do you have idea if is possible detect if a call to device(1) is from queue? (i.e. if app_queue set some variable) exten => 800,1,queue(sales) ; queue pick exten 20 exten => 20,1,noop(detect variables) exten => 20,n,Dial(SIP/20) (1) its through a lo..
Author : Marek Červenka
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whats your experience with asterisk compiled with libsrtp 2.x and WebRTC(pjsip)?
issues/crashes/speed/cpu usage?
Marek
official status https://wiki.asterisk.org/wiki/display/AST/lib..
i have webrtc client chrome69/jssip which is connecting to asterisk 13.23.1/pjsip i have strange problem where pjsip aor stays in status created sip trace on asterisk looks ok. do you think if this can be bug? test*CLI> pjsip show aors A..
i met with this interesting situation [Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport 8 since no request was received in 32 seconds [Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c: Shutting d..
I want use asterisk+pjsip as voip client with multiple registrations (perf testing), but my problem is contact on the o..
i know about architecture limits of app_queuehttps://issues.asterisk.org/jira/browse/ASTERISK-25806what CPUs are you actually using for asterisk + app_queue ? (my actual scenario 90simult calls, 50agents, call recording to SSD (mi..
is there somebody who is using say.conf mode=new in Asterisk 13?im searching for tips what to try inhttps://issues.asterisk.org/jira/browse/ASTERISK-..
do you have someone example ofhttp://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/in node.js asterisk-ari ?t..
i have mix of realtime and static configuration of pjsiphttps://pastebin.com/YVFwVsMDpjsip.conf[global]endpoint_identifier_order=username,ip,anonymous user_agent=ipbx… transport definition extconfig.conf[settings]ps_endpoints => odbc,configDb ps_au..
can you someone confirmhttps://issues.asterisk.org/jira/browse/ASTERISK-27065its easy to repl..