Hey people.I just released B9 version 0.3.This version contains new commands (create conference, invite conference), but the major feature is socket connection that can be configured in a console admin page.In the page you can enable socket connectio..
Asterisk Project Security Advisory – AST-2014-019 ProductAsterisk SummaryRemote Crash Vulnerability in WebSocket ServerNature of AdvisoryDenial of Service SusceptibilityRemote Unauthenticated Sessions Severity ModerateExploits KnownNo Reported On30 Octo..
I have a bunch of these in my logs:[Dec9 08:21:21] NOTICE[-1][C-00000285] chan_sip.c: Failed to authenticate deviceeinstein;tage696e737465696e0131323530333532333739The problem is that I already know my own IP address.How do Idetermine the address..
Im working with a SIP provider to try and transition our sip connection with them to PJSIP.I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages.Im currently running Aster..
It is possible to disable/remove INVITE method in 200 OK responses?I want to receive from another SIP/PBX the the media path redirection in a UPDATE message rather than an INVITE, after calls are transfered.My asterisk is version 11.e.g:—————————-..
In my office have setuped the Elastix machine and i have a static IP(external IP given by ISP), now the issue is that whenerve call from outside sip extensions which is register to the sip server , am not able hear audio from both side. both callee ..