You might also want to look at https://freeswitch.org/stash/users/davidcsi/repos/freeswitch/browse/src/mod/... It's a freeswitch module to generate outgoing calls and bridge them to wherever you want, application, conference, transfer, whatever. On Tue, Jul 18, 2017 at 3:44 PM Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
it's just a dialer that originates calls from a FreeSWITCH server via ESL. It's up to you to tune FreeSWITCH configuration for your needs (such as media transport, codecs, signaling, etc.)
You can use one instance of FS to send calls to another one that is under tests.
On Tue, Jul 18, 2017 at 12:30 AM, robert mundkowsky <rfmundkowsky@yahoo.com> wrote:
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 tools for testing a system using SIP and/or webRTC?
Robert
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