Default status is the baseline for the product, each version can override it (e.g. patched versions marked unaffected).
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| n/a | n/a | affected |
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Configuration 2 [-]
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No data available yet.
| Source | ID | Title |
|---|---|---|
EUVD |
EUVD-2016-10725 | 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. |
No CVSS v4.0
No CVSS v3.1
This CVE is not in the KEV list.
Exploit prediction score: 3.4% (3.429% probability within 30 days)
Key SSVC decision points have not yet been added.
No history.