Taskpool + PJSIP Has Landed!
Joshua C. Colp
No Comments
Last year I wrote about a new API I created, taskpool, and how stasis was moved over to it. If you haven’t read that post
Last year I wrote about a new API I created, taskpool, and how stasis was moved over to it. If you haven’t read that post
Sangoma provides two SIP trunking services which are available to customers, SIPStation which is a great solution for the everyday user and VoIP Innovations which
Exchanging media between Asterisk and external apps has fairly involved for a few reasons, the main one being that RTP, the primary protocol for exchanging
Historically, using ARI required using HTTP for making REST requests and getting their responses, and a Websocket for receiving events. With Asterisk 20.14.0, 21.9.0 and
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 Extension to HTTP
Up until recently Asterisk only supported RFC 4733 RTP events when using 8KHz codecs like G.711. However, with this recent change, Asterisk now supports the
I recently did a STIR/SHAKEN webinar about the new implementation available in Asterisk. You can find the recording for that here. This webinar reinforced my
Overview If you’re familiar with Asterisk, you probably know that it uses a third-party project called pjproject. This is a major part of the PJSIP
Overview Realtime has been around for a while now, but operating systems are constantly evolving. Because of this, guides can become outdated. I find myself
If you keep an eye on the Asterisk gitlog, you may have seen some additions to app_voicemail. These changes include the ability to ‘show’ a
