Interesting, but are you using SIP/RTP? I don't see the SIP and RTP ports in your options? I am guessing you are using FS's ESL instead? On Monday, July 17, 2017, 5:35:12 PM EDT, Stanislav Sinyagin <ssinyagin@gmail.com> wrote: I made this simple call generator:https://github.com/voxserv/freeswitch-perf-dialer On 17 Jul 2017 18:25, "robert mundkowsky" <rfmundkowsky@yahoo.com> wrote: Anyone have suggestions on free automated calling toolsfor testing a system using SIP and/or webRTC? Robert ______________________________ ______________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www. freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch. org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists. freeswitch.org http://lists.freeswitch.org/ mailman/listinfo/freeswitch- users UNSUBSCRIBE:http://lists. freeswitch.org/mailman/ options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org