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:



On 17 Jul 2017 18:25, "robert mundkowsky" <rfmundkowsky@yahoo.com> wrote:

Anyone have suggestions on free automated calling tools for testing a system using SIP and/or webRTC?

 

Robert


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