| Package | Affected versions | Patched versions |
|---|---|---|
| asterisk | = 1.8.0= 1.8.1= 1.8.1.1= 1.8.1.2= 1.8.11.0= 1.8.11.1= 1.8.13.0= 1.8.2= 1.8.2.1= 1.8.2.2= 1.8.2.3= 1.8.2.4= 1.8.3= 1.8.3.1= 1.8.3.2= 1.8.3.3= 1.8.4= 1.8.4.1= 1.8.4.2= 1.8.4.3= 1.8.4.4= 1.8.5= 1.8.5.0= 1.8.6.0= 1.8.7.0= 1.8.7.1= 1.8.8.0= 1.8.8.1= 1.8.8.2= 1.8.9.0= 1.8.9.2= 1.8.9.3= 10.0.0= 10.0.1= 10.1.0= 10.1.1= 10.1.2= 10.1.3= 10.2.0= 10.2.1= 10.3.0= 10.3.1= 10.4.0= 10.4.1= 10.4.2= 10.5.0= 10.5.1 | |
| asterisk business edition | = c.3.1= c.3.3= c.3.7.4 | |
| asteriske | = 1.8.8.0= 1.8.9.1 | |
| certified asterisk | = 1.8.11 |
channels/chan_sip.c in Asterisk Open Source 1.8.x before 1.8.13.1 and 10.x before 10.5.2, Asterisk Business Edition C.3.x before C.3.7.5, Certified Asterisk 1.8.11-certx before 1.8.11-cert4, and Asterisk Digiumphones 10.x.x-digiumphones before 10.5.2-digiumphones does not properly handle a provisional response to a SIP reINVITE request, which allows remote authenticated users to cause a denial of service (RTP port exhaustion) via sessions that lack final responses.
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