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Search Parameters:
  • Results Type: Overview
  • Keyword (text search): cpe:2.3:a:digium:certified_asterisk:16.8:cert1-rc2:*:*:*:*:*:*
  • CPE Name Search: true
There are 6 matching records.
Displaying matches 1 through 6.
Vuln ID Summary CVSS Severity
CVE-2021-32558

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: July 30, 2021; 10:15:16 AM -0400
V3.1: 7.5 HIGH
V2.0: 5.0 MEDIUM
CVE-2021-26713

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: February 19, 2021; 3:15:13 PM -0500
V3.1: 6.5 MEDIUM
V2.0: 4.0 MEDIUM
CVE-2021-26712

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: February 18, 2021; 4:15:11 PM -0500
V3.1: 7.5 HIGH
V2.0: 5.0 MEDIUM
CVE-2021-26906

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: February 18, 2021; 3:15:12 PM -0500
V3.1: 5.9 MEDIUM
V2.0: 4.3 MEDIUM
CVE-2021-26717

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: February 18, 2021; 3:15:12 PM -0500
V3.1: 7.5 HIGH
V2.0: 5.0 MEDIUM
CVE-2020-28327

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.

Published: November 06, 2020; 2:15:14 PM -0500
V3.1: 5.3 MEDIUM
V2.0: 2.1 LOW