Did you know that sipp can play back RTP streams? A lot of long time users may not! Did you know that you can use Wireshark to extract a RTP stream...
If you didn’t already know, both SIP and HTTP share the same digest authentication mechanism described all the way back in RFC-2069 “An...
Overview When recording audio, it can be useful to split streams from one another. For example, if you are in a call with someone, you might want to...
As mentioned in this post, Asterisk now supports the use of RFC4733 digits in common bitrates beyond 8kHz. At the end of the post, we mention the...
AstriCon 2025 registration is now open! You can take advantage of early bird pricing by registering here. AstriCon this year will span a total...
A recurring theme I’m seeing lately is people deploying VoIP, running into issues, and not approaching their issues from the perspective of...
It’s been a full year since we migrated Asterisk to GitHub. It didn’t go perfectly smoothly but knowing what we know now, would we do it again?...
In a previous blog post we talked about using Asterisk’s uni-cast functionality as a bridge between the PSTN and an external service. This...
Up until recently Asterisk only supported RFC 4733 RTP events when using 8KHz codecs like G.711. However, with this recent change, Asterisk now...
We are thrilled to announce the creation of the Asterisk 22 branch! This marks a significant milestone in our ongoing commitment to providing robust...
