Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[Freeswitch-users] Recommended tools for creating/extending a sip test suite?


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users
View previous topic :: View next topic  
Author Message
mfedyk at mikefedyk.com
Guest





PostPosted: Tue Apr 14, 2009 11:15 pm    Post subject: [Freeswitch-users] Recommended tools for creating/extending Reply with quote

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.
Back to top
stevecrozz at gmail.com
Guest





PostPosted: Wed Apr 15, 2009 12:48 am    Post subject: [Freeswitch-users] Recommended tools for creating/extending Reply with quote

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 (mfedyk@mikefedyk.com)> wrote:
Quote:
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 (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

Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services