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So you hear the remote party with a bad quality? What version of PhonerLite do you use? Are you using a special headset? Did you try a different sampling rate? How do you initiate calls? Do you use a special provider?

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You can create a profile in PhonerLite without Registrar/Proxy - just to make IP to IP calls. You should enable UPnP (if your router supports it) or maybe STUN. Then make a call to "904@mouselike.org". There is a special menu with some echo test and so on.

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I told that before. It is not implemented. You may use a service like http://www.mysipswitch.com/ or any other service. In Germany routers from AVM (FRITZ!Box) are very often use. Some of them have an integrated SIP Registrar, so you can register directly at this router and the router registers to the several providers.

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PhonerLite has some configuration for hotkeys. Any additional solution is not supported.

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It works well! Just use the above address to test!

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Some sound hardware works better using a different sampling rate. So I have an onboard sound device that has bad timing with 8000 Hz sampling rate. The same works very well with 11025 Hz.
If you are using a wideband codec you should use 16000 Hz to use the advantages of wideband.

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I don't know what you mean with "#s". This is a placeholder for any phone number?

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You need a public IP address. Therefore I told you to use a STUN server or use UPnP if your router supports it.

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This should be fixed in the latest beta version.

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Yes you can answer the second call. The first call is set to hold automatically - except you have enabled conferencing.

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All other doesn't work at all for you? Or you don't like them

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You should specify the device, else no headset support can be done.

Uninstalled PL and extracted beta 1.6.2

Account w no Proxy/register: public address in STUN server , UPnP NAT checked in Network 904@mouselike.org not crashing program, BUT it simply makes some beeping noises.  I hear no menu to test anything in detail.  Status at the bottom is flashing green|red.

I thoulght any SIP account would call other SIP numbers/addresses, regardless of other "phone" capabilities. Why the need for a special account to call a SIP address?

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Did you put in a real STUN server or not? You can try "stun.ekiga.net" or any other.

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Then you want to register somewhere but it fails.

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In theory you can send your normal used SIP provider a SIP-URI not addressing a destination within the providers domain. But unfortunately providers block that.
If you configure a SIP server in PhonerLite, all initial request are sent to this server. You can compare this with an HTTP proxy configured in your vrowser. You can put in a completely different URL but it is send to the proxy.
If you just configure a username and a domain/realm in PhonerLite (no Proxy/Registrar) for each "number" the destination is chosen by analyzing the resulting destination URI.

I collated some open questions from above:

Does answering a call on “another” line inadvertently switch the lines?

What does "Default" speakers mean?  It implies that it automatically selects the headset, IF available, and the built in mic and speakers otherwise.

Why is recording hardware selected separate?  What is good enough for recording is good enough for the operator/user.   

I only see 2 hotkey functions (answer and hang-up) and only the default key options worked.  On my keyboard it is a reach to make the combinations. Several other softphones use "Esc" to hang up.  It seems intuitive and bound to become a standard.

Voipdiscount requires dialing "00 country code phone-number".  "+1 phone-number" also works for calling within the US.  Can Phoner lite prepend +1 onto a 10 digit entry?

DTMF does work, but the keys don't.  Typed digits go into the destination # box for the next "dial out".  Can Phoner send DTMF from the keyboard too?

I just noticed Beta 1.63 and downloaded it. For the first time, dtmf worked from the keyboard; That's good news!  Incoming calls were not ringing the program, so I switched the "current account" to my DID account.  That worked, but one would have to anticipate incomming calls to answer them.  The intended/reported behaviour to ring through regardless of the "current account" seems an important requisite to entering two or more accounts.  ??

"If you have one active call and a new one is ringing in and you answer that new call, the first call goes on hold and you are speaking with the new one. As long you do no action on that second call, you keep connected with the first one."
I may have an active call, but probably just sitting idle between calling out on the other account.  Ending the second call ends the original call, I assume.   

It seems that one would choose headset or speakerphone and use either for both communication and recording.  It is not intuitive to me to use a different "pickups" for recording.

Ctrl PgUp and Ctrl PgDwn are connecting and disconnecting the line.  I have no success trying to set other keys for these functions, but they are better for me than the previous keys.

At least one of the other softphones registers "accounts" to specific "lines".  That doesn't make sense, but I mistakenly used "lines" in my question, when I ment "accounts".  So, with PhonerLite, I think one with a separate DID and call termination account would leave the DID on in the background to answer calls and switch accounts to make a call.  Incoming calls while engaged in a call on the other account will go straight to voicemail (undetected), right?  Then, after ending a dialed call, the account should be switched back to the DID to field incoming calls.  Would you consider allowing two simultaneous active accounts?  It seems this would be useful to many.  Doesn't BetaMax offer about the cheapest call termination in most countries, and don't people in most countries have access to free or very cheap DID numbers (cheaper even than BetaMax)?  Multiple call termination accounts to call different countries might still require the account switching, but it would be nice to have at least one "base" profile that did not require DID and call termination from the same provider.

I set "Ctrl + right" and "Ctrl + End" as the hotkeys and pressed ok.  They did not work, and on returning to the settings, I find that they reverted to "Ctrl + 6" and "Ctrl + 1". It is essentially requireing the number pad.

Recording:  I have a headset and mic in my laptop to choose from. Maybe others have and use a higher quality handheld mic.  Whichever one they choose for recording, why not transmit the same signal to the other party?  Why record the handheld mic and transmit another mic?  Regarding audio from the other party, it would go through the "record" circuit first (or in parallel?), so the choice of speakers doesn't matter, right?