HelloI get the following error when using our Lets Encrypt ssl certificate for webRTC calls :[Jun2 14:29:28] == DTLS ECDH initialized (secp256r1), faster PFS enabled[Jun2 14:29:28] ERROR[27360][C-00000ae5]: res_rtp_asterisk.c:1441 ast_rtp_dtls_set_configurati..
Author : Jonas Kellens
Home » Articles posted by Jonas Kellens (paged 2)
Hellousing Asterisk 1.8.32.3.What is the best way of knowing a call is being transfered (attended and unattended) ? And also knowing whereto (sip user) the call is being transfered and who is the transferer ?So I can log this information.Kind ..
Hellousing asterisk 1.8.32.3I am not able to make a call with video support. I do not know what I am missing to make this video call.Codec h264 should be supported.sip*CLI> core show codecs Disclaimer: this command is for informational purposes on..
Hellohow can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ?None of the below Dial() command give the correct result :exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762@myprovider.biz)exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762@myprovider.biz/${EXTEN})ex..
Hellousing Asterisk 1.8.32.3Current music on hold :myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_cof..
Hellowhen using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members.Example 1 :[Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15] Queue(SIP/incoming-00000246, myqueue1,,,,300,..
HelloIm a bit confused on how to group agents (give agents a group number) when using realtime queues.I read on the wiki :* If you include groups in your queue definition the calls get routedin the order of the group regardless of the specified strate..
HelloI keep getting the following error when trying to connect to the Asterisk server using AMI :$socket = fsockopen(tls://11.22.33.44,5039, $errno, $errstr, 5);Erorr on CLI :[Oct 26 14:38:19] ERROR[2992]: tcptls.c:609 handle_tcptls_connection: Prob..
HelloI am experiencing a freeze of the Asterisk proces when issuing a sip reload.I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3.I do not have this on versions certified-13.8-cert2, certified-13.8-ce..
Helloa call goes out and is answered :[Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108[Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-00000..