PCBEST NETWORK Robust SIP ActiveX PreConfig Reference

Click to open the SIP phone with pre-configured phone number and HTML buttons working.
Click to open an invisible SIP phone with HTML buttons working.

Sample(Auto Answer):

download this sample.

<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=UTF-8">
<title>Webphone</title>

<SCRIPT LANGUAGE="JavaScript" FOR="window" EVENT="onLoad()">
<!--
GTSIPPhoneObj.SetConfigValue("gtsrv.sip.callcontrol.auto.answercall", "1");
GTSIPPhoneObj.Start();
-->
</SCRIPT>

</head>
<body bgcolor=#ffffff>

<OBJECT id=GTSIPPhoneObj style="WIDTH: 187px; HEIGHT: 300px"
CODEBASE="http://www.pcbest.net/webphone/GTSIPPhone.cab#version=1,0,7,4"
classid=clsid:77342465-7794-44A8-87FC-FB7487C49EE0 >
<PARAM NAME="_Version" VALUE="65536">
<PARAM NAME="_ExtentX" VALUE="9551">
<PARAM NAME="_ExtentY" VALUE="9181">
<PARAM NAME="_StockProps" VALUE="0">
</OBJECT>

</body>
</html>

The method SetConfigValue can be used to preset configuration value. 
Click to open ActiveX Reference Manual to know more about methods and events of ActiveX.
The tags can be:

The following tags are about Phone GUI

"gtphone.show.main.window"
If you want to show the phone. "1" = show the phone. "0" = hide the phone

"gtphone.background.color"
If you hide your phone, what kind of background color you want the phone be.

"gtphone.show.digit.buttons"
If you want to show the digital buttons on the phone.

"gtphone.show.options.button"
If you want to show options button  on the phone.

"gtphone.show.callhistory.button"
If you want to show call log button  on the phone.

"gtphone.show.dial.button"
If you want to show dial button  on the phone.

"gtphone.show.hungup.button"
If you want to show hungup button  on the phone.

"gtphone.show.hold.button"
If you want to show hungup button  on the phone.

"gtphone.show.transfer.button"
If you want to show hungup button  on the phone.

"gtphone.show.mic.and.speaker"
If you want to show microphone and speaker buttons on the phone.

"gtphone.show.line.buttons"
If you want to show phone line buttons on the phone.

"gtphone.show.audiorecord.button"
If you want to show audio recording button  on the phone.

"gtphone.show.vmail.button"
If you want to show voice mailbox button  on the phone.

"gtphone.show.acf.button"
If you want to show advanced call feature button  on the phone.

"gtphone.show.nat.warning"
If showing the warning popup box when startup for firewalled or blocked network.

The following tags are about SIP account

"gtsrv.sip.reg.client.num"
How many SIP accounts the phone will register. It is "1" usually.

"gtsrv.sip.reg1.displayname"
The display name of first account. Sample: "Bob Wall"

"gtsrv.sip.reg1.username"
The user name of first account. Sample: "12345678"

"gtsrv.sip.reg1.domain"
The domain of first account. Sample: "pcbest.net"

"gtsrv.sip.reg1.proxy"
The proxy of first account. It is same as domain usually. Sample: "pcbest.net"

"gtsrv.sip.reg1.authorization"
The authorization code of first account. It is same as username usually. Sample: "12345678"

"gtsrv.sip.reg1.password"
The password of first account.

"gtsrv.sip.reg1.expire"
The expire time to be registered on the SIP server, in seconds. Defaultly it is 3600, which is one hour.

"gtsrv.sip.reg1.register"
If this tag is set to "0", this account will not be registered on the SIP server to take incoming calls, but the phone can still use this account to make outbound calls. Defaultly it is "1".

The following tags are SIP phone parameters

"gtsrv.sip.ip.port"
The local port number of SIP. It can be any value from 1024 to 65535. Statndard port is 5060. The default value for ActiveX phone is 7720. 

"gtsrv.sip.rtpstartrange" and "gtsrv.sip.rtpendrange"
The local RTP port range.

"gtsrv.sip.use.nat.addr"
Set it to "1" if the SIP proxy server is in local network.

"gtsrv.sip.stun.server"
The stun server to be used to discover global IP address.

"gtsrv.sip.prefered.codec"
The order and prefered codecs.
        18 - G.729a/b, not supported yet because of patent. If you want this codec in your phone, please contact us.
        102 - Speex.
        104 - iLBC 30ms
        103 - iLBC 20ms
        3 - GSM
        98 - G.726-32
        0 - G.711 mulaw
        8 - G.711 alaw

        Sample that only have mulaw and alaw: "0,8".

"gtsrv.sip.callcontrol.auto.answercall"
If it answers incoming calls automatically.

"gtsrv.sip.callcontrol.auto.transfercall"
If trasnfering is allowed.

"gtsrv.sip.dxsound.device"
The key word of your sound capture and playback device. You don't need to set it usually if the default windows playback and capture device is the sound hardware your want to use with ActiveX phone.

"gtphone.audio.record.enabled"
If enable audio-record. "1" = enable audio-record. "0" = no audio-record

"gtphone.audio.record.rootdir"
Root folder for audio-record.

"gtphone.moh.enabled"
If enable music on hold. "1" = enable. "0" = no music-on-hold

"gtphone.moh.folder"
Root folder for music-on-hold music files. Music files must be wav files with 8K Mono 16Bit PCM, or 8Bit Mulaw, or 8Bit Alaw format.

"gtsrv.sip.vad.enabled"
If enable VAD(Voice Activity Detection). "1" = enable. "0" = no

"gtsrv.sip.dtmf.method"
Method for DTMF detection and generation. "0" = In audio. "1" = SIP INFO. "2" = "RTP(RFC2833)". "3" = "Auto(RFC2833 or in audio)"

"gtphone.preset.phonenum"
Set your pre-configured number here.

"gtphone.cleanup"
If clean up Windows registry. "1" = YES, "0" = no

"gtsrv.licence.key"
Preset your licence key here. Note: It is not safe to explicitly put your licence key to web javascript or vbscript code.

"gtphone.show.nat.warning"
If show the warning "You may not be able to make Internet calls because you have a blocked or firewalled network......".
1 = showing the warning
0 = do not show
Default value is 0.

"gtphone.play.local.ring.sound"
If play local ring tone when a new call is coming in.
1 = yes
0 = no
Default value is 1.

"gtphone.play.remote.ring.sound"
If play remote ring tone when an outgoing call is confirmed that remote is ringing.
1 = yes
0 = no
Default value is 1.

"gtphone.play.busy.sound"
If play busy tone when an outgoing call is confirmed that remote is busy.
1 = yes
0 = no
Default value is 1.