `chan_unistim` module must be loaded and configured with a valid `UNISTIM UDP port` that the attacker can reach. The attacker must register as a configured `phone MAC`, or the deployment must have an equivalent `production/autoprovisioning` path that allows the client to reach the main page. 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 crash Asterisk through a buffer over-read in the `PJSIP MWI body parser` by sending a carefully crafted `SIP NOTIFY` request. The attacker must pass SIP authentication using either a username/password or `source IP address matching`. 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 stack buffer overflow in `res_xmpp` `XMPP namespace prefix` handling using carefully crafted packets. The `res_xmpp` module must be loaded, an XMPP connection must be configured in `xmpp.conf`, and the attacker must have an XMPP account on the same or a federated server. 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 Asterisk REST Interface (ARI) attacker with read-only privileges may use `setChannelVar` to execute write dialplan functions, including `FILE()` writes, even when `live_dangerously` in `asterisk.conf` is set to no. The Asterisk HTTP web server must be enabled and reachable, and the attacker must already have valid `read-only ARI credentials`. HTTP is disabled by default and binds only to 127.0.0.1 by default even when enabled. 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. In affected deployments, an authenticated attacker may cause a `T.140 RED` out-of-bounds write during `generation accumulation` using carefully crafted packets, potentially allowing remote code execution or causing Asterisk to crash. The `chan_sip` channel driver must be installed and configured, and its `textsupport` option must be enabled; `textsupport` defaults to no. `The chan_sip driver was deprecated in Asterisk 20 and completely removed in all later versions.` Repository comparisons verify patch containment in Asterisk 20.20.1, 21.12.3, 22.10.1 and 23.4.1, and Certified Asterisk 20.7-cert11 and 22.8-cert3. The affected release-track scope remains unresolved: structured metadata includes later release tracks even though the advisory says the required `chan_sip` driver was removed there. |
| Asterisk is an open source private branch exchange and telephony toolkit. In affected deployments, an authenticated attacker may cause a `T.140 RED` heap buffer overflow using carefully crafted packets, potentially allowing remote code execution or causing Asterisk to crash. The `chan_sip` channel driver must be installed and configured, and its `textsupport` option must be enabled; `textsupport` defaults to no. `The chan_sip driver was deprecated in Asterisk 20 and completely removed in all later versions.` Repository comparisons verify patch containment in Asterisk 20.20.1, 21.12.3, 22.10.1 and 23.4.1, and Certified Asterisk 20.7-cert11 and 22.8-cert3. The affected release-track scope remains unresolved: structured metadata includes later release tracks even though the advisory says the required `chan_sip` driver was removed there. |
| 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, a stack buffer overflow in the H.323 `ooTrace()` trace-logging function can occur when unbounded `vsprintf` writes into a fixed `2048-byte buffer`. A remote attacker may send a carefully crafted packet that causes the Asterisk process to crash. 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 unbounded `memcpy` of `OGG/Speex` packet data in `format_ogg_speex` can cause a heap buffer overflow and crash Asterisk when reading a file containing corrupted Speex audio. The attacker must create the file and make it available to Asterisk either through sufficiently elevated privileges or by convincing an Asterisk administrator to make it available. Asterisk must also be configured to play that file. 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, 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 is an open source private branch exchange and telephony toolkit. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, user supplied/control values for Cookies and any GET variable query Parameter are directly interpolated into the HTML of the page using ast_str_append. The endpoint at GET /httpstatus is the potential vulnerable endpoint relating to asterisk/main /http.c. This issue has been patched in versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2. |
| Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, the ast_xml_open() function in xml.c parses XML documents using libxml with unsafe parsing options that enable entity expansion and XInclude processing. Specifically, it invokes xmlReadFile() with the XML_PARSE_NOENT flag and later processes XIncludes via xmlXIncludeProcess().If any untrusted or user-supplied XML file is passed to this function, it can allow an attacker to trigger XML External Entity (XXE) or XInclude-based local file disclosure, potentially exposing sensitive files from the host system. This can also be triggered in other cases in which the user is able to supply input in xml format that triggers the asterisk process to parse it. This issue has been patched in versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2. |
| Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, when ast_coredumper writes its gdb init and output files to a directory that is world-writable (for example /tmp), an attacker with write permission(which is all users on a linux system) to that directory can cause root to execute arbitrary commands or overwrite arbitrary files by controlling the gdb init file and output paths. This issue has been patched in versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2. |
| Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, the asterisk/contrib/scripts/ast_coredumper runs as root, as noted by the NOTES tag on line 689 of the ast_coredumper file. The script will source the contents of /etc/asterisk/ast_debug_tools.conf, which resides in a folder that is writeable by the asterisk user:group. Due to the /etc/asterisk/ast_debug_tools.conf file following bash semantics and it being loaded; an attacker with write permissions may add or modify the file such that when the root ast_coredumper is run; it would source and thereby execute arbitrary bash code found in the /etc/asterisk/ast_debug_tools.conf. This issue has been patched in versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2. |
| A local privilege escalation vulnerability exists in the safe_asterisk script included with the Asterisk toolkit package. When Asterisk is started via this script (common in SysV init or FreePBX environments), it sources all .sh files located in /etc/asterisk/startup.d/ as root, without validating ownership or permissions.
Non-root users with legitimate write access to /etc/asterisk can exploit this behaviour by placing malicious scripts in the startup.d directory, which will then execute with root privileges upon service restart. |
| PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In versions up to and including 2.11.1 when in a dialog set (or forking) scenario, a hash key shared by multiple UAC dialogs can potentially be prematurely freed when one of the dialogs is destroyed . The issue may cause a dialog set to be registered in the hash table multiple times (with different hash keys) leading to undefined behavior such as dialog list collision which eventually leading to endless loop. A patch is available in commit db3235953baa56d2fb0e276ca510fefca751643f which will be included in the next release. There are no known workarounds for this issue. |
| PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In versions 2.11.1 and prior, parsing an incoming SIP message that contains a malformed multipart can potentially cause out-of-bound read access. This issue affects all PJSIP users that accept SIP multipart. The patch is available as commit in the `master` branch. There are no known workarounds. |
| PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In affected versions if the incoming STUN message contains an ERROR-CODE attribute, the header length is not checked before performing a subtraction operation, potentially resulting in an integer underflow scenario. This issue affects all users that use STUN. A malicious actor located within the victim’s network may forge and send a specially crafted UDP (STUN) message that could remotely execute arbitrary code on the victim’s machine. Users are advised to upgrade as soon as possible. There are no known workarounds. |
