Kia Ora! (Or Be Healthy) (Don’t ask – I like greetings from various languages) If you haven’t noticed from the various emails and posts Asterisk 18.0.0 has now been released and is available for download here! As previously mentioned in our blog post for 18.0.0-rc1 this is an LTS release, meaning it will be supported
Video has been a continued theme of Asterisk for some years now. We put into place the foundation to allow us to do video better, and have over time taken advantage of this and advanced things further. I thought I would take a little bit of time to reflect back on what has been done.
Codec negotiation in Asterisk has been one of its deepest darkest secrets. It’s been around since the beginning and over the past two decades it’s grown and mutated into one of the least understood parts of Asterisk. With Advanced Codec Negotiation that’s about to change! One of the Asterisk team’s goals for 2020 was to dig
What is STIR/SHAKEN? STIR/SHAKEN (Secure Telephony Identity Revisited / Signature-based Handling of Asserted information using toKENs) is a new technology that the telecommunications industry is using to help combat telephony fraud. We’ve all received spam calls, and some of us may have even received a call from a caller ID we recognized, but ended up
Greetings Everyone! Another year has flown by and while we now live in interesting times this has not stopped the development and progress of Asterisk or the Astricon conference[1], so much so that the first release candidate of Asterisk 18 is here[2]. The first step in creating the 18.0.0 release of Asterisk is tagging a
When stream support was initially added to Asterisk we did it in the most backwards compatible way possible to ensure that we did not have to modify the entirety of Asterisk. This has allowed us to gradually improve parts of Asterisk as we’ve expanded our stream and video support. To that end the next part
In a blog post long ago we talked about the addition of the create and dial ARI functionality for allowing channels to exist within ARI applications before they have been answered. This has seen use by various people and it came to light that it presented a slight difference in API definition in comparison to
In a blog post long ago we talked about the addition of the create and dial ARI functionality for allowing channels to exist within ARI applications before they have been answered. This has seen use by various people and it came to light that it presented a slight difference in API definition in comparison to
When PJSIP support was written for Asterisk we naturally needed the ability to display the SIP messages being sent and received. This took the form of the res_pjsip_logger module which hooks into the message sending and receiving path and logs the messages. The functionality was written to be familiar to users of chan_sip by allowing
Within the Asterisk project we are constantly trying to improve our processes and data collection when a problem is encountered to reduce the back and forth for getting information. This initially started with the ast_coredumper script that is now the recommended way to collect information from a running Asterisk process or from a core dump
