Hi Michael and welcome to the forums! You raise some good questions that I hope I can match with some good answers... Let's see!
1. Your mobile uses the ActiveSync connection which is fine for certain tasks like downloading mail and browsing the internet. Unfortunately, it is completely unusable for more complex protocols like SIP and thus can't be used to make VoIP calls.
2. This sounds like a network issue. AGEphone Mobile reports a "... NAT" or "Blocked" message on startup if left to the default options. If you see this message, please let me know what it is. If you do not get the message, please go to "Menu - Settings - Network" and change "NAT Traversal" to "STUN / UPnP". If it is already set, please change the STUN server to one listed in
this wiki.
3. We are always open to suggestions in this matter and may offer different evaluation methods in the future.
4. There is no such document and instead I'll just list the few settings that influence the call quality here. For starters, you can try lowering the buffers under audio a bit which will take out much of the delay that many VoIP calls bring with them. If you want to go a bit farther, close AGEphone Mobile and open the "sipd.conf" under "My Documents / My Phone Booth". Under [General] you will find a string called "MediaTypes=3 0 8" where
3 = GSM-FR
0 = G.711u
8 = G.711a
101 = DTMF Setting (do not change!)
You can rearrange those numbers and if you take out the "3" AGEphone Mobile will always use G.711 (64 kbit/s) which - while using a higher bitrate - provides better audio quality than the GSM-FR (13 kbit/s) codec. Depending on your 3G connection you may or may not be able to use G.711 as you have to add 20 kbit/s overhead to the audio stream in both directions. WiFi (and the attached landline internet connections) should be able to provide the necessary upstream in nearly any case.