[dropshadowbox align=”none” effect=”lifted-both” width=”auto” height=”” background_color=”#ffffff” border_width=”1″ border_color=”#dddddd” ]tl;dr: We fixed the vulnerabilities. If you’d like to read the conclusion of this admittedly long saga,
Do you use WebRTC with Asterisk? Did you notice calls stop working after updating Google Chrome to version 57? Are you curious why that happened?
In the previous post, Josh introduced the forthcoming addition of streams to Asterisk. I’m going to piggyback on that to introduce a unified SDP API
A new feature that was initially implemented during a recent visit to SIPit has now been merged into the 13, 14, and master Asterisk branches.
When the PJSIP work for Asterisk began one of the primary concerns kept in mind was that it be extensible. One of the APIs derived
The PJSIP library now used by Asterisk to provide SIP support has included basic SIP DNS support for quite some time. However through using it
Asterisk 13.8.0 will come with a new option for enabling PJSIP functionality. This functionality is called bundling and comes courtesy of a community member, George
Debugging SIP Messages the Traditional Way Since its release, the PJSIP stack has provided logging of SIP message traffic via the pjsip set logger CLI
Outbound SIP registrations are a commonly used practice in Asterisk. They allow an upstream server, such as one in use by an ITSP, to know
