Brekeke JTAPI SDK 1.0 README Date: October 15, 2007 1. Obtain a product ID If you purchased a Brekeke JTAPI SDK runtime license, please obtain the product ID which was given by our sales representative. If you are using a JTAPI evaluation license, please obtain a product ID sent via an auto reply message when downloading the license. 2. Installation and License Activation Execute brekeke_jtapi_1_0_X_X.exe for installing Brekeke JTAPI SDK package. Default install directory (Install_dir) is My Documents\Brekeke\JTAPI1.0 Please enter your product ID during this installation process. It will attempt to activate your license automatically through the Internet. 3. Test JTAPI SDK using JTAPI sample program JTAPI sample programs are in Install_dir\sample directory. (1)Change the following part in each program. myprovider = peer.getProvider("BrProvider; pdir="); (Change \ to / in the path) (2)Compilation Specify the following path in the build class path and compile the sample program. Install_dir\work\jar\jtapi-spec.jar (JTAPI version 1.4) Install_dir\work\jar\jtapibase.jar (3)Setting Properties Please find the file "provider.properties" in Install_dir\work. Set SIP user name, auth user, password, SIP Proxy IP address, SIP listening port, total inbound and outbound concurrent sessions in the provider.properties. (4)Run the sample program Run the program putting the following into the Classpath. Install_dir\work\jar\jtapibase.jar Install_dir\work\jar\jtapi-spec.jar Install_dir\work\jar\log4j.jar Install_dir\work\jar\log4j-core.jar 5. Implementation of JTAPI 1.4 BREKEKE JTAPI SDK 1.0 implements some portions of JTAPI 1.4. All Classes and methods you can find in the sample programs will work. Please refer to the document in Install_dir\doc\jtapi_en.html . 6. Change History 1.0.8.0 (October 15, 2007) -Added a property setting "record_tail_cut_millisecond" for setting the cut length (milliseconds) from the tail of call recording. -Improved DTMF recognition -Added a license activation tool See Install_dir\doc\HowToActivateLicense.txt for the detail. -Fixed the bug that Refer-To header contained invalid URL. -Fixed minor bugs 1.0.7.0 (August 1, 2007) -Fixed a bug that Send DTMF didn't stop even when the call is disconnected -Fixed a bug that the result was returned OK even when sending DTMF failed. -Fixed minor bugs 1.0.6.0 (May 18, 2007) -Added Attended REFER transfer (see sample3) -Added Send DTMF method (See new sample6) -Added a property that you can set register expiration (ex. regist.local1.expires) -Added a method to obtain the error response code when call is failed. (see sample 1) -Supported to pass the sound from call destination to caller when the call is still ringing phase. (see sample 3) -Fixed minor bugs 1.0.5.5 (Aug 24, 2006) - Added the codec G.711alaw, iLBC support - Supports several timeout parameters (p_Duration, p_InitialTimeout, p_InterSigTimeout) for DTMF recognition (retrieveSignals). Refer to sample3 IncallDTMFtransfer.java for the usage. - Support an absolute path for play and record 1.0.5.3 (July 11, 2006) - Fixed the bug of Terminal.getAddresses() and Address.getTerminals(). 1.0.5.2 (June 23, 2006) - Fixed the problem that the whole system will be shutdown when Provider.shutdown() is issued. 1.0.5.1 (June 12, 2006) - Fixed the problem in the native BrPeer.dll 1.0.5.0 (May 5, 2006) - Added Call Conference feature - Enhanced Playing sound, Recording, DTMF features - Support REFER for call transfer 1.0.1.2 Beta (Apr 26, 2006) - Changed some file structures 1.0.1.0 Beta (Mar 23, 2006) - Added a property for setting a display name for each registering user - Fixed the bug that Terminal's listener is changed wrongly after Call Transfer is made - Changed the default RTP range from 10000-10999 to 31000-31999 1.0.0.9 Beta (Feb 22, 2006) - Added a property setting "usesameport_in_out". If you set usesameport_in_out=true, JTAPI will use the port set in the property "mysipport" for receiving and making calls. If usesameport_in_out=false or the property doesn't exist in provider.properties, JTAPI will use the port set in mysipport for receiving calls and use mysipport + 2 for making calls. 1.0.0.7 Beta (Feb 16, 2006) - Added a property setting "sipdomain"for using the domain name for SIP-URL for REGISTER & INVITE 1.0.0.6 Beta (Feb 13, 2006) - Supported SIP-URL with a domain name (such as sip:user1@domain for regist.localx.username property.) for REGISTER