Asterisk

Asterisk is an open source private branch exchange and telephony toolkit. In Asterisk prior to versions 18.20.1, 20.5.1, and 21.0.1; as well as certified-asterisk prior to 18.9-cert6; Asterisk is susceptible to a DoS due to a race condition in the hello handshake phase of the DTLS protocol when handling DTLS-SRTP for media setup. This attack can be done continuously, thus denying new DTLS-SRTP encrypted calls during the attack. Abuse of this vulnerability may lead to a massive Denial of Service on vulnerable Asterisk servers for calls that rely on DTLS-SRTP. Commit d7d7764cb07c8a1872804321302ef93bf62cba05 contains a fix, which is part of versions 18.20.1, 20.5.1, 21.0.1, amd 18.9-cert6.

Published: 2023-12-14T19:47:46.306Z
Updated: 2025-02-13T17:18:55.224Z

Asterisk is an open source private branch exchange and telephony toolkit. In Asterisk prior to versions 18.20.1, 20.5.1, and 21.0.1, as well as certified-asterisk prior to 18.9-cert6, it is possible to read any arbitrary file even when the `live_dangerously` is not enabled. This allows arbitrary files to be read. Asterisk versions 18.20.1, 20.5.1, and 21.0.1, as well as certified-asterisk prior to 18.9-cert6, contain a fix for this issue.

Published: 2023-12-14T19:40:46.157Z
Updated: 2025-02-13T17:18:40.277Z

Asterisk is an open source private branch exchange and telephony toolkit. In Asterisk versions 18.20.0 and prior, 20.5.0 and prior, and 21.0.0; as well as ceritifed-asterisk 18.9-cert5 and prior, the 'update' functionality of the PJSIP_HEADER dialplan function can exceed the available buffer space for storing the new value of a header. By doing so this can overwrite memory or cause a crash. This is not externally exploitable, unless dialplan is explicitly written to update a header based on data from an outside source. If the 'update' functionality is not used the vulnerability does not occur. A patch is available at commit a1ca0268254374b515fa5992f01340f7717113fa.

Published: 2023-12-14T19:43:30.945Z
Updated: 2025-02-13T17:01:26.636Z

An issue was discovered in Asterisk through 19.x and Certified Asterisk through 16.8-cert13. The func_odbc module provides possibly inadequate escaping functionality for backslash characters in SQL queries, resulting in user-provided data creating a broken SQL query or possibly a SQL injection. This is fixed in 16.25.2, 18.11.2, and 19.3.2, and 16.8-cert14.

Published: 2022-04-15T00:00:00.000Z
Updated: 2024-08-03T05:11:43.391Z

An SSRF issue was discovered in Asterisk through 19.x. When using STIR/SHAKEN, it's possible to send arbitrary requests (such as GET) to interfaces such as localhost by using the Identity header. This is fixed in 16.25.2, 18.11.2, and 19.3.2.

Published: 2022-04-15T00:00:00.000Z
Updated: 2024-08-03T05:03:32.912Z

An issue was discovered in Asterisk through 19.x. When using STIR/SHAKEN, it is possible to download files that are not certificates. These files could be much larger than what one would expect to download, leading to Resource Exhaustion. This is fixed in 16.25.2, 18.11.2, and 19.3.2.

Published: 2022-04-15T00:00:00.000Z
Updated: 2024-08-03T05:03:32.984Z

res_pjsip_t38 in Sangoma Asterisk 16.x before 16.16.2, 17.x before 17.9.3, and 18.x before 18.2.2, and Certified Asterisk before 16.8-cert7, allows an attacker to trigger a crash by sending an m=image line and zero port in a response to a T.38 re-invite initiated by Asterisk. This is a re-occurrence of the CVE-2019-15297 symptoms but not for exactly the same reason. The crash occurs because there is an append operation relative to the active topology, but this should instead be a replace operation.

Published: 2022-08-30T00:00:00.000Z
Updated: 2024-08-04T05:17:42.455Z

An issue was discovered in Sangoma Asterisk 13.x before 13.38.3, 16.x before 16.19.1, 17.x before 17.9.4, and 18.x before 18.5.1, and Certified Asterisk before 16.8-cert10. If the IAX2 channel driver receives a packet that contains an unsupported media format, a crash can occur.

Published: 2021-07-27T05:19:34.000Z
Updated: 2024-08-03T23:25:30.855Z

An issue was discovered in res_pjsip_session.c in Digium Asterisk through 13.38.1; 14.x, 15.x, and 16.x through 16.16.0; 17.x through 17.9.1; and 18.x through 18.2.0, and Certified Asterisk through 16.8-cert5. An SDP negotiation vulnerability in PJSIP allows a remote server to potentially crash Asterisk by sending specific SIP responses that cause an SDP negotiation failure.

Published: 2021-02-18T19:50:04.000Z
Updated: 2024-08-03T20:33:41.368Z

An issue was discovered in Sangoma Asterisk 16.x before 16.16.1, 17.x before 17.9.2, and 18.x before 18.2.1 and Certified Asterisk before 16.8-cert6. When re-negotiating for T.38, if the initial remote response was delayed just enough, Asterisk would send both audio and T.38 in the SDP. If this happened, and the remote responded with a declined T.38 stream, then Asterisk would crash.

Published: 2021-02-18T19:39:46.000Z
Updated: 2024-08-03T20:33:40.586Z

A stack-based buffer overflow in res_rtp_asterisk.c in Sangoma Asterisk before 16.16.1, 17.x before 17.9.2, and 18.x before 18.2.1 and Certified Asterisk before 16.8-cert6 allows an authenticated WebRTC client to cause an Asterisk crash by sending multiple hold/unhold requests in quick succession. This is caused by a signedness comparison mismatch.

Published: 2021-02-19T19:30:30.000Z
Updated: 2024-08-03T20:33:40.660Z

Incorrect access controls in res_srtp.c in Sangoma Asterisk 13.38.1, 16.16.0, 17.9.1, and 18.2.0 and Certified Asterisk 16.8-cert5 allow a remote unauthenticated attacker to prematurely terminate secure calls by replaying SRTP packets.

Published: 2021-02-18T20:10:20.000Z
Updated: 2024-08-03T20:33:40.803Z

A buffer overflow in res_pjsip_diversion.c in Sangoma Asterisk versions 13.38.1, 16.15.1, 17.9.1, and 18.1.1 allows remote attacker to crash Asterisk by deliberately misusing SIP 181 responses.

Published: 2021-02-18T19:57:57.000Z
Updated: 2024-08-04T17:09:15.196Z

An issue was discovered in res_pjsip_diversion.c in Sangoma Asterisk before 13.38.0, 14.x through 16.x before 16.15.0, 17.x before 17.9.0, and 18.x before 18.1.0. A crash can occur when a SIP message is received with a History-Info header that contains a tel-uri, or when a SIP 181 response is received that contains a tel-uri in the Diversion header.

Published: 2021-01-29T07:22:40.000Z
Updated: 2024-08-04T17:09:14.847Z

An Integer Signedness issue (for a return code) in the res_pjsip_sdp_rtp module in Digium Asterisk versions 15.7.1 and earlier and 16.1.1 and earlier allows remote authenticated users to crash Asterisk via a specially crafted SDP protocol violation.

Published: 2019-03-28T16:18:45.000Z
Updated: 2024-08-04T20:46:45.910Z

An issue was discovered in res_pjsip_t38.c in Sangoma Asterisk through 13.x and Certified Asterisk through 13.21-x. If it receives a re-invite initiating T.38 faxing and has a port of 0 and no c line in the SDP, a NULL pointer dereference and crash will occur. This is different from CVE-2019-18940.

Published: 2019-11-22T16:59:19.000Z
Updated: 2024-08-05T02:02:39.983Z

An issue was discovered in channels/chan_sip.c in Sangoma Asterisk 13.x before 13.29.2, 16.x before 16.6.2, and 17.x before 17.0.1, and Certified Asterisk 13.21 before cert5. A SIP request can be sent to Asterisk that can change a SIP peer's IP address. A REGISTER does not need to occur, and calls can be hijacked as a result. The only thing that needs to be known is the peer's name; authentication details such as passwords do not need to be known. This vulnerability is only exploitable when the nat option is set to the default, or auto_force_rport.

Published: 2019-11-22T16:22:55.000Z
Updated: 2024-08-05T02:02:38.262Z

An issue was discovered in manager.c in Sangoma Asterisk through 13.x, 16.x, 17.x and Certified Asterisk 13.21 through 13.21-cert4. A remote authenticated Asterisk Manager Interface (AMI) user without system authorization could use a specially crafted Originate AMI request to execute arbitrary system commands.

Published: 2019-11-22T17:31:16.000Z
Updated: 2024-08-05T01:54:14.490Z

main/translate.c in Sangoma Asterisk 13.28.0 and 16.5.0 allows a remote attacker to send a specific RTP packet during a call and cause a crash in a specific scenario.

Published: 2019-09-09T12:50:30.000Z
Updated: 2024-08-05T00:56:22.040Z

res_pjsip_t38 in Sangoma Asterisk 15.x before 15.7.4 and 16.x before 16.5.1 allows an attacker to trigger a crash by sending a declined stream in a response to a T.38 re-invite initiated by Asterisk. The crash occurs because of a NULL session media object dereference.

Published: 2019-09-09T20:48:12.000Z
Updated: 2024-08-05T00:42:03.821Z

An issue was discovered in Asterisk Open Source through 13.27.0, 14.x and 15.x through 15.7.2, and 16.x through 16.4.0, and Certified Asterisk through 13.21-cert3. A pointer dereference in chan_sip while handling SDP negotiation allows an attacker to crash Asterisk when handling an SDP answer to an outgoing T.38 re-invite. To exploit this vulnerability an attacker must cause the chan_sip module to send a T.38 re-invite request to them. Upon receipt, the attacker must send an SDP answer containing both a T.38 UDPTL stream and another media stream containing only a codec (which is not permitted according to the chan_sip configuration).

Published: 2019-07-12T19:24:37.000Z
Updated: 2024-08-04T23:41:10.494Z

Buffer overflow in res_pjsip_messaging in Digium Asterisk versions 13.21-cert3, 13.27.0, 15.7.2, 16.4.0 and earlier allows remote authenticated users to crash Asterisk by sending a specially crafted SIP MESSAGE message.

Published: 2019-07-12T19:19:52.000Z
Updated: 2024-08-04T23:32:55.236Z

An issue was discovered in Asterisk through 13.19.1, 14.x through 14.7.5, and 15.x through 15.2.1, and Certified Asterisk through 13.18-cert2. res_pjsip allows remote authenticated users to crash Asterisk (segmentation fault) by sending a number of SIP INVITE messages on a TCP or TLS connection and then suddenly closing the connection.

Published: 2018-02-22T00:00:00.000Z
Updated: 2024-08-05T06:24:11.758Z

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.

Published: 2018-02-22T00:00:00.000Z
Updated: 2024-08-05T06:24:11.787Z

A Buffer Overflow issue was discovered in Asterisk through 13.19.1, 14.x through 14.7.5, and 15.x through 15.2.1, and Certified Asterisk through 13.18-cert2. When processing a SUBSCRIBE request, the res_pjsip_pubsub module stores the accepted formats present in the Accept headers of the request. This code did not limit the number of headers it processed, despite having a fixed limit of 32. If more than 32 Accept headers were present, the code would write outside of its memory and cause a crash.

Published: 2018-02-22T00:00:00.000Z
Updated: 2024-08-05T06:24:11.866Z

There is a stack consumption vulnerability in the res_http_websocket.so module of Asterisk through 13.23.0, 14.7.x through 14.7.7, and 15.x through 15.6.0 and Certified Asterisk through 13.21-cert2. It allows an attacker to crash Asterisk via a specially crafted HTTP request to upgrade the connection to a websocket.

Published: 2018-09-24T22:00:00.000Z
Updated: 2024-08-05T10:47:04.106Z

An issue was discovered in Asterisk Open Source 13.x before 13.21.1, 14.x before 14.7.7, and 15.x before 15.4.1 and Certified Asterisk 13.18-cert before 13.18-cert4 and 13.21-cert before 13.21-cert2. When endpoint specific ACL rules block a SIP request, they respond with a 403 forbidden. However, if an endpoint is not identified, then a 401 unauthorized response is sent. This vulnerability just discloses which requests hit a defined endpoint. The ACL rules cannot be bypassed to gain access to the disclosed endpoints.

Published: 2018-06-12T04:00:00.000Z
Updated: 2024-08-05T08:30:59.565Z

An issue was discovered in Asterisk 13.18.4 and older, 14.7.4 and older, 15.1.4 and older, and 13.18-cert1 and older. A select set of SIP messages create a dialog in Asterisk. Those SIP messages must contain a contact header. For those messages, if the header was not present and the PJSIP channel driver was used, Asterisk would crash. The severity of this vulnerability is somewhat mitigated if authentication is enabled. If authentication is enabled, a user would have to first be authorized before reaching the crash point.

Published: 2017-12-23T00:00:00.000Z
Updated: 2024-08-05T21:06:49.280Z

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.

Published: 2017-12-13T20:00:00.000Z
Updated: 2024-08-05T20:59:17.343Z

An issue was discovered in chan_skinny.c in Asterisk Open Source 13.18.2 and older, 14.7.2 and older, and 15.1.2 and older, and Certified Asterisk 13.13-cert7 and older. If the chan_skinny (aka SCCP protocol) channel driver is flooded with certain requests, it can cause the asterisk process to use excessive amounts of virtual memory, eventually causing asterisk to stop processing requests of any kind.

Published: 2017-12-02T00:00:00.000Z
Updated: 2024-08-05T20:43:59.739Z

An issue was discovered in Asterisk Open Source 13 before 13.18.1, 14 before 14.7.1, and 15 before 15.1.1 and Certified Asterisk 13.13 before 13.13-cert7. A memory leak occurs when an Asterisk pjsip session object is created and that call gets rejected before the session itself is fully established. When this happens the session object never gets destroyed. Eventually Asterisk can run out of memory and crash.

Published: 2017-11-09T00:00:00.000Z
Updated: 2024-08-05T20:35:19.786Z

A Buffer Overflow issue was discovered in Asterisk Open Source 13 before 13.18.1, 14 before 14.7.1, and 15 before 15.1.1 and Certified Asterisk 13.13 before 13.13-cert7. No size checking is done when setting the user field for Party B on a CDR. Thus, it is possible for someone to use an arbitrarily large string and write past the end of the user field storage buffer. NOTE: this is different from CVE-2017-7617, which was only about the Party A buffer.

Published: 2017-11-09T00:00:00.000Z
Updated: 2024-08-05T20:35:19.933Z

The DB dialplan function in Asterisk Open Source 1.8.x before 1.8.32, 11.x before 11.1.4.1, 12.x before 12.7.1, and 13.x before 13.0.1 and Certified Asterisk 1.8 before 1.8.28-cert8 and 11.6 before 11.6-cert8 allows remote authenticated users to gain privileges via a call from an external protocol, as demonstrated by the AMI protocol.

Published: 2014-11-24T15:00:00.000Z
Updated: 2024-08-06T13:18:47.847Z

ConfBridge in Asterisk 11.x before 11.14.1, 12.x before 12.7.1, and 13.x before 13.0.1 and Certified Asterisk 11.6 before 11.6-cert8 allows remote authenticated users to (1) gain privileges via vectors related to an external protocol to the CONFBRIDGE dialplan function or (2) execute arbitrary system commands via a crafted ConfbridgeStartRecord AMI action.

Published: 2014-11-24T15:00:00.000Z
Updated: 2024-08-06T13:18:48.431Z

Use-after-free vulnerability in the PJSIP channel driver in Asterisk Open Source 12.x before 12.7.1 and 13.x before 13.0.1, when using the res_pjsip_refer module, allows remote attackers to cause a denial of service (crash) via an in-dialog INVITE with Replaces message, which triggers the channel to be hung up.

Published: 2014-11-24T15:00:00.000Z
Updated: 2024-08-06T13:18:47.980Z

Race condition in the chan_pjsip channel driver in Asterisk Open Source 12.x before 12.7.1 and 13.x before 13.0.1 allows remote attackers to cause a denial of service (assertion failure and crash) via a cancel request for a SIP session with a queued action to (1) answer a session or (2) send ringing.

Published: 2014-11-24T15:00:00.000Z
Updated: 2024-08-06T13:18:48.176Z

The res_pjsip_acl module in Asterisk Open Source 12.x before 12.7.1 and 13.x before 13.0.1 does not properly create and load ACLs defined in pjsip.conf at startup, which allows remote attackers to bypass intended PJSIP ACL rules.

Published: 2014-11-24T15:00:00.000Z
Updated: 2024-08-06T13:18:48.273Z

The (1) VoIP channel drivers, (2) DUNDi, and (3) Asterisk Manager Interface (AMI) in Asterisk Open Source 1.8.x before 1.8.32.1, 11.x before 11.14.1, 12.x before 12.7.1, and 13.x before 13.0.1 and Certified Asterisk 1.8.28 before 1.8.28-cert3 and 11.6 before 11.6-cert8 allows remote attackers to bypass the ACL restrictions via a packet with a source IP that does not share the address family as the first ACL entry.

Published: 2014-11-24T15:00:00.000Z
Updated: 2024-08-06T13:18:48.295Z

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.

Published: 2014-06-17T14:00:00.000Z
Updated: 2024-08-06T11:04:28.479Z

Asterisk Open Source 1.8.x before 1.8.19.1, 10.x before 10.11.1, and 11.x before 11.1.2; Certified Asterisk 1.8.11 before 1.8.11-cert10; and Asterisk Digiumphones 10.x-digiumphones before 10.11.1-digiumphones, when anonymous calls are enabled, allow remote attackers to cause a denial of service (resource consumption) by making anonymous calls from multiple sources and consequently adding many entries to the device state cache.

Published: 2013-01-04T15:00:00.000Z
Updated: 2024-08-06T21:21:28.317Z

Multiple stack consumption vulnerabilities in Asterisk Open Source 1.8.x before 1.8.19.1, 10.x before 10.11.1, and 11.x before 11.1.2; Certified Asterisk 1.8.11 before 1.8.11-cert10; and Asterisk Digiumphones 10.x-digiumphones before 10.11.1-digiumphones allow remote attackers to cause a denial of service (daemon crash) via TCP data using the (1) SIP, (2) HTTP, or (3) XMPP protocol.

Published: 2013-01-04T11:00:00.000Z
Updated: 2024-08-06T21:21:28.331Z

Stack-based buffer overflow in the milliwatt_generate function in the Miliwatt application in Asterisk 1.4.x before 1.4.44, 1.6.x before 1.6.2.23, 1.8.x before 1.8.10.1, and 10.x before 10.2.1, when the o option is used and the internal_timing option is off, allows remote attackers to cause a denial of service (application crash) via a large number of samples in an audio packet.

Published: 2012-09-18T18:00:00.000Z
Updated: 2024-08-06T18:53:35.794Z

Stack-based buffer overflow in the ast_uri_encode function in main/utils.c in Asterisk Open Source before 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.1, 1.8.1.2, 1.8.2.; and Business Edition before C.3.6.2; when running in pedantic mode allows remote authenticated users to execute arbitrary code via crafted caller ID data in vectors involving the (1) SIP channel driver, (2) URIENCODE dialplan function, or (3) AGI dialplan function.

Published: 2011-01-20T18:00:00.000Z
Updated: 2024-08-06T21:58:24.451Z

The SIP channel driver in Asterisk Open Source 1.2.x before 1.2.34, 1.4.x before 1.4.26.1, 1.6.0.x before 1.6.0.12, and 1.6.1.x before 1.6.1.4; Asterisk Business Edition A.x.x, B.x.x before B.2.5.9, C.2.x before C.2.4.1, and C.3.x before C.3.1; and Asterisk Appliance s800i 1.2.x before 1.3.0.3 does not use a maximum width when invoking sscanf style functions, which allows remote attackers to cause a denial of service (stack memory consumption) via SIP packets containing large sequences of ASCII decimal characters, as demonstrated via vectors related to (1) the CSeq value in a SIP header, (2) large Content-Length value, and (3) SDP.

Published: 2009-08-12T10:00:00.000Z
Updated: 2024-08-07T05:59:57.107Z

SQL injection vulnerability in the Postgres Realtime Engine (res_config_pgsql) in Asterisk 1.4.x before 1.4.15 and C.x before C.1.0-beta6 allows remote attackers to execute arbitrary SQL commands via unknown vectors.

Published: 2007-11-30T01:00:00.000Z
Updated: 2024-08-07T15:54:26.995Z

SQL injection vulnerability in the Call Detail Record Postgres logging engine (cdr_pgsql) in Asterisk 1.4.x before 1.4.15, 1.2.x before 1.2.25, B.x before B.2.3.4, and C.x before C.1.0-beta6 allows remote authenticated users to execute arbitrary SQL commands via (1) ANI and (2) DNIS arguments.

Published: 2007-11-30T01:00:00.000Z
Updated: 2024-08-07T15:54:26.987Z

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.

Published: 2007-10-12T23:00:00.000Z
Updated: 2024-08-07T15:31:57.205Z

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.

Published: 2007-07-31T10:00:00.000Z
Updated: 2024-08-07T14:46:38.776Z

Integer signedness error in format_jpeg.c in Asterisk 1.2.6 and earlier allows remote attackers to execute arbitrary code via a length value that passes a length check as a negative number, but triggers a buffer overflow when it is used as an unsigned length.

Published: 2006-04-18T20:00:00.000Z
Updated: 2024-08-07T17:27:29.111Z