[Freeswitch-users] free testing tool - automated calling (SIP and/or webRTC)?

[David Villasmil]

You might also want to look at

https://freeswitch.org/stash/users/davidcsi/repos/freeswitch/browse/src/mod/applications/mod_dialer/mod_dialer.c?at=refs%2Fheads%2Ffeature%2Fmod_dialer

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 at 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 at 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 at 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 at 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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