Mizu Webphone - Mizu Voip

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2: system default3: manual (must be set by the httpproxyurl applet parameter –deprecated after version 3.5)4: autoDefault value is 4.httpproxyurl -deprecated(string)Used only for HTTP tunneling when usehttpproxy is set to 3 or 4.Set your http proxy address here. Example: 192.168.1.1:8080Default value is empty.This feature is removed after version 3.5 because it is not compatible with older JVM and using the browser capabilities offers better proxy handling.httpserveraddress(string)Useful when the transport parameter is set to 4 (auto) to specify the http tunneling gateway address.Default value is null (address loaded from the “serveraddress” applet parameter)transport(number)Transport protocol.0=UDP1=TCP2=TLS (beta)3=HTTP tunneling (both signaling and media. Supported only by mizu server or mizu tunnel)4=HTTP proxy connect5=Auto (automatic failowering from UDP to HTTP if needed)Default is 0.has_video(boolean)Enable/disable video features.You must download the phone video package for this to work and read its documentation for more details. The video module is provided “as is”. <strong>Mizu</strong>techcurrently doesn’t provide direct technical support for this functionality.dtmfmode(number)DTMF send method0=disabled1=sip INFO method2=RFC2833 in the RTP (if RTP stream is working, otherwise it will send also SIP INFO)3=both INFO and RFC28334=RFC2833 (will not send SIP INFO even if there is no RTP stream negotiated)Default is 2.voicemail(number)Subscribe to voicemail notifications (MWI). Accepted values:0=disabled1=display voicemail only if NOTIFY is received automatically without subscription2=auto-detect if voicemail SUBSCRIBE is needed3=subscribe for voicemail messages after successful registration4=subscribe for voicemail messages on startup
Default value is 2. Set to 3 if your server has support for MWI to be sure that the webphone will check the voicemail.voicemailnum(String)Specify the voicemail address. Most servers will automatically send the voicemail access number so you don’t need to set this parameter.transfertype(number)-1=default transfer type (same as 6)0=call transfer is disabled1=transfer immediately and disconnect with the A user when the Transf button is pressed and the number entered (unattended transfer)2=transfer the call only when the second party is disconnected (attended transfer)3=transfer the call when the webphone is disconnected from the second party (attended transfer)4=transfer the call when any party is disconnected except when the original caller was initiated the disconnect (attended transfer)5=transfer the call when the webphone is disconnected from the second party. Put the caller on hold during the call transfer (standard attended transfer)6=transfer the call immediately with hold and watch for notifications (unattended transfer)Default is -1 (which is the same as 6)If you have any incompatibility issue, then set to 1 (unattended is the simplest way to transfer a call and all sip server and device should support it correctly)transfwithreplace(number)Specify if replace should be used with transfer so the old call (dialog) is not disconnected but just replaced.This way the A party is never disconnected, just the called party is changed. The A party must be able to handle the replace header for this.-1=auto0=no1=yesDefault is -1allowreplace(int)Allow incoming replace requests.0=no1=yes and always disconnect old ep2=yes and don't disconnect if in transfer3=yes but never disconnect old epDefault is 2.discontransfer(number)Specify if line should disconnect after transfer0=no1=on refer sent2=on refer received3=on both4=all (default)discontransfer(number)Specify if line should disconnect after transfer-1=auto0=never1= on C party connected status2= on timeout3= on connected or timeout4= on ok for referDefault is -1
Page 1 and 2: 2014Mizu WebSIPPhoneA java applet i
Page 3 and 4: AboutThe Mizu WebSIPPhone is a SIP
Page 5 and 6: Demo versionWe are providing a demo
Page 7 and 8: DeploymentAlthough the webphone is
Page 9 and 10: Click to call / VOIP enabled forums
Page 11 and 12: Main ParametersThe parameters can b
Page 13 and 14: The following applet parameters are
Page 15 and 16: hideusernamepwdinput(boolean)Set to
Page 17: 1: change status to ring2: start me
Page 21 and 22: audiodeviceout(string)Audio device
Page 23 and 24: 2=yes3: yes with extra (this might
Page 25 and 26: (number)G.729 codec setting. 0=neve
Page 27 and 28: 3= yes4=alwaysprack(boolean)Enable
Page 29 and 30: Set to 0 to disable call cut off on
Page 31 and 32: 0=no (default)1=local (in the user
Page 33 and 34: Text logs are sent to the following
Page 35 and 36: Applet parameter securityGeneral co
Page 37 and 38: }public String encrypt(String str)
Page 39 and 40: Will connect to the SIP server. Thi
Page 41 and 42: ‣ -1: unknown‣ 0: no‣ 1: not
Page 43 and 44: string API_GetVersion()Return the p
Page 45 and 46: You will usually have to display th
Page 47 and 48: webphoneobj.API_SetParameter("passw
Page 49 and 50: Version 2.4Version 2.6-improvement:
Page 51 and 52: Version 4.0-fix: codec change on re
Page 53 and 54: Do I need to install java on my web
Page 55 and 56: Select UDP (and not TCP or TLS) for
Page 57 and 58: If the java (JVM or the browser) is
Page 59 and 60: 1. speex wideband (enabled low prio
Page 61 and 62: wp_tel.serveraddress = ''; // yours
Page 63 and 64: theApplet = theApplet.getSubApplet(
Page 65: 9. import Cross Root CA: -keytool -

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