Archives : April-2016
all,Im trying to configure a few conference bridges.Ive started with the very basic:[general][default_bridge]type=bridge[default_user]type=user[default_bridge]type=bridge[5340]type=bridgeHowever:confbridge list Conference Bridge Name UsersMarked Locked?=============================..
Asterisk users, This issue started after we updated all of our systems to Asterisk 13.7.2. Details:CentOS 5.11DAHDI Version: 2.9.1.1Asterisk updated from 13.1.0 to 13.7.2Asterisk takes around 5 rings to answer the call (before it took 2)Only seems..
Asterisk Project Security Advisory – AST-2016-005 ProductAsterisk SummaryTCP denial of service in PJProjectNature of AdvisoryCrash/Denial of Service SusceptibilityRemote Unauthenticated Sessions Severity CriticalExploits KnownNo Reported OnFebruary ..
Asterisk Project Security Advisory – AST-2016-004 ProductAsterisk SummaryLong Contact URIs in REGISTER requests can crashAsteriskNature of AdvisoryRemote CrashSusceptibilityRemote Authenticated Sessions Severity Major Exploits KnownNo Reported OnJanu..
The Asterisk Development Team has announced security releases for Certified Asterisk 13.1 and Asterisk 13. The available security releases are released as versions 13.1-cert5, and 13.8.1.These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases..
Hey all. This isnt directly an Asterisk question, but it is Asterisk related because I am using SIP on asterisk.The last couple of days I found that our asterisk box was having all packets originating from port 5060 being blocked.If I moved my SIP p..
Hello.I developed a little project (a PoC) to integrate Asterisk IVRs withother softwares, allowing that data already entered in IVR can be used in other stages of a customer service, for example.The main goal is to provide more efficiency and interoperabil..
Olle,Redirecting the question to users mailing list. Could you point out how can I *dynamically* pass both the SIP peer and request-URI in the dial command. I want be able to use same SIP peer to route to different SIP end points. Im currently do..
I want to use Asterisk to use Kamailio as an outbound proxy for routing calls to remote SIP end points, one option could be to use a default peer, but in my case, my outbound proxy can change based on the remote end point, so this option doesnt wo..
All;I was wondering what people are doing to bill customers for minutes. Iknow that A2Billing is a popular option, but I was wondering if there are other good alternatives. They dont necessarily have to be free, but they need to be cost effective. ..