Digium Asterisk 13.0.0
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cpe:2.3:a:digium:asterisk:13.0.0:*:*:*:*:*:*:*
part: a version: 13.0.0 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 10:15:41.800184 |
Vulnerability references
| Identifier | cpeApplicability | Submitted | db.gcve.eu details | Rationale |
|---|---|---|---|---|
CVE:CVE-2016-9938 |
vulnerable | 2026-06-08 05:08:25.039977 |
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 05:08:13.023989 |
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-2015-1558 |
vulnerable | 2026-06-08 05:06:25.835432 |
Details available
Asterisk Open Source 12.x before 12.8.1 and 13.x before 13.1.1, when using the PJSIP channel driver, does not properly reclaim RTP ports, which allows remote authenticated users to cause a denial of service (file descriptor consumption) via an SDP offer containing only incompatible codecs.
Published: 2015-02-09T11:00:00.000Z
Updated: 2024-08-06T04:47:17.146Z |
Imported from gcve-enriched-dumps CVE data |
CVE:CVE-2014-9374 |
vulnerable | 2026-06-08 05:06:11.562211 |
Details available
Double free vulnerability in the WebSocket Server (res_http_websocket module) in Asterisk Open Source 11.x before 11.14.2, 12.x before 12.7.2, and 13.x before 13.0.2 and Certified Asterisk 11.6 before 11.6-cert9 allows remote attackers to cause a denial of service (crash) by sending a zero length frame after a non-zero length frame.
Published: 2014-12-12T15:00:00.000Z
Updated: 2024-08-06T13:40:25.047Z Reference links
|
Imported from gcve-enriched-dumps CVE data |
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