Archives : August-2017
Asterisk Project Security Advisory – AST-2017-007 ProductAsterisk SummaryRemote Crash Vulerability in res_pjsipNature of AdvisoryDenial of Service SusceptibilityRemote Unauthenticated Sessions Severity ModerateExploits KnownNo Reported OnAugust 30, 2..
Asterisk Project Security Advisory – AST-2017-005 ProductAsterisk SummaryMedia takeover in RTP stack Nature of AdvisoryUnauthorized data disclosureSusceptibilityRemote Unauthenticated Sessions Severity CriticalExploits KnownNo Reported OnMay 17, 2..
The Asterisk Development Team has announced security releases for Asterisk11, 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.These releases ..
Is there a way that I can modify the source code for the voicemail application?I need to change some of the options in the users interface to make it work like an existing system that Im replacing. ..
is there somebody who is using say.conf mode=new in Asterisk 13?im searching for tips what to try inhttps://issues.asterisk.org/jira/browse/ASTERISK-..
Were experimenting with using Asterisk (14.6.0) for video conferences. This test has three endpoints, a Polycom Trio with its video accessory, and two desktops running Linphone.The video is all H.264.Were using Opus for audio on the Linphone Wind..
Let me provide the details first:* Asterisk 1.8.32 on CentOS behind the NAT firewall* Two (2) SIP trunks with canreinvite=no and directmedia=noIf a call comes from either trunk and is bridged to a local extension there is never a problem with aud..
folks.I have a couple of questions regarding RTP.The background of my inquiry is that I have packet captures of SIP and RTP traffic on an Asterisk and Broadworks SIP trunk and the RTP many times has a time stamp that rewinds by 480 using g.711u. ..
since al long time I have used UNIQUEID for identify calls in my dialplan, statistics…Now I have had an problem, after I have checked log file I found out following:calls same time ( hours:seconds) came in.CallID, DID, channel name (00003cf9 to 00003c..
Ive had two Asterisk crashes today that seem to be caused by errors where chan->tech_pvt is pointing to something that cant be deallocated and I think I see a reference count bug in the above function.It contains:if (data->chan_old_vsrc) {ast_channel_unref(data->chan_old_vsrc);}Shoul..