Is there any way to make DTMF tones? For example, to check my voice mail box I have to press a number, generating appropriate DTMF tone.

If it's not implemented yet - is there any plans to support that soon?

Thanks you in advance!

Hmm, interesting.

I connect to FreeSWITCH as SIP server. If I use CSipSimple (SIP client on Android smartphone) I hear DTMF tone all the time I press a button and FreeSWITCH accepts and interprets the tone properly. But when I try to click on a number in PhonerLite dial pad - there is no tone as a feedback (perhaps, by design?) and, most importantly, there is no any reaction on FreeSWITCH side. It looks like it accepts clicking on '#' though, ending the call...

When I click on a number (let say - '2') Debug window shows:
Info Request (Dial): 2
Info Confirm
I don't hear DTMF tone and SIP server did not react on that too.

Is it perhaps some mis-configuration on my side or what? 
But in PhonerLite I don't see any options related to DTMF though...

I'm using PhonerLite v.1.90 on WXP.

Thank you for the important hint. I've made a call to a different extension and I now can hear DTMF tone and see in the log:
Generate DTMF: 2
Facility (DTMF Send) Request (2)
Facility Confirm (DTMF)

While working with voice mail box in FreeSWITCH, I've discovered some strange problem. The easy way to check voicemail is to call my own extension. For example, if I'm registered with extension 1003 and, if I want to check my mail box, I just call 1003. That's how I do it with my cell phone, registered with FreeSWITCH. But when I try to call my own extension in PhonerLite, there is no any INVITE messages sent to SIP server. The local PhonerLite log shows:
INVITE sip:1003@leo.zoo SIP/2.0 (T)
INVITE sip:1003@leo.zoo SIP/2.0 (R)
SIP/2.0 486 Busy Here (T)
SIP/2.0 486 Busy Here (R)
ACK sip:1003@leo.zoo SIP/2.0 (T)
ACK sip:1003@leo.zoo SIP/2.0 (R)
And that's it...

It doesn't send requests to SIP server at all. As a result I can't call to myself, checking my personal voice mail box.

That helped! Thanks a lot again.

The setting was offered by default. Is there a case, when I'd need to set "Multicast DNS"?

One more and (I hope) simple question:
How to start multiple PhonerLite instances at a time. Let say I'd like to test a calling group (more then one phone connected to SIP server at the same time). I know, I may use a heavy virtual machine to run next instance of PhonerLite in it. But, I hope, there is a simple solution - changing something it configuration or using a command line option, may be? I know that PhonerLite listens on port 5060 and only one instance could use it. But, in many cases I don't need incoming SIP sessions on that port... So, if, by default, first instance of PhonerLite would use it and others don't - that could be fine in most cases. Or, even simpler, what about using a command line option to specify a new port number for that instance of PhonerLite...

Only to check my own voicemail

And thank you for the hint about supporting direct calls between different computers running PL in one LAN. It works!