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On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka <cervajs2 at gmail.com> wrote: > i tested this > > # cat /etc/asterisk/extconfig.conf > [settings] > queue_log => sqlite3,cdrDb > > # cat /etc/asterisk/res_config_sqlite3.conf > [cdrDb] > dbfile = /var/lib/asterisk/realtime.sqlite3 > > sqlite3 /var/lib/asterisk/realtime...

hi, our customer reports problem when 2 agents answer the call in the same time faster operator (device) answer the call, but the second is showed up (on device) and call is without sound asterisk 13.9/app_queue with strategy ringall/operators via Local channel with sip device (chan_sip) do you have any tips/info before i will dig deep into logs/debug? checked google&issues.asterisk.org

hi, is it possible log cel/queue_log to sqlite? via odbc? any experience? marek

Hi All We have about 15 different asterisk boxes around the place and on my list has been automate deployment updates and keep a revision history. They are mostly not publicly accessible, and external SIP access is closely firewalled , so updates happen straight away when its something like heartbleed, but take a while to trust/test new releases. Our boxes are Ubuntu LTS - mostly 14.04 at

hi, do you have someone example of http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/ in node.js asterisk-ari ? thanks Marek

hi, i want use asterisk+pjsip as voip client with multiple registrations (perf testing) i'm using this example configuration for one account [308] type=registration outbound_auth=308 server_uri=sip:308 at example.com:5060 client_uri=sip:308 at example.com:5060 [308](auth-userpass) username=308 password=pass [308](aor-single-reg) contact=sip:example.com:5060 [308](endpoint-basic)

hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711

hi, i have strange problem with asterisk 13.15.0+pjsip bundled/centos 7/systemd start script we 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 ago today i have problems with stopping/crashing asterisk /var/log/asterisk/messages dont show any clues [May 9 12:10:52] WARNING[25762] pjproject: tsx0x7fbb28024088