by Mizutech. We can also provide our - WebPhone
Table of contents
- About
- Usage
-
Features, technology and licensing
- Feature list
- Requirements
- Technical details
-
Version history
- Version 0.2 (February 12, 2015)
- Version 0.6 (July 3, 2015)
- Version 0.9 (September 9, 2015)
- Version 1.0 (November 6, 2015)
- Version 1.2 (January 18, 2016)
- Version 1.3 (February 05, 2016)
- Version 1.4 (April 11, 2016)
- Version 1.5 (April 27, 2016)
- Version 1.6 (July 1, 2016)
- Version 1.8 (November 28, 2016)
- Version 1.9 (December 24, 2016)
- Version 2.0 (February 22, 2017)
- Licensing
- Integration and customization
-
Parameters
- SIP account settings
-
Engine related settings
- engine priority
- webrtcserveraddress
- rtmpserveraddress
- stunserveraddress
- turnserveraddress
- turnparameters
- turnusername
- turnpassword
- icetimeout
- autodetectwebrtc
- offersoftphone
- android_nativedialerurl
- ios_nativedialerurl
- accessnumber
- callbacknumber
- useragent
- customsipheader
- checkmicrophone
- transport
- localip
- signalingport
- rtpport
- sendrtponmuted
- mediaencryption
- dtmfmode
- playdtmfsound
- earlymedia
- prefcodec
- codec
- vcodec
- video
- video_bandwidth
- video size parameters
- codecframecount
- aec
- agc
- jittersize
- enablepresence2
- autostart
- loglevel
- logtoconsole
- NS and Java extra settings
-
Call divert and other settings
- normalizenumber
- techprefix
- numpxrewrite
- blacklist
- callforwardonbusy
- callforwardonnoanswer
- callforwardalways
- calltransferalways
- autoignore
- autoaccept
- acceptcall_onsharedevice
- beeponconnect
- redialonfail
- rejectonbusy
- disablesamecall
- allowcallredirect
- muteholdalllines
- automute
- autohold
- transfertype
- transfwithreplace
- changesptoring
- ringtimeout
- calltimeout
- mediatimeout
- bargeinheader
- voicerecupload
-
User interface related settings
- brandname
- companyname
- logo
- colortheme
- language
- featureset
- showserverinput
- useloginpage
- chatsms
- showincomingchatas
- conferencerooms
- callparknumber
- hasringcounter
- hasfiletransfer
- filetransferurl
- displaynotification
- displayvolumecontrols
- displayaudiodevice
- displaypeerdetails
- savetocontacts
- hasincomingcallpopup
- header
- footer
- version
- messagepopup
- showsynccontactsmenu
- defcontacts
- disableoptions
- hidesettings
- extraoption
- logsendto
- links
- Parameter security
-
JavaScript API
- About
- Basic example
-
Functions
- setparameter (param, value)
- getparameter (param)
- start()
- register ()
- unregister ()
- call (number)
- videocall (number)
- hangup ()
- accept ()
- reject ()
- ignore ()
- forward (number)
- mute (state, direction)
- hold (state)
- transfer (number)
- conference (number, add)
- dtmf (msg)
- sendchat (number, msg)
- sendsms (number, msg, from)
- voicerecord (start, url)
- audiodevice()
- getaudiodevicelist(dev, callback)
- getaudiodevice(dev, callback)
- setaudiodevice(dev, devicename, immediate)
- getvolume(dev, callback)
- setvolume(dev, volume)
- setsipheader(header)
- getsipheader(header, callback)
- getsipmessage(dir, type, callback)
- getlastcalldetails (callback)
- setline (line)
- getline ()
- isregistered ()
- isincall ()
- ismuted ()
- isonhold ()
- isencrypted ()
- checkpresence (userlist)
- setpresencestatus (status)
- getenginename ()
- delsettings (level)
- jvoip(name, jargs)
- getlogs ()
- getstatus ()
-
Events
- onLoaded (callback)
- onStart (callback)
- onRegistered (callback)
- onUnRegistered (callback)
- onCallStateChange (callback)
- onChat (callback)
- onCdr (callback)
- onDisplay (callback)
- onLog (callback)
-
getEvents (callback)
-
Notifications
- STATUS,line,statustext,peername,localname,endpointtype
- PRESENCE,peername,presence
- CHAT,line,peername,text
- CHATCOMPOSING,line,peername,composing
- CHATREPORT,line,peername,status,text
- CDR,line,peername,caller,called,peeraddress,connecttime,duration,discparty
- START,what
- EVENT,TYPE,txt
- POPUP,txt
- ACTION,txt
- LOG,TYPE,txt
- VAD,parameters
- Other notifications
-
Notifications
-
FAQ
- How to get my own webphone?
- What about support?
- What I will receive once I have made the payment for the webphone?
- Can Mizutech do custom development if required?
- Should I have programmer skills to be able to use the webphone?
- What software/service do I need to be able to use/deploy the webphone?
- Server side and connectivity requirements
- Web server requirements
- Is it working with my VoIP server?
- I wish to use the webphone but I don't have a SIP server or service
- What are the main benefits?
- Usage examples
- Folders and file structure
- Does the webphone depends on Mizutech services?
- How to configure the webphone
- How to handle WebRTC?
- How to handle Flash?
- How to handle Java, Native and App engines?
- How to setup the webphone for Asterisk?
- What are the advantages over pure WebRTC solutions?
- Known limitations
- OS/browser related issues
- The webphone is not loading/starting
- Cant connect to SIP server
- Failed outgoing calls
- Calls are disconnecting
- Using the webphone on local LAN
- Using the webphone without internet connection
- Using the webphone in controlled environment
- Including the webphone to all your pages
- Multiple phones on the same page
- Load the webphone on demand
- How to keep the webphone call between page loads?
- Single webphone instance on multiple pages
- New parameters was set but the old settings was loaded
- How to prevent unwanted unload event
- RTP statistics
- NAT settings
- Server failover/fallback
- I have WebRTC related issues
- Media access or Media stream permission denied popup
- VoIP calls without microphone device
- I have call quality issues
- I have one way audio
- Audio device cannot be opened
- No ringback tone
- Chat is not working
- The webphone doesnt receive incoming calls
- What is the best codec?
- Optimized VoIP for callcenter
- P2P
- Register vs Login vs Credentials
- How to find out registration status
- Caller ID display
- How to catch incoming calls?
- New settings not applied
- How to upgrade to a newer version of the webphone?
- I got an upgrade for my feature/issue request, but nothings seems to be changed
- How to uninstall or (re)install the webphone service
- How to upgrade from the old java applet websipphone?
- Auto-provisioning
- How to translate?
- How to add a color theme?
- How to use the webphone via URL parameters?
- Click to call from email signature
- How to manage multiple lines?
- How can I set the engine to be used?
- What are the best settings?
- How to set the webphone parameters dynamically?
- How to get the logs?
- Resources