can you share your best practices for ARI reconnect when asterisk is restarted or when ari app is started before asterisk is fullybooted?we are using node.js + ari-client so we are thinking about these options:1) wait for AMI event FullyBooted2) w..
Author : Marek Cervenka
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before i fill bug in asterisk issue tracker, is there someone who is using chan_pjsip + transport tcp in production with endpoints behind ..
im evaluating performance of CentOS7i did tests on CentOS6 x86_64/distro kernel 2.6.32, asterisk 11.16.0 with 500calls (7sec alaw, simple dialplan, pass trough – sipp generators/asterisk receiver with answer/playback)scenario – sipp generators – aster..
is there somebody using systemd start script on fedora/CentOS7 + asterisk 13 in production?i have strange problem with high cpu usage when asterisk is started via systemdthanks for feedbackp.s. systemd script is not in vanilla asterisk. only in fed..
im fighting with webrtc for 14 days reporting my experience to minimize number of crazy asterisk usersi have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + chan_pjsip + secure websockets + secure audio + audio in both waysproblems fir..
is it possible simultaneously use chan_sip and chan_pjsip?if yes, can you recommend settingsim thinking about- chan_sip – for sip hardphones/softphones(sip udp 5060)- chan_pjsip – ..
Im facing strange problem:Aasterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked. Call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chro..
There is updated skills based routing patch for asterisk queue. Please test if you have time: https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%2..
is it possible to play queue periodic-announce without stopping agents ringing? actual situation is sequential – ring agents, play announce (for 15 sec), ring agents , … (i need to connect agent with caller asap when agent is free)is it possible w..
my webrtc calls ends after ~60seconds with res_rtp_asterisk.c: DTLS failure occurred on RTP instance 0xb6c02a94 due to reason sslv3 alert unexpected message, terminating. any ideas where can be problem? or howto debug this problem?asterisk13.4.0-..