Download the latest beta version of PhonerLite and run it once. After closing PhonerLite you will have a new entry in the file called "PhonerLite.ini". Change the line "AutoRestore=true" into "AutoRestore=false" and save the file.

There is no hangup option available. You can use the COM interface to do so.
I don't offer a donate button because of all the following stuff like taxes. There is less trouble if I keep everything as it is. But thanks for your comments.

PhonerLite can call URIs directly.
Do you have configured a SIP server (Proxy/Registrar) in PhonerLite? If so, all requests go directly to this server.
You can enter such server in the "Domain/Realm" field instead. That has the following advantage: If you dial a phone number this domain is attached (@domain) and the message is sent to this domain. But if you enter a full domain, the call is sent to that entered domain.

As I told before: Cut the data from the "Registrar/Proxy" field and paste it into "Domain/Realm". The "Registrar/Proxy" field should be empty after this. Save.
After that you can enter a valid URI or just a telephone number.

some my tips and questions:

1) Cut and Paste (Shortcut) in Debug Windows not work

2) i send to my provider this

m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 112 113 114 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:112 G726-16/8000
a=rtpmap:113 G726-24/8000
a=rtpmap:114 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv


provider send back this and use iLBC codec 

t=0 0
m=audio 16406 RTP/AVP 97 3 8 101
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

I think, this is not bug, but feature (provider ignore my preference PCMA and set self preference No way... (if i not disable iLBC and GSM codec) - sure?

And how i get G.729a codec? is any chance get work sjg932.dll etc.?

3) how i set permanent record as profile settings - is possible please?

4) <Unauthorized> = after i click to green semaphore - this is unregistered, not incoming call? I think not, then how effect?

5) i report badlink in HELP: The actual implementation is tested against Grandstream GXP2000 

6) G.726 in accordance with RFC 3551 is same as G.726 AAL2?

7) how much numbers is possible registered simultaneously? (Fritz!Box = 20 Sip accounts together

many Thx.

onee wrote on 05. Jun 2010 at 06:13:


Dear admin,

please answer

Thx for reply.

But if i permanently registered in my TSP (for outgoing and incoming calls from PSTN), then how i call direct SIP URI???

Is normall, if i call 1234@mydevice.com:5060, then call goes to mydevice.com:5060 and NOT goes to my TSP!

1&2 - but if i call different number from my laptop, then all works fine. Then please, try to call this testing number (sip uri) :

200@hd-telefonie.avm.de
or
500@hd-telefonie.avm.de

and tell me... (G722 enabled)


-----------------

Quote:

There is no legal way to use this codec without paying license fees. Sorry.


---- please make usable these example for personaly testing 
www.voiceage.com/media/G729.zip
not include in distribution, if i download extra from here, thne is legal, i use only for my tests... you not distribute, then is all OK

Quote:

There are some legal issues because I don't do that. In some countries recording without telling the other party about it - is not allowed. 

---- but this is issue on user of application, not author of aplication - you.
Know you any callrecorders for Nokia Phones (Symbian S60) and Beep Supression function? This is commercial aplication and not problem for creators... In my country (EU) is 100% legal recording for personal purposes, i consult this with government.

Maybe if you play aqny vaw file, i delete this beeping vaw file and supress beep