Asterisk Blog

STIR/SHAKEN in Asterisk

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

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Asterisk-18.0.0-rc1 Released!

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

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Common SIP URI Issues

When transitioning from the chan_sip channel driver to chan_pjsip one of the items that can catch people off guard is the use of SIP URIs within PJSIP. This is because in chan_sip these are generated on your behalf based on different configuration options while in chan_pjsip we leave this up to the user. Let’s take

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Local Channel Multistream and Re-Negotiation Support

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

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ARI Create Channel With Variables

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

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ARI Create Channel With Variables

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

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Asterisk Versioning and Releases

Prompted by a post on the Asterisk users mailing list I thought I’d write a bit about Asterisk versioning and releases for this blog post. The Asterisk project at any given time has between 2 and 3 major versions of Asterisk being released. When I say major version I’m referring to an entirely different branch

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Jared Smith’s Departure from Sangoma

Those of you who know me know that I’m very passionate about open source and about open source telephony projects. As the VP of Open Source Community Development here at Sangoma, I’m proud of the work that we’ve been able to do to ensure that the Asterisk and FreePBX open source projects remain strong and

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New PJSIP Logging Functionality

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

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