Digium Asterisk 13.3.1
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cpe:2.3:a:digium:asterisk:13.3.1:*:*:*:*:*:*:*
part: a version: 13.3.1 update: *
| Vendor | Digium (05ad29b7-5b41-56d5-935d-a279ab7f14bc) |
|---|---|
| Product | Asterisk (a75a6886-b0b4-5160-9cfa-f749f3c86956) |
| Edition | * |
| Language | * |
| Software edition | * |
| Target software | * |
| Target hardware | * |
| Other | * |
| Notes | Imported from NVD CPE 2.0 feed |
PURL mappings
| PURL | Source | Last updated |
|---|---|---|
pkg:github/asterisk/asterisk |
purl2cpe | 2026-06-01 10:15:41.800333 |
Vulnerability references
| Identifier | cpeApplicability | Submitted | db.gcve.eu details | Rationale |
|---|---|---|---|---|
CVE:CVE-2016-9938 |
vulnerable | 2026-06-08 05:08:25.045400 |
Details available
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.
Published: 2016-12-12T21:00:00.000Z
Updated: 2024-08-06T03:07:31.471Z |
Imported from gcve-enriched-dumps CVE data |
CVE:CVE-2016-7551 |
vulnerable | 2026-06-08 05:08:13.029800 |
Details available
chain_sip in Asterisk Open Source 11.x before 11.23.1 and 13.x 13.11.1 and Certified Asterisk 11.6 before 11.6-cert15 and 13.8 before 13.8-cert3 allows remote attackers to cause a denial of service (port exhaustion).
Published: 2017-04-17T16:00:00.000Z
Updated: 2024-08-06T02:04:55.787Z |
Imported from gcve-enriched-dumps CVE data |
CVE:CVE-2015-3008 |
vulnerable | 2026-06-08 05:06:36.548003 |
Details available
Asterisk Open Source 1.8 before 1.8.32.3, 11.x before 11.17.1, 12.x before 12.8.2, and 13.x before 13.3.2 and Certified Asterisk 1.8.28 before 1.8.28-cert5, 11.6 before 11.6-cert11, and 13.1 before 13.1-cert2, when registering a SIP TLS device, does not properly handle a null byte in a domain name in the subject's Common Name (CN) field of an X.509 certificate, which allows man-in-the-middle attackers to spoof arbitrary SSL servers via a crafted certificate issued by a legitimate Certification Authority.
Published: 2015-04-10T14:00:00.000Z
Updated: 2024-08-06T05:32:21.258Z Reference links |
Imported from gcve-enriched-dumps CVE data |
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