How to make asterisk work correctly in the media path [Asterisk SIP] (2)
Possible segfault in chan_sip with Asterisk 13.24.1 [Asterisk SIP] (2)
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Testing asterisk with SIPp UAC and UAS scenarios [Asterisk SIP] (2)
Pjsip rewrite port in from/to headers on reply [Asterisk SIP] (11)
What’s the all steps if i need to call from number mobile by gsm to Asterisk server [Asterisk Support] (2)
WebRTC - Asterisk [Asterisk WebRTC] (6)
Problem PJSIP 2.6 [Asterisk Endpoints] (3)
Voicemail not being recorded on Custom Device [Asterisk SIP] (3)
Can not register two sip phone [Asterisk Support] (7)
Panasonic Door Intercom kx-ntv160 wil not register [Asterisk SIP] (15)
Problem with Asterisk server, when call from outside “GSM” [Asterisk SIP] (2)
Exclude manager events reading [Asterisk APIs] (4)
Asterisk 16.1 WebRTC configuration not working when calling web client (other way is ok) [Asterisk WebRTC] (8)
Asterisk - Screensharing feature [Asterisk WebRTC] (2)
Asterisk SIP Lagged issue [Asterisk SIP] (2)
JSSIP 401 error [Asterisk SIP] (4)
Make dialing multiple channels not give up if one has congestion [Asterisk Support] (1)
How to send SMS using Asterisk [Asterisk Integration] (9)
DAHDI hangup not received [Asterisk Hardware] (7)
Trouble with queue_log table [Asterisk Support] (5)
Asterisk 16.2.0-rc2 Now Available [Asterisk News] (3)
Asterisk 13.25.0-rc2 Now Available [Asterisk News] (4)
PHP Agi in not working in cron [Asterisk APIs] (7)
Multiple PJSIP registrations to one SIP server [Asterisk SIP] (2)
Pjsip: How to identify endpoint by transport [Asterisk SIP] (12)
Asterisk 16 and NAPTR/SRV [Asterisk SIP] (3)
Does Asterisk supports camp feature? [Asterisk Support] (11)
Hangup on a defined response code - 183 [Asterisk Support] (3)
Streaming Audio Server-Music On Hold [Asterisk Support] (1)