[asterisk-app-dev] Testing an ARI application using SIPp

[Tickling Contest]

Hello,

I have an ARI application that I would like to load test using SIPp (
http://sipp.sourceforge.net/).

I understand how to REGISTER and send an INVITE out to a callee, but it is
not clear to me:

(a) how to choose a RANDOM port for each client I launch using say
http://tomeko.net/other/sipp/sipp_cheatsheet.php?lang=pl
(REGISTER_INVITE_client.xml). Should I generate this beforehand and pass it
as field3, say? Or is there a method already available in SIPp to do this?
(b)  How do I receive a call into a SIPp client? In other words, how do I
simulate a SIPp client to REGISTER and then wait for a call (from a SIPp
client connected to the same PBX)?
(c) Can I play an audio file (or an mp3) after the call starts from both
the caller and callee?

Your help is received with thanks!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-app-dev/attachments/20160302/ed136374/attachment.html>