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..
im using hangup handlers on Asterisk13with standard answered calls i have 1 CDR per callwith scenario call from voip->mobile, call rejected on mobile i have 2 CDRsi dont want the second CDRwithout hangup handlers i have 1 CDRdo you think its bug or ..
i have strange problem with asterisk 13.15.0+pjsip bundled/CentOS 7/systemd start scriptwe are using chan_pjsip only for webrtc endpoints . switched from sipml5 to jssip with upgrade to 13.15.0(from 13.9.0) few days agotoday i have problems with stopping/crash..
what kernel version are you using for asterisk?are you satisfied with distro kernel (CentOS 6 2.6.32, CentOS 7 3.10, …) ?are you using newer kernels from elrepo.org?which kernel features are most critical for Asterisk performance pattern?t..
i have similar problem to https://issues.asterisk.org/jira/browse/ASTERISK-25806do you know about some workarounds/patches for better scalability?t..
im trying get report about missed calls per agent. im using queue_log and RINGNOANSWER event but i found problem descr..
i have strange problem with no rtp packets from asterisk after dns query. see pcap belowCentOS6/asterisk 13.9 + chan_sip172.23.0.3 – asterisk172.23.5.1/2 – voip phonesany ideas/hints?1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PC..
our customer reports problem when 2 agents answer the call in the same timefaster operator (device) answer the call, but the second is showed up (on device) and call is without soundasterisk 13.9/app_queue with strategy ringall/operators via Local chan..