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CVE Record

CVE-2016-9938: An issue was discovered in Asterisk Open Source 11.x before 11.25.1, 13.x before 13.13.1, and 14.x before 1...

An issue was discovered in Asterisk Open Source 11.x before 11.25.1, 13.x before 13.13.1, and 14.x before 14.2.1 and Certified Asterisk 11.x before 11.6-cert16 and 13.x before 13.8-cert4. The chan_sip channel driver has a liberal definition for whitespace when attempting to strip the content between a SIP header name and a colon character. Rather than following RFC 3261 and stripping only spaces and horizontal tabs, Asterisk treats any non-printable ASCII character as if it were whitespace. This means that headers such as Contact\x01: will be seen as a valid Contact header. This mostly does not pose a problem until Asterisk is placed in tandem with an authenticating SIP proxy. In such a case, a crafty combination of valid and invalid To headers can cause a proxy to allow an INVITE request into Asterisk without authentication since it believes the request is an in-dialog request. However, because of the bug described above, the request will look like an out-of-dialog request to Asterisk. Asterisk will then process the request as a new call. The result is that Asterisk can process calls from unvetted sources without any authentication. If you do not use a proxy for authentication, then this issue does not affect you. If your proxy is dialog-aware (meaning that the proxy keeps track of what dialogs are currently valid), then this issue does not affect you. If you use chan_pjsip instead of chan_sip, then this issue does not affect you.

UnknownCVSS not scoredNot KEV-listedUpdated
Glexia's TakeAutomated analysis

Security readout for executives and security teams

This Asterisk issue can let unauthenticated SIP calls be accepted in specific proxy-backed deployments. The risk is not broad: it requires chan_sip plus an authenticating SIP proxy that is not dialog-aware. If present, it can bypass call admission controls and create operational or fraud exposure. Exposure is limited to Asterisk Open Source 11.x, 13.x, 14.x and Certified Asterisk 11.x or 13.x using chan_sip behind a non-dialog-aware authenticating SIP proxy. chan_pjsip and deployments without proxy-based authentication are stated as unaffected. Treat as priority remediation for Asterisk voice systems using proxy-based SIP authentication. The business risk is unauthorized call processing, including fraud or service abuse, but only in the affected deployment pattern described by the vendor. Mitigation focus: Upgrade to the fixed Asterisk or Certified Asterisk versions named by the vendor.; Confirm whether chan_sip is in use; migrate to chan_pjsip where appropriate.; Use a dialog-aware SIP proxy if proxy-based authentication is required..

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Vulnerability profileCVE Program record
Severity
Unknown
CVSS
Not scored
Known Exploited
No
Published
Official CVE source material

CNA and ADP enrichment extracted from CVE v5

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