111

New Features

1) Added the SHD-30C-CT/PCI and SHD-60C-CT/PCI digital boards, which, designed on the basis of the SHD-30A-CT/PCI/SS7 and SHD-60A-CT/PCI/SS7 boards, support T1 ISDN and PCI-X slot;

 

2) Added the SHD-30B-CT/PCI/FJ digital board, which, designed on the basis of the SHD-30A-CT/PCI/FJ board, supports T1 ISDN monitoring, PCI-X slot and hardware-based G729A codec for recording;

 

3) Added the SHD-60B-CT/PCI/FJ digital board, which supports E1 ISDN and SS7 monitoring, T1 ISDN monitoring, PCI-X slot, A-Law, μ-Law, ADPCM codecs for DMA recording, and hardware-based G729A codec for recording.

 

4) Added a new configuration item CardType under the [BoardId=x] section for the digital trunk board in the configuration file ShConfig.ini. It is used to set the operating mode of a digital trunk board: E1 or T1.

 

5) Enable our fax boards to support the ECM mode.

New Features

1) Digital call recording board:

    a. Supports LG TelKom PBX.

    b. Supports the feature that the corresponding monitored channel will turn to the idle state if the ’Hold?button is pressed in the use of Panasonic TDA series PBX.

    c. Supports Mitel Sx2000 PBX.

2) SS1 signaling:

    a. The event E_RCV_R2 (with lParam representing the channel number and wParam set to 0) will be returned when the R2 signal disappears.

    b. Adds two new functions in SHP_A3.dll: SsmSetSendCASFlag, which sets if the channel is allowed to send CAS signals in the connection; and SsmGetSendCASFlag, which obtains the flag that tells if the channel is allowed to send CAS signals.

    c. Adds a new function in SHP_A3.dll: SsmSendR2Ex, which generates an R2 signal lasting for a specified period of time on the channel. This signal won’t be interrupted until the function SsmStopSendR2 is invoked.

3) SS7 signaling:

    a. Adds a new function in SHP_A3.dll: SsmSendSs7MsuEx, which packs the TUP and ISUP message headers according to the channel number (ch) and leaves the message content to be configured by the application itself.

    b. Enables the use of the function SsmSetIsupParameter to configure the ANM message.

    c. Adds a new configuration item Ss7CircuitMap to [BoardId=x] in the configuration file ShConfig.ini, which is used in cooperation with the configuration item Ss7SignalingTS to enable the feature that TS16 will stop being the voice path when TS1 serves as the signaling link.

    d. 16E1 boards support any of the time slots to serve as the signaling link./

4) ISDN:

    a. Adds a new configuration item TransferCapability to [ISDN] in the configuration file ShConfig.ini, which can be used to set ‘voice?or ?.1kHz audio?as the content of the TransferCapability field in the signaling message.

    b. Adds a new configuration item PresentNumber to [ISDN] in the configuration file ShConfig.ini, which can determine whether it’s allowed to display the calling party number in the signaling message.

    c. Supports the driver to receive ??as part of the ISDN calling party number.

    d. 16E1 and 8E1 digital trunk voice boards support ISDN protocol.

5) Audio transcoding:

    a. Adds a new function in ShPcmHandle.dll: fPcm_ALawConvertMp3, which converts the A-law formatted file to the MP3 formatted file.

    b. Adds a new function in ShPcmHandle.dll: fPcm_MemGSMToPcm16, which converts the GSM formatted data stored in the buffer to the 16 Bit PCM formatted data.

    c. Adds a new function in ShPcmHandle.dll: fPcm_ULawConvertGSM, which converts the μ-Law formatted file to the GSM formatted file.

6) Others:

    a. Adds two new functions in SHP_A3.dll: SsmLoadChIndexData, which loads voice segment data from the specified file to the memory which has been assigned to the corresponding channel; SsmFreeChIndexData, which unloads the voice segment data which have been loaded to the memory.

    b. Enables the function SsmRecordMemBlock to support GSM format.

    c. Sets DSP3WorkMode=2 in the section [boarded=x] of ShConfig.ini to support hardware-based GSM format for recording on the SHT-16B-CT/PCI/FAX board.

    d. Supports the output of the callback function based on the time interval set by the function SsmRecToFileA while using software-based GSM encoding format for recording.

    e. Adds a new function in SHP_A3.dll: SsmGetEventMode, which can be used to obtain the driver’s event output mode and parameters set by the application.

    f. Enables the USB box to support the FSK detection and transmission feature.

    g. Adds a new function in SHP_A3.dll: SsmTxFlashEx , which generates a flash signal on the designated channel (equivalent to a rapid clap on the hook switch) and sets the current channel state upon this flash. The flash signal duration is set by the parameter ‘time?

    h. Adds two new board models: SHD-120D-CT/PCI and SHD-240D-CT/PCI, which support PCI-X slot, SS7 and ISDN protocols, and recording/playing in DMA (Direct Memory Access) mode.

    i. Adds two new board models: SHD-120D-CT/PCI/EC and SHD-240D-CT/PCI/EC, each of which has the enhanced echo cancellation capability apart from all features of SHD-120D-CT/PCI or SHD-240D-CT/PCI.

    j. Enables the DTP Series boards to support the monitoring of India SS1 Signaling.

Troubleshooting

1) Digital call recording board

    a. Fixes the following bugs:

        - The self-defined callback function can’t be invoked in the DST event callback programming mode when EventInterfaceType=1.

        - The configuration item EventInterfaceType is invalid when the ‘default?button on the user interface of ShCtiConfig.exe is clicked.

        -There exist noises and the channel always stays in the disconnected state while the digital call recording board is working with ISDN(2B+D) PBX.

        - The CallerId and LCD information can’t be obtained while the digital call recording board is working with Ericsson MD110 PBX and the phone DCB213 used in it.

        - The calling and called party information are misplaced during an incoming call while the digital call recording board is working with Nortel M1SL100 PBX and the phones 3903 and 2616 used in it.

        - The channel state transition is inaccurate while the digital call recording board is working with NEC Aspila EX PBX and the phone IP1WW-12TXH used in it.

        - The callerId cannot be received while the digital call recording board is working with Nortel M1SL100 PBX and the phones 3904 and 2616 used in it,.

        - The monitoring channel goes abnormal after 10 minutes since the application program starts while the digital call recording board is working with NEC EDK21 PBX and the phone DTP-8D-1 used in it.

2) SS7 signaling

    a. Deletes ‘Synway?and other related information in Ss7Monitor.exe (i.e. SS7 server).

    b. Fixes the following bugs:

        - Ss7Monitor.exe aborts while it is parsing a wrong protocol.

        - An error occurs when the function SsmSetTxCallerId is invoked to set a callerId consisting of more than 20 digits.

        - When transmitting the CPG message, the function behavior to send some message parameters conflicts with the role played by the configuration item. This bug has been fixed by giving priority to the function.

3) Others

    a. Fixes the following bugs:

        - The configuration item AlwaysEnableRxDtmf is invalid to the SHD-60A-CT/PCI/FJ board.

        - Only 10 boards can be detected via the function GetTotalPciBoard in Shinitpci.dll. The maximum number of boards to be detected has been changed to 20.

        - DTMF signals cannot be received when the digital call recording board is operating based on LineSide protocol.

        - An error occurs in getting the return value of SsmGetRingEchoToneTime while it is involed to set different durations of ringback tones.

        - The system will prompt error upon the immediate call of SsmPlayMemBlock after the call of SsmStopPlayMemBlock.

        - The SETUP message in the ISDN log file and the debugging information about the function SsmSetTxOriginalCallerID which is output by Debugview are incomplete.

        - The channel fails to go to the idle state after a call based on Brazil SS1.

        - The function SsmGetConfChInfo sometimes gets the wrong channel speaking status.

        - Something is wrong in playing the MP3 encoded and formatted audio file which is smaller than 1S.

        - An error occurs once the function fPcm_AdpcmToMp3 which converts ADPCM to MP3 is invoked for the fifth time.

        - The DTMF-related default configuration values of the cPCI board is different from that of the PCI board.

        - The event E_Proc_AutoDial will be returned if there is no change in the autodial progress of the ISDN channel.

        - Although an A-Law formatted audio file larger than 7M can be converted successfully via the function fpcm_AlawConvertGC8, the size of the converted audio file is 56 bytes.

        - The program will crash when the SHT-16B-CT/PCI/MP3 is used to record hardware-based MP3 codec wav file.

        - ShCtiConfig.exe lacks the default value of the configuration item EventInterfaceType. This bug has been fixed with the default value set to 0.

        - An error occurs in the driver if the set number of channels is less than the actual number of channels on the SHT-8B-PCI or SHT-16B-CT/PCI board.

Release date: August 8, 2007

 

Ver1.2.12.1.0

1) New functions:

---- Supports the E1 ISDN board.

 

Fix the following bugs in the driver:

• 1. Error occurs when the fax-decoding board is initialized at the second time;
• 2. Error occurs in decoding read-only files by the fax-decoding board;
• 3. 100% CPU gets occupied as the fax-decoding board is decoding a file.

Troubleshooting

1) Updates the header file shpa3api.h under both directories ShCti4.7.3.1.1 demo-english\Api\Vc6.0 and ShCti\api\Vc6.0, fixing two bugs that E_RCV_DTR_RING was repeatedly defined and the VERSION structure was not described.

 

2) Updates the library file SHP_A3.lib under the directory ShCti4.7.3.1.1 demo-english\Api\Vc6.0.

New function

1)Digital Call Recording Board:

1. The digital call recording board supports:

• Samsung DCS-60i and the phone models DCS-S1BW and DCS-S2EW used in it; Samsung Compact 2 and the phone models DS2100B and IDS8-D used in it.
• Nortel NorStar, NorStar BCM and Meridian1.
• the following PBX and phone models from NEC: Electra Elite 192 (6 phone models used in it), NEAX 2400 IMS (11 phone models used in it), NEAX1000 (2 phone models used in it), Professional (6 phone models used in it), NEAX 2000 (11 phone models used in it).
• Ericsson ELU25.
• Alcatel 4200/4400.
• Avaya Definity 2W, Definity 4W.
• Siemens HiPath Hicom Series.

Note: The PBX models mentioned above support the message output and the state machine on D-channel.

2. The digital call recording board supports:

• ISDN BRI.
• The digital call recording board supports Nortel C11 and the phone model 2216 used in it.

Note: The PBX models mentioned above support the state machine.

3. The digital call recording board supports recording of GC8 (hardwared-based) data in Pingpong Buffer mode.
4. Added a new Board model: SHR-24DA-CT/PCI.

Note: For detailed information about PBX and phone models supported by the digital  call recording board, see the file DST Boards Supported PBX Models.

 

2)SS7 signaling:

1. Added a new function in SHP_A3.dll: SsmSetTupParameter, which is used to set the optional parameters in the outbound tup signaling message, and so far only gets effective to the original called number address nature indicator in IAI.
2. Added a new item DefaultIAM_OriginalCalleeParam (Original Called Number parameter) to the parameter nType in the function SsmSetIsupFlag so as to support the setting of the ISUP Original Called Numer parameter.
3. Added a new item C_ISUP_PARAM_USI (User Service Information parameter) to the parameter ucParamTypeCode in the function SsmSetIsupParameter so as to support the setting of the User Service Information parameter.
4. Added new functions in SHP_A3.dll: SsmIsupGetUsr and SsmIsupSendUsr, which support the receipt and transmission of USR messages (User-to-user message); correspondingly added the new event E_RCV_Ss7IsupUtuinf (triggered by the receipt of USR messages).
5. Added a new configuration item IgnorBlockInGra to [TUP] in the configuration file ShConfig.ini, giving the choice on whether to ignore the GRA message status indicator: 0-don't ignore, 1-ignore.
6. The driver has a new capability to process TRM (Traffic-restart-allowed message) in ISUP messages.
7. The function SsmGetRemotePCMBlockStatus is updated to support ISUP.
8. Added a new event E_CHG_Mtp2Status which corresponds to the function SsmGetMtp2Status.

 

3)ISDN?

1. Added a new configuration item LogMode to [ISDN] in the configuration file ShConfig.ini so as to support the feature of saving a log each day.
2. Added a new function in SHP_A3.dll: SsmGetUserCallerID, which supports ISDN to obtain the CallerID from the ‘User' field.

 

4)Others:

1. The driver supports the memory recording of GSM (software-based) data.
2. The driver supports both the file and memory recording of G729A (hardware-based) data on the digital call recording board, and the file playing of G729A (software-based) data on all board models.
3. Added three configuration items IsEchoQuickDetect, EchoOnTolerance, EchoOffTolerance to [SystemConfig] in the configuration file ShConfig.ini. When IsEchoQuickDetect is 0, the driver must detect the ringback tone according to the configuration; when it is 1, the driver is allowed to detect the ringback tone according to the error limit set by EchoOnTolerance and EchoOffTolerance. In this way, the efficiency in judging the ringback tone is greatly improved.
4. Set the configuration item mfcr2_Protocol under the [SS1Config] section in the configuration file ShConfig.ini to br, supporting Brazil SS1 signalling monitor.
5. Added a new function in ShPcmHandle.dll: fPcm_AdpcmToMp3, which supports transcoding from ADPCM to MP3.
6. Added a new function in SHP_A3.dll: SsmSetLogOutput, which is used to output a log where lists the application-used API functions and the driver-output events; added two configuration items SetEventRange and SetChRange under [DebugView] in the configuration item ShConfig.ini. SetEventRange is used to record the relative information about output events within the specified number range to the log while SetChRange to record the relative information about output channels within the specified number range.
7. Added a new function in bmputil.dll: fBmp_ValidateFaxFile, which is used to judge whether *.tif is a format supported by our driver.
8. In the ISDN configuration item under the ShCtiConfig.exe configuration program: added the option of ‘unknown type' to the number field, and the option of ‘unknown type' and a default number to the called-party number field.
9. Added the 16E1-8E1 link warning message to the function SsmGetPcmLinkStatus; added new configuration items AutomaticAisGeneration and AutomaticRaiGereration to [BoardId=x] in the configuration item ShConfig.ini, which are used to determine whether to enable the automatic warning feature or not.
10. Added a new function SsmGetDllVersion to SHP_A3.dll, which is used to acquire the version information of ‘shp_a3.dll'.
11. The ShCtiConfig.exe configuration program is able to set the ratio of DTMF frequency with high energy to that with low energy as well as the percentage of DTMF energy to overall energy.

 

Troubleshooting
Fix the following bugs in the driver: 

1)Digital call recording board?

1. Working with Nortel M1 and the phones M3904 and M3905 used in it, the digital call recording board can't obtain the complete message from the PBX and fails to get the caller ID in the ACD mode.
2. While the digital call recording board is working with Ericsson MD110 and the phone 4223 used in it, the driver error in anaylizing displayed messages causes that the caller ID and the callee ID can't be obtained as usual.
3. Working with AVAYA 8700 and the phone 2402 used in it, the digital call recording board can't obtain the caller ID in the ACD mode.
4. While the digital call recording board is working with Panasonic 1231 and the phone KXT7433 used in it, the channel is still in the ACTIVE state after the phone hangs up.
5. While the digital call recording board is working with Panasonic TDA600 and the phone KXT76XX used in it, the channel state transfers in an inaccurate way.
6. While the digital call recording board is working with Panasonic TDA100 and the phone KXT7633 used in it, occasionally the board can't detect the ringing tone.
7. While the digital call recording board is working with AVAYAG350 and the phone 2402 used in it, there exists some noise in the recorded voice file.
8. While the digital call recording board is working with Nortel C11 and the attendant console KXT7433 used in it, the channel is still in the ACTIVE state after the phone hangs up.

 

2)SS7 signaling:

1. While the board is working in the ISUP mode, the function call of SsmGetReleaseReason leads to pointer overflow.
2. While the board is working in the SS7 mode, an error occurs in configuring the point code of 14 bits.
3. In the TUP mode, the 16th digit of the caller ID can't be obtained.

 

3)Others:

1. The memory overflows when the voice playing by index finishes.
2. Transcoding from the ADPCM to ALAW goes in a low quality.
3. While the board is working in the Lineside mode, the driver will output useless messages once the VC debugging mode is used to start the application.
4. When the entry parameter bEnable of the function SsmSetEvent is set to -1, the driver can't work under the event mode.
5. The defaults of the ShCtiConfig.exe program can't apply to the Hi-Z parallel-connection board.
6. Some content which does not belong to the ISDN message may be obtained during the function call of SsmGetIsdnMsu.
7. If the ISDN system starts a call before receiving the progress indicator, the driver can't receive the DTMF digits.

1) New functions:
----Allowed to read information about all channels or a specified one and to delete channels in the console;
----Enable remote control over each channel's hangup, flash and dial-up behaviors as well as the deletion of channels;
----Support display of the callername on the IP phone.
----Support asterisk 1.2.12.1:
OS: Redhat 2.4.20-8, FC4 2.6.11-1, Centos2.6.9-34
Asterisk: Ver1.2.12.1
Driver: 4.7.1.7-for-asterisk
2) Fix the following bugs:
----Modify the method to transmit data from the board to Asterisk, preventing data loss in the call while using Asterisk.
----Modify the algorithm for the voice data received by the board from Asterisk, eliminating voice pause and echoes during the call.

Fix the following bugs:
1) When the file is full and a new one is used for data recording, the data which remain in buffer because of the limited file size are lost.
2) When data are recorded via memory, the callback function which should be output once the data in buffer reach the set bytes fails to be output.
3) When data are recorded into files, the size of the last file for data recording may differ from what is set by users.

New function
Support the SHV-240A-CT/cPCI voice-alteration board.

Troubleshooting
Fix the following bugs in the driver:
1) A USB channel can still hear voices from other participants after it exits the conference.
2) In a multi-board-multi-process system, stopping one of the applications that are respectively based on different boards after running for some time may cause system BSOD (blue screen of death) or restart.
----No events are returned when using the 'event callback' mode for programming;
----The application prompts 'invalid handle error' in case some channels are shielded by ShConfig.ini when using the 'polling' mode for programming;
----The driver fails to do resetting of fax channels at channel initialization.

Identical to Ver4.7.1.6 under windows
New voice board model support include:
1) SHD-240A-CT/cPCI
2) SHD-240S-CT/cPCI

1) New function
Support asterisk 1.2.9.1
2) Os
Redhat 2.4.20-8 , FC4 2.6.11-1, Centos2.6.9-34
3) Driver
4.7.1.4-for-asterisk or above
4) Troubleshooting
-Improve data transfer algorithm and voice quality
-Deadlock bug at system's uptime to improve system's stability
-Disable output of driver events, which are irrelative with asterisk software

New function
1) new function Rd_RecToMem() and Rd_PlayToMem() to support record/playback in memory mode
2) function Rd_StopRec() supports memory mode
3) function Rd_StopPlay() supports memory mode

Identical to Ver4.7.1.6 under windows
New voice board model support include:
Edition Upgrade Views
1) SHT-120A-CT/cPCI
2) SHT-2B/USB
3) SHT-4B/USB

1) New function
Support asterisk Ver 1.2.5
2) OS
Redhat 2.4.20-8
3) Driver
Asterisk driver: Ver1.2.5
ShCti Driver: Ver4.7.1.4-for-asterisk
4) Hardware requirement
SHT-8B/PCI, SHT-16B-CT/PCI.

New function
1) New configuration item "UsageMode" to set CAS to ChannelBank mode for digital.
trunk voice board, which is not available for 16E1 and high-impedance monitoring
voice board at present.
2) New function SsmSetLine0OutTo() to set playback source on channel 0 for USB
voice board.
3) Support for SHT-2B/USB, SHT-4B/USB voice boards.

1) New function
-FXS, FXO based inbound/outbound call
-Retrievel of CallerId in FSK/DTMF mode
-Music On Hold
-Call Transfer
-Threeway Calling
-VoiceMail
-Fax Transmit/receive
-Call Waiting
-Call Conference
-Codec support: A_law, U_law , IMA ADPCM.
2) Hardware requirement
SHT-8B/PCI, SHT-16B-CT/PCI
3) OS
Redhat 2.4.20-8
4) Driver
Asterisk: Ver1.0.7
ShCti Driver: 4.7.1.4-for-asterisk.

New function

1)New support SUSE 10.0(K2.6.13-15-default)
2)Support USB board (under FC4 and SUSE 10.0 only)

New function

• 4-channel serial data recording/playback
• Up to 10 boards per system

New function

• Identical to Ver4.6.0.1 under windows
• For digital tap voice board, the driver is identical to Ver4714 under windows
• New configuration item to set whether to save log information for ss7bg, ss7fg.
  [Monitor]
  ShowMSU=0
  AutoTranslate=1
  ShowSNT=0
  ShowSNM=0
  ShowStatus=0
• Support ultra-capacity station voice board