Asterisk 13.24.0-rc1, 15.7.0-rc1, and 16.1.0-rc1 Now Available [Asterisk News] (2)

The Asterisk Development Team would like to announce the release of Asterisk 13.24.0-rc1, 15.7.0-rc1, and 16.1.0-rc1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/as…

Asterisk 15.6.2 and 16.0.1 Now Available (Security) [Asterisk News] (2)

The Asterisk Development Team would like to announce security releases for Asterisk 15 and 16. The available releases are released as versions 15.6.2 and 16.0.1. These releases are available for immediate download at …

Welcome to the Asterisk Community [Uncategorized] (6)
How to shutdown span via CLI or API? [Asterisk APIs] (1)
Sip and pjsip configuration issue? [Asterisk Support] (10)
Support for RFC5168 in Asterisk13 [Asterisk Support] (4)
Making sense of PJSIP, relationship between AORs and Endpoints [Asterisk SIP] (3)
Cdr show in cli? [Asterisk Support] (8)
Digium TE420 Quad Span Card only uses one processor core [Asterisk Hardware] (2)
Asterisk Ports & Phones [Asterisk Hardware] (4)
Queue variables [Asterisk Dialplan] (3)
Message/ast_msg_queue stuck on Hangup() Application [Asterisk Dialplan] (1)
Dialing between web browser and soft phone [Asterisk WebRTC] (13)
Dial # before entering extension [Asterisk Dialplan] (10)
App_followme not waiting for Local/ Dial with AGI script [Asterisk Support] (2)
Asterisk 15 WebRTC Configured SIPML5 - but couldnt hear audio in browser [Asterisk WebRTC] (15)
Does anyone know what kind of sound is "pt-sss"? [Asterisk Support] (3)
Asterisk port/network issue [Asterisk SIP] (3)
Transcodin isuue with h264 codec in asterisk [Asterisk Support] (4)
Asterisk Script Help [Asterisk Dialplan] (3)
DTMF Issue since upgrade from 11.21 to 13.24.1 [Asterisk Support] (15)
Connecting Two Asterisk System [Asterisk Support] (3)
WebRTC SFU: How to learn whom does the video track belongs to? [Asterisk WebRTC] (3)
Remote peers audio quality issue [Asterisk SIP] (2)
Asterisk 13. RealTime [Asterisk Support] (11)
Calls are getting stuck with Pjsip and also some call-drop [Asterisk SIP] (7)
Multiple times Originate With Asterisk and PHP [Asterisk APIs] (7)
Problem with pjsip dial [Asterisk SIP] (5)
USB Versus PoE Phones [Asterisk Endpoints] (7)
Transfer recalls via REFER NOTIFY events [Asterisk SIP] (3)