| 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 `PostgreSQL` and `TDS/MSSQL` Channel Event Log backends `cel_pgsql` and `cel_tds` may permit arbitrary SQL execution when untrusted data is set in the CEL `eventtype` column by the `CELGenUserEvent` dialplan application. The `cel_pgsql` or `cel_tds` backend must be loaded and configured, and the administrator must call `CELGenUserEvent` in the dialplan without ensuring that only valid data is passed to the application. 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, `ast_loggrabber` executes a `Python script` from the world-writable `/tmp` directory. An attacker already authorized to open a shell session on the Asterisk server may place a malicious script there and gain elevated privileges or execute code when an administrator runs `ast_loggrabber` as a user with elevated privileges. 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, `app_sms.c` copies externally controlled `SMS` lengths into `fixed in-struct buffers` without sufficient bounds checks, allowing an attacker to corrupt memory or crash Asterisk. The administrator must explicitly use the `SMS` dialplan application, which is useful when routing calls to or from `analog devices`, and must not have ensured that data passed to the application is valid. 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, a NULL pointer dereference in the `Asterisk Manager Interface (AMI)` may allow an unauthenticated attacker to crash Asterisk when `Digest Authentication` is used via `HTTP`. The Asterisk `HTTP` web server, 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. 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. From 21.0.0-pre1 until Asterisk 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 22.8-cert3, the `chan_ooh323` `Q.931 party-number parser` lacks the `CVE-2022-37325` fix because that fix was never forward-ported from Asterisk 20 to later versions. Malformed Q.931 elements in an `OOH323 request` can therefore crash Asterisk. This issue affects Asterisk versions greater than 20. 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 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 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 buffer overflow through an out-of-bounds write in the `Codec2` decoder caused by a `floor/ceil sample-count mismatch`. The `codec_codec2` module must be loaded and `Codec2` must be configured on the attacked endpoint. The attacker must pass SIP authentication by username/password or `source IP address matching`, establish a call using `Codec2`, and send carefully crafted `Codec2` audio packets. 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 `sscanf` in `res_pjsip_pubsub` when parsing the `MWI NOTIFY` `Message-Account` field may allow an unauthenticated attacker to send a carefully crafted packet that causes a stack buffer overflow and corrupts the underlying transport, permanently disabling all SIP functionality. An Asterisk restart is required to restore full operation. 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 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. |
| 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. |
