Filtered by vendor Digium
Subscriptions
Total
119 CVE
| CVE | Vendors | Products | Updated | CVSS v3.1 |
|---|---|---|---|---|
| CVE-2021-31878 | 1 Digium | 1 Asterisk | 2024-11-21 | 6.5 Medium |
| An issue was discovered in PJSIP in Asterisk before 16.19.1 and before 18.5.1. To exploit, a re-INVITE without SDP must be received after Asterisk has sent a BYE request. | ||||
| CVE-2021-26906 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 5.9 Medium |
| 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. | ||||
| CVE-2021-26717 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 7.5 High |
| 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. | ||||
| CVE-2021-26713 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 6.5 Medium |
| 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. | ||||
| CVE-2021-26712 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | 7.5 High |
| 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. | ||||
| CVE-2020-35776 | 1 Digium | 1 Asterisk | 2024-11-21 | 6.5 Medium |
| 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. | ||||
| CVE-2020-35652 | 1 Digium | 1 Asterisk | 2024-11-21 | 6.5 Medium |
| 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. | ||||
| CVE-2020-28327 | 2 Digium, Sangoma | 2 Certified Asterisk, Asterisk | 2024-11-21 | 5.3 Medium |
| A res_pjsip_session crash was discovered in Asterisk Open Source 13.x before 13.37.1, 16.x before 16.14.1, 17.x before 17.8.1, and 18.x before 18.0.1. and Certified Asterisk before 16.8-cert5. Upon receiving a new SIP Invite, Asterisk did not return the created dialog locked or referenced. This caused a gap between the creation of the dialog object, and its next use by the thread that created it. Depending on some off-nominal circumstances and timing, it was possible for another thread to free said dialog in this gap. Asterisk could then crash when the dialog object, or any of its dependent objects, were dereferenced or accessed next by the initial-creation thread. Note, however, that this crash can only occur when using a connection-oriented protocol (e.g., TCP or TLS, but not UDP) for SIP transport. Also, the remote client must be authenticated, or Asterisk must be configured for anonymous calling. | ||||
| CVE-2019-7251 | 1 Digium | 1 Asterisk | 2024-11-21 | N/A |
| 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. | ||||
| CVE-2019-18976 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 7.5 High |
| 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. | ||||
| CVE-2019-18790 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 6.5 Medium |
| 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. | ||||
| CVE-2019-18610 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 8.8 High |
| 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. | ||||
| CVE-2019-15639 | 1 Digium | 1 Asterisk | 2024-11-21 | 7.5 High |
| 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. | ||||
| CVE-2019-15297 | 1 Digium | 1 Asterisk | 2024-11-21 | 6.5 Medium |
| 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. | ||||
| CVE-2019-13161 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | 5.3 Medium |
| 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). | ||||
| CVE-2019-12827 | 1 Digium | 2 Asterisk, Certified Asterisk | 2024-11-21 | N/A |
| 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. | ||||
| CVE-2018-7287 | 1 Digium | 1 Asterisk | 2024-11-21 | N/A |
| An issue was discovered in res_http_websocket.c in Asterisk 15.x through 15.2.1. If the HTTP server is enabled (default is disabled), WebSocket payloads of size 0 are mishandled (with a busy loop). | ||||
| CVE-2018-7286 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | N/A |
| 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. | ||||
| CVE-2018-7285 | 1 Digium | 1 Asterisk | 2024-11-21 | N/A |
| 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-2018-7284 | 2 Debian, Digium | 3 Debian Linux, Asterisk, Certified Asterisk | 2024-11-21 | N/A |
| 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. | ||||