It seems to me like the freeswitch platform itself would be a good place to start. I haven't thoroughly thought this out, but maybe you could write a test library using mod_<language-of-your-choice> designed to do human-like things such as issuing dtmf tones, pausing, speaking, etc. You could even run test scripts using the event socket (api commands) and test the results by subscribing to related events. I'd love to hear about what you come up with. --Stephen On Tue, Apr 14, 2009 at 8:59 PM, Mike Fedyk <mfedyk@mikefedyk.com> wrote:

Hi all,

I'm looking for suggestions on which open source tools to use for creating (or extending if there is already a project for this) a sip test suite.

I have already heard of sipp, but I want to know what others are using and how they go about this before starting from scratch myself.

Some things I'd like to do: - Dialplan/ voice menu/provider/did testing: Call number, press 1, expect to receive call on another extension. (kinda like expect) - Load testing

Basically I want to be able to automate how a human may interact with my installation to reproduce bugs and make sure they don't come back. That way I can make sure my changes (wherever they may be in my stack, dialplan, freeswitch, openser/kamailio/opensips, etc.).

Any pointers and/or tips will be much appreciated.

_______________________________________________ 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