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Page 1: Version 2 -  · PDF fileringback tone when calling out? ... FAQ Manual (Version 2.7) ... 4. Do VoIP boards support the access to IMS devices?

Synway Archives

Version 2.7

Synway Information Engineering Co., Ltd

www.synway.net

Page 2: Version 2 -  · PDF fileringback tone when calling out? ... FAQ Manual (Version 2.7) ... 4. Do VoIP boards support the access to IMS devices?

Synway Information Engineering Co., Ltd

Contents

Contents ............................................................................................................ i

Copyright Declaration .................................................................................. viii

Revision History ............................................................................................. ix

Chapter 1 Driver & Board Installation........................................................ 1

1. How to check if a Synway voice board and the driver are successfully installed?........................................................................................................................ 1 2. Which operating systems does the driver support?....................................... 1 3. Board interrupt cannot be detected at board initialization. Why?................ 1 4. Certain types of board cannot be found at initialization. Why?.................... 1 5. The board cannot be found in the manage device after you insert and install it. Why?............................................................................................................... 2 6. How to check the version of the driver installed on the computer? ............ 2 7. What to do if the UAC dialog appears when you install or uninstall the driver under WINDOWS 7?........................................................................................ 4 8. The following prompts appear when the board is running: "The device driver for the 'SHT-XXX-CT/PCI' device is preventing the machine from entering standby. Please close all applications and try again". Why?............... 4 9. Why must the board be grounded? How to ground it?.................................. 4 10. What is probably the cause for failure to detect boards, for appearance of aberrant voices or other unexpected problems?............................................... 5

Chapter 2 Product Specifications .............................................................. 8

1. Which version of PCI bus, PCIe bus and TDM bus should you use? ........ 8 2. How many boards and channels can be supported by a single-computer system?.......................................................................................................................... 8

Chapter 3 SHT Series Boards..................................................................... 9

1. Does the analog voice board come with an audio jack for direct playback through a headset or sound speaker?..................................................................... 9 2. What purpose does the external power supply on the analog board serve? .................................................................................................................................. 9 Analog Trunk Channel: ............................................................................................. 10 3. What is the difference between DTMF and FSK CallerIDs (calling-party numbers)? ................................................................................................................... 10 4. Why could the trunk or recording channel on the analog board not receive CallerID?...................................................................................................................... 10 5. Why do two adjacent trunk modules sometimes receive the same CallerID at the same time? .......................................................................................................11 6. Why does a channel repeatedly receive the same CallerID? .....................11

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7. Why does the record or trunk channel receive gibberish when trying to retrieve CallerID? ........................................................................................................11 8. Why do the trunk channels sometimes appear engaged, without response, or crossed lines between two adjacent channels (i.e. both lines appear to have received a call when in fact only one has)?...........................................................11 9. How to properly configure tone analysis parameters for an analog board? .................................................................................................................................11 10. Why does the analog trunk channel sometimes undesirably pick up busy tones?........................................................................................................................... 13 11. Why does the application fail to invoke the function SsmAutoDial to dial a number from the trunk channel?........................................................................ 13 12. How does an analog trunk channel detect pickup from remote phone in a dial-up process? ..................................................................................................... 13 13. Why does the analog trunk channel sometimes fail to detect hangup from remote end?....................................................................................................... 13 14. How does the analog trunk module detect connection drop on a line? 13 15. Is it possible for driver to make a misjudgment when PBX plays prompt tones in a call from the analog trunk channel to a mobile phone?................... 14 16. Why does an analog trunk channel switch from "idle" to "ringing" only after two rings have been detected?...................................................................... 14 17. Why does an analog trunk channel take a longer time to determine a ringback tone when calling out?.............................................................................. 14 18. When a remote phone places a call to an analog trunk channel, the phone cannot hear anything played by the channel, or hears only a short section before it is cut off. Why?............................................................................. 14 19. What to do if a station on analog boards can hear echoes from itself? 14 20. Can multiple analog trunk channels share the same called party number? ............................................................................................................................ 15 Station Channel: ......................................................................................................... 15 21. What is the maximum length of a station phone line from the station module on an analog board?................................................................................... 15 22. Which phenomenon indicates the damage of the station module?....... 15 23. Can the external power supply for B-type boards be used for C-type boards, and vice versa? ........................................................................................... 15 24. When a station phone makes a call from a station module via an analog trunk channel, the volume heard at the remote phone sounds rather low. Why? ............................................................................................................................ 15 25. Two station channels and a trunk channel join the same conference. When one of the station channels plays sound with a station phone on-hook, other channels can still hear the sound played. Why? ....................................... 16 26. Why does a station phone hear howling when it makes a call from a station module connected through a trunk module? ........................................... 16 27. Why is there much noise when the station phone connected to an on-board station module is off-hook?..................................................................... 16

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Chapter 4 SHD Series Boards .................................................................. 17

1. What is frame-synchronization signal?........................................................... 17 2. What new features do C-/D-/E-type boards have, compared with the old A-/B-type boards?...................................................................................................... 17 3. What does it mean if the synchronization indicator is off or blinking?...... 17 4. What causes an SS1 call to be abnormal?.................................................... 18 5. What causes an ISDN call to be abnormal?.................................................. 18 6. What does the SS7 server program Ss7Monitor.exe do? Must we run it?.. ................................................................................................................................ 18 7. What causes the following issues when running SS7 TUP or ISUP protocol? ...................................................................................................................... 18 8. How to deal with call failure or PCM instability in connecting digital board to optical transceiver? ............................................................................................... 21 9. Why there is so much noise and why do calls sometimes cut off when using digital trunk boards? ....................................................................................... 21 10. How does a digital trunk voice board receive numbers with different prefixes? ...................................................................................................................... 21 11. Is it mandatory to set TS16 for signaling transfer on a digital board which supports SS7? ............................................................................................................ 22 12. How to deal with the echoes produced by the device connected to Synway digital boards?............................................................................................. 22 13. How to deal with link breakup on digital trunk boards and its induced call cutoff?........................................................................................................................... 22 14. What is probably the cause for appearance of alarm, PCM asynchronization or unusable channel state on the end of PBX when it connects to Synway E1 boards?............................................................................. 22 15. What is the problem and what to do if the impedance of digital board does not match that of PBX?................................................................................... 23 16. How to enable the Hide-A-Caller function for digital trunk boards? ...... 23 17. What problems may be caused if board clock is wrongly configured? How to set it properly? .............................................................................................. 24 18. When you use a DTP series board monitoring the ISDN E1 line, if such problems as "what you have recorded is nothing but silence", "a single call is divided and recorded into several voice files", or "several calls are recorded into a voice file" occur, what is the reason for them and how can you deal with them? ........................................................................................................................... 25

Chapter 5 SHN Series Boards .................................................................. 27

1. What VoIP protocols and voice transmission formats do Synway VoIP boards support? ......................................................................................................... 27 2. Which part of SIP protocol is supported by Synway VoIP boards at present? ................................................................................................................................ 27 3. Do VoIP boards support conferencing?.......................................................... 27 4. Do VoIP boards support the access to IMS devices?.................................. 27

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Synway Information Engineering Co., Ltd

5. Why are there a lot of echoes on the line when making an IP call? ......... 27 6. The IP board driver is loaded successfully but the channel state shows unusable. Why?.......................................................................................................... 27 7. Which purposes can VoIP boards be used for?............................................ 28 8. What methods do Synway VoIP boards support for DTMF transmission? What is the difference among them? ..................................................................... 28 9. What contributes to failure of initializing VoIP boards? ............................... 28 10. What causes call failure on IP channels?................................................... 29 11. What is the SIP log for? How to set its levels?.......................................... 29 12. What may contribute to failure of using VoIP boards for registration?.. 29 13. When a B-type VoIP board is configured with proper registration information to register a server on the public network, the channel state shows registration failed while the application is running, why? How to locate the exact reasons? ........................................................................................................... 30 14. What do you do if the DTMF digits received by SHN series boards fail to be passed over bus to the cooperating SHD/SHT series boards and consequently fail to be sent out by them?............................................................. 30 15. How to deal with such voice problems as poor voice effect, no voice pass, single voice pass, noise, etc, appearing in normal calls on a VoIP board? ............................................................................................................................ 31

Chapter 6 ATP Series Boards................................................................... 35

1. What should the input impedance on a recording module be? ................. 35 2. How to sample voices from a microphone or radio using a microphone module or recording module?.................................................................................. 35 3. A recording channel is monitoring a trunk phone in parallel, but the recording channel constantly appears to be off-hook even if the trunk phone hangs up. Why? ......................................................................................................... 35 4. How does voice-activated recording work? ................................................... 35 5. Does the recording module support polarity reversal detection? .............. 35 6. Can a trunk module be used as a recording module?................................. 35 7. Why does the recording system generate a 1K size recorded file? ......... 36 8. Where can I find FAQs about the ATP2400 recording server?.................. 36

Chapter 7 DST Series Boards................................................................... 37

1. When the digital recording board is connected in parallel to the digital phone, the channels stays disconnected or between idle and disconnected. Why? ............................................................................................................................ 37 2. How to debug the digital recording board for monitoring? What to do if the monitored PBX or phone models are not supported by Synway driver? ........ 37 3. What are the supported digital PBX and phone models?........................... 37 4. How to record one-way voice files while monitoring both parties of a call? ................................................................................................................................ 38

Chapter 8 SynIPR Series........................................................................... 39

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1. What protocols and RTP codecs do SynIPR support now?....................... 39 2. Can SynIPR work under Linux?....................................................................... 39 3. Can SynIPR run on virtual machines?............................................................ 39 4. Can SynIPR work under Win8 operating system?....................................... 39 5. What are the commonly used events to start/stop recording in SynIPR programming? ............................................................................................................ 39 6. Can SynIPR obtain callerID and calleeID?.................................................... 40 7. How to judge the call direction in SynIPR?.................................................... 40 8. What is the difference between the two monitoring modes "Monitor by Station" and "Monitor All"?........................................................................................ 40 9. Can a SynIPR channel be bound with a designated IP phone?................ 40 10. What to do to increase/decrease the amount of SynIPR channels or to prolong the time of authentication? ........................................................................ 40 11. Why does the application detect the event E_RCV_IPR_AUTH_OVERFLOW (authentication overflow)? How to deal with it? ............................................................................................................................ 41 12. Why does the recorded file come out silent?............................................. 41 13. Why does the recorded file only contain one party's voice?................... 41

Chapter 9 Fax Application ........................................................................ 42

1. How can WORD or PDF files be faxed? ........................................................ 42 2. What file formats does the fax board support? ............................................. 42 3. What contributes to failure of establishing a fax handshake or carrier wave? ................................................................................................................................ 42 4. How to append fax file during a fax transmission? ...................................... 42 5. Why does a received tif file sometimes show white text on dark background when opened with ACDSEE? ........................................................... 43 6. What are features of the Synway soft-faxing? .............................................. 43 7. Is it possible to convert tif files without doing it manually? ......................... 43 8. Can additional information such as company name be added to the fax documents in transmission? .................................................................................... 43 9. Can we choose to fax certain pages of the document only?...................... 43

Chapter 10 Voice CODEC Support ............................................................. 44

1. Which CODECs are supported for file recording/playback? ...................... 44 2. Which compression format does recording use, and at what rate?.......... 44 3. How to play WAV files recorded by a sound card with a Synway voice board? .......................................................................................................................... 44 4. Why are noises sometimes heard in the process of file playback? .......... 45 5. How to play or record GSM, MP3 voice files? .............................................. 45

Chapter 11 Conferencing and Bus Applications ...................................... 46

1. How to play background music in a teleconference? .................................. 46 2. Does teleconferencing occupy additional channels?................................... 46 3. How to record conversation of all conference members at one time? ..... 46

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4. Why does howling occur when operation is carried out on a bus or in a conference? ................................................................................................................ 46 5. How can an outsider listen to the entire conference conversation? ......... 46 6. Up to how many conference groups and participants are allowed in a conference? ................................................................................................................ 46 7. Sometimes when a few channels from a digital board join the conference as dynamic members, a speaker's voice cannot be heard by other members or is only heard after a delay. Why?....................................................................... 47 8. In a teleconference, how does the board prevent DTMF key tones in a channel from being heard by other channels?..................................................... 47

Chapter 12 About Linux.............................................................................. 48

1. How to deal with the failure of running test program at common user access to Linux? ........................................................................................................ 48 2. What to do if the driver reports "insmod shdpci.ko error" during its installation under Linux?........................................................................................... 48

Chapter 13 About Asterisk ......................................................................... 49

Chapter 14 About UMCT ............................................................................. 50

1. What to do if the switch shuts down with a blue screen during the runtime? ................................................................................................................................ 50 2. What to do if the UMCT switch can't find the HD upon startup? ............... 50 3. Why does the SSW080B UMCT keep alarming upon startup? ................. 51 4. What to do if the USB CD-ROM connected with UMCT cannot be started normally? ..................................................................................................................... 51 5. How to set the auto-start-upon-power-on feature for the UMCT series SCU03 boards?.......................................................................................................... 51 6. How to start the UMCT without keyboard? .................................................... 52 7. How to solve the problem of blank screen upon startup?........................... 52 8. What to do if all parameters (such as Time) of the UMCT are restored to factory settings after power-off and restart? ......................................................... 52 9. Not being used for a time, the SSW080A UMCT fails to start (the pwr indicator is on but the machine has no response after you press the start button) and the PC fails to detect the HD. Such problem disappears after you power off the machine, wait for a short time and then restart it. What is the cause?.......................................................................................................................... 53 10. What shall you pay attention to while using UMCT intelligent switches?. ............................................................................................................................ 53 11. When the UMCT switch is restarted after abnormal power-off, it can neither transmit nor receive data packets although the network card driver is good. How to solve the problem? ........................................................................... 54

Chapter 15 Issues about Application......................................................... 55

1. When debugging, the application is deliberately forced to end and succeeding re-initialization fails. Why?.................................................................. 55

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2. Do Synway boards support multi-thread programming?............................. 55 3. Can two applications supported by the same driver be run on one system? ................................................................................................................................ 55 4. When the application generates a smaller buffer for recording and uses a dual-buffer for playback, noise is heard in playback. Why? .............................. 55 5. Why does voice become intermittent during a playback with DTMF signals available on the line? ................................................................................................ 55 6. How to transfer an incoming call to an analog trunk channel to a call-agent? ................................................................................................................................ 56 7. How to send CallerID via a station channel to a phone that can receive CallerID?...................................................................................................................... 56 8. Which programming platforms does the driver work on? ........................... 56 9. What is the difference between the programming modes CALLBACK and EVENT_POLLING?................................................................................................... 56 10. Why does the application sometimes not detect the change in call or monitoring status?...................................................................................................... 56 11. What is the difference between CIC and Ch? ........................................... 57 12. The application program fails to invoke the function SsmRecToFile and the returned message says "Error creating file! ". Why?.................................... 57

Appendix A Technical/sales Support ........................................................... 59

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Copyright Declaration

All rights reserved; no part of this document may be reproduced or transmitted in any form or by any means, electronic or mechanical, without prior written permission from Synway Information Engineering Co., Ltd (hereinafter referred to as ‘Synway’).

Synway reserves all rights to modify this document without prior notice. Please contact Synway for the latest version of this document before placing an order.

Synway has made every effort to ensure the accuracy of this document but does not guarantee the absence of errors. Moreover, Synway assumes no responsibility in obtaining permission and authorization of any third party patent, copyright or product involved in relation to the use of this document.

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Revision History

Version Date Comments

Version 1.0 2008-10 Initial publication

Version 2.0 2008-12 New revision

Version 2.1 2010-3 New revision

Version 2.2 2011-3 New revision

Version 2.3 2011-10 New revision

Version 2.4 2012-10 New revision

Version 2.5 2014-1 New revision

Version 2.6 2014-11 New revision

Version 2.7 New revision 2015-5

Note: Please visit our website http://www.synway.net to obtain the latest version of this document.

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Chapter 1 Driver & Board Installation

1. How to check if a Synway voice board and the driver are

successfully installed?

The following 3 conditions are indicators that the hardware and driver are properly installed:

1) In ‘Device Manager’ of Windows operating system, under ‘Sound, video and game controllers’, check that all models of boards installed on the computer are present and without the ‘!’ mark before their names.

2) Run shconfig.exe under the installation directory (usually ‘/ShCti/’). Click on ‘Default’. The number of boards and their models in the list should correspond with those actually installed. Click on ‘Apply’. A message should appear on the top-left corner or a dialog should pop up to indicate that the current configuration is verified.

3) The test program – test.exe – can be started; the number and type of channels in the list should fit well with the actual situation.

2. Which operating systems does the driver support?

The driver supports Windows 2012, Windows 8, Windows 7, Windows vista, Windows 2000, Windows XP, and Linux operating systems.

3. Board interrupt cannot be detected at board

initialization. Why?

Possible causes:

A. Poor connection due to board not being properly inserted into the slot.

B. Incompatibility with the host (or mainboard)

C. Llinux system restart after the sleep mode as the driver will release the source during the sleep time.

4. Certain types of board cannot be found at initialization.

Why?

Possible causes:

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A. If the error message states a model that is different from the one installed, it could be that the ShConfig.ini file is not properly configured or the incorrect driver was installed.

B. If the error message states a model that is in fact already installed, it could be that:

① The voice board is not positioned properly in the slot.

② The slot is damaged.

③ The voice board itself could be faulty (check if obvious physical damages are present).

5. The board cannot be found in the manage device after you

insert and install it. Why?

Possible causes:

A. The voice board is not positioned properly in the slot or the slot is damaged.

B. Incompatibility with the host (or mainboard)

C. The voice board itself could be faulty (check if obvious physical damages are present).

D. The power supply is not appropriate for the board (See Q10 in Chapter 1 for the table ‘Power Supply for Each Board Model’).

6. How to check the version of the driver installed on the

computer?

1) Execute GetInfoTool.exe under the driver installation directory and you can see the current driver version on the “Driver Version” page shown as the figure below.

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2) Select [Device Manager]/[Sound, video and game controllers], double click a board in the list and view version information of current driver on the popped up dialog box ‘Board Properties’ shown as the figure below.

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3) Execute ShCticonfig.exe under the installation directory and you can see the current driver version on the title bar shown as the figure below.

7. What to do if the UAC dialog appears when you install or

uninstall the driver under WINDOWS 7?

User Account Control (UAC) is a technology and security infrastructure introduced with Microsoft's Windows Vista operating systems. It aims to improve the security of Microsoft Windows by limiting application software to standard user privileges until an administrator authorizes an increase or elevation. In this way, only applications trusted by the user may receive administrative privileges, and malware should be kept from compromising the operating system.

If the UAC dialog pops up when you install or uninstall the Synway driver under Windows 7, click ‘Yes’ and it will have no effect on the use of Synway boards. If you don’t want the UAC dialog to pop up, close it by the following steps:

Step 1: Click Start, and then click Control Panel.

Step 2: In Control Panel, click User Accounts.

Step 3: In the User Accounts tasks window, click Turn User Account Control on or off.

Step 4: Clear the Use User Account Control (UAC) to help protect your computer check box, and then click OK.

8. The following prompts appear when the board is running:

"The device driver for the 'SHT-XXX-CT/PCI' device is

preventing the machine from entering standby. Please close

all applications and try again". Why?

As the system is expected to run for long hours unmanned, ‘energy-saving’ mode should be turned off for both the CPU and the HD in CMOS or WINDOWS operating system. This is to ensure full-speed operation of the computer, or it may lead to a drop in performance or unexpected errors after running for some time.

9. Why must the board be grounded? How to ground it?

A chassis installed with voice boards must be grounded for safety reasons, according

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to standard industry requirements. No or improper grounding may cause instability in operation as well as decrease in lightning resistance. A simple way is earthing with the third pin on the plug. If the third pin of the plug cannot be grounded, choose a stubby conductor (such as the water pipe or the ground wire of the building) to connect the IPC chassis with the ground. To avoid a poor grounding, we suggest that you fix the conductor on the chassis with screws. Note that the resistance of the conductor itself should be less than 1Ω. Besides, the telephone wire is not recommended to work as the conductor.

10. What is probably the cause for failure to detect boards, for

appearance of aberrant voices or other unexpected

problems?

This is perhaps because the power supply is not appropriate for the board. Refer to the table below to find the proper power supply for each board model.

PC Power Supply Series Bus Board Physical Interface +5V +12V -12V +3.3V VIO

USB SHT-2B/USB SHT-4B/USB RJ11

SHT-8A/PCI (V2.0) RJ11 SHT-8B/PCI RJ11 SHT-16B-CT/PCI RJ45 SHT-16C-CT/PCI/FAX RJ45 PCI

ATP-24A/PCI ATP-24A/PCI+ RJ-21

ATP

PCIe ATP-24A/PCIe ATP-24A/PCIe+ RJ-21

USB SHT-2B/USB SHT-4B/USB RJ11

SHT-8B/PCI RJ11 SHT-16B-CT/PCI RJ45 SHT-8C/PCI/EC SHT-8C/PCI/FAX RJ11 PCI

SHT-16C-CT/PCI/EC SHT-16C-CT/PCI/FAX RJ45

SHT

PCIe SHT-16D-CT/PCIe RJ45 SHD-30A-CT/PCI/SS1 SHD-60A-CT/PCI/SS1 SHD-120A-CT/PCI/SS1 SHD-30B-CT/PCI/SS7/FAXSHD-60B-CT/PCI/SS7/FAX

RJ48C

SHD-30C-CT/PCI SHD-30C-CT/PCI/FAX SHD-60C-CT/PCI SHD-60C-CT/PCI/FAX

RJ48C

SHD-120D-CT/PCI SHD-120D-CT/PCI/EC SHD-120D-CT/PCI/FAX

RJ48C

SHD

PCI

SHD-240D-CT/PCI RJ48C/M

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SHD-240D-CT/PCI/EC SHD-30A-CT/cPCI/SS7 SHD-60A-CT/cPCI/SS7 SHD-120A-CT/cPCI/SS7 SHD-60B-CT/cPCI/FAX

RJ48C

cPCI SHD-240A-CT/cPCI SHD-480A-CT/cPCI SHD-240S-CT/cPCI SHD-480S-CT/cPCI

RJ-48T

SHD-30E-CT/PCIe SHD-30E-CT/PCIe/EC SHD-30E-CT/PCIe/FAX SHD-60E-CT/PCIe SHD-60E-CT/PCIe/EC SHD-60E-CT/PCIe/FAX SHD-120E-CT/PCIe SHD-120E-CT/PCIe/EC SHD-120E-CT/PCIe/FAX

RJ48C

PCIe

SHD-240E-CT/PCIe SHD-240E-CT/PCIe/EC SHD-240E-CT/PCIe/FAX

RJ48C/M

SHN-32A-CT/PCI

PCI SHN-60B-CT/PCI SHN-60B-CT/PCI+ SHN-120B-CT/PCI HN-120B-CT/PCI+

RJ-45

SHN

PCIe

SHN-8B-CT/PCIe+ SHN-16B-CT/PCIe+ SHN-32B-CT/PCIe+ SHN-60B-CT/PCIe+ SHN-120B-CT/PCIe+

RJ-45

SHD-30A-CT/PCI/FJ SHD-60A-CT/PCI/FJ RJ48C

SHD-30B-CT/PCI/FJ RJ48C

PCI SHD-60B-CT/PCI/FJ DTP-30C/PCI DTP-30C/PCI+ DTP-60C/PCI DTP-60C/PCI+ DTP-120C/PCI DTP-120C/PCI+

RJ48C

DTP

PCIe

DTP-30C/PCIe DTP-30C/PCIe+ DTP-60C/PCIe DTP-60C/PCIe+ DTP-120C/PCIe DTP-120C/PCIe+

RJ48C

SHR-16DA-CT/PCI RJ-45 SHR-24DA-CT/PCI PCI DST-24B/PCI DST-24B/PCI+

RJ-21 DST

PCIe DST-24B/PCIe DST-24B/PCIe+ RJ-21

FXM PCI FXM3200P FXM3201P FXM3210P FXM3211P

RJ11

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PCIe FXM3200E FXM3201E FXM3210E FXM3211E

RJ11

PCI TEJ100P TEJ101P TEJ200P TEJ201P

RJ48C

TEJ

PCIe TEJ100E TEJ101E TEJ200E TEJ201E

RJ48C

PCI BRI1610P BRI1611P RJ45

BRI PCIe BRI1610E

BRI1611E RJ45

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Chapter 2 Product Specifications

1. Which version of PCI bus, PCIe bus and TDM bus should

you use?

PCI for A-/B-type Boards: PCI 2.1, 32 bits, 33 MHz;

PCI for C-/D-type Boards: PCI 2.2, 32/64 bits, 33 MHz, compatible with PCI-X;

PCIe(X1) for E-type Boards: PCIe Express 1.0, compatible with X2, X4, X8 and X16;

TDM for PCI, PCIe Boards: H.100;

TDM for cPCI Boards: H.110.

2. How many boards and channels can be supported by a

single-computer system?

A single computer can accommodate more than 10 boards (can be of different models). A single computer with all analog boards gives up to ‘number of boards×16’ channels, one with all digital PCI boards gives up to ‘number of boards×240’ channels (for boards with 8 E1) and one with all digital cPCI boards gives up to ‘number of boards×480’ channels (for boards with 16 E1).

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Chapter 3 SHT Series Boards

1. Does the analog voice board come with an audio jack for

direct playback through a headset or sound speaker?

Yes. There is an audio jack which plays signals from Channel 0 through a connected headset or sound speaker.

2. What purpose does the external power supply on the analog

board serve?

The purpose is to provide feed power for station modules. The feed power for 16-/8-channel B-type boards is 48V, for Version 2.0 of 16-/8-channel B-type boards is 40V, for 16-/8-channel C-type boards is 40V, and for 16-channel D-type boards is 12V. Each B-/C-type board has an external power interface which is used for connecting the external feed power and the ringing current power. For 16-channel D-type boards, multiple power supply schemes are available to provide the 12V feed power: one is to connect the P4 D-type hard disk power plug of the computer, and another is to short the S17 pin header and obtain the power via the PCIe slot (see the figure below). When the power supply inside the computer is insufficient, a 12V power adapter externally connected can work as a substitute. For more information, refer to the corresponding hardware manual on our website.

We strongly advise users to buy the power supply from Synway, and not to use any power supply with uncertain voltage or polarity, or it might cause damage to the voice board.

D1 Indicator

S17 Pin Header

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Note: Ensure that polarity is correct and only establish the connection with power off.

Analog Trunk Channel:

3. What is the difference between DTMF and FSK CallerIDs

(calling-party numbers)?

DTMF CallerID is transmitted in DTMF mode before the first ring, whereas FSK CallerID is transmitted in FSK mode between the first and second rings.

4. Why could the trunk or recording channel on the analog

board not receive CallerID?

1) Input connection to board is broken.

2) Certain PBXs do not support CallerID transmission.

3) PBX side does not open incoming call display function to relative phone line.

4) PBX side did not send CallerID on this occasion.

5) Poor grounding may also cause a CallerID to be missed.

6) Incorrectly set receiving mode for CallerID (DTMF or FSK mode) in configuration file or application program.

7) When in FSK mode, the application could have tried to retrieve CallerID when the driver has detected the first ring or even earlier. It is suggested that the application retrieves CallerID only after the second ring is detected.

8) When there is a high volume of incoming calls at one time, it can happen that one caller calls the voice board and hangs up while the PBX is sending its CallerID to the voice board. If another caller now calls in before the previous PBX has completely sent CallerID of the first call, and the voice board picks up this second call, the result could be that the voice board receives no CallerID or that of the first call, or gibberish. We suggest that when expecting high call volumes, the voice board should be set to pick up calls immediately upon receipt of CallerID, and not wait for ring-detection in its receiving procedure.

9) When in DTMF mode, it could be due to:

A. Sensitivity for DTMF reception is too low. This can be adjusted in the configuration file.

B. Received CallerID is less than 4 digits, while the driver is configured to filter out numbers that are less than 4 DTMF digits. This case only occurs for SynCTI Ver. 5.3.0.1 and below versions. For versions above Ver. 5.3.0.1, the

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driver is configured to filter out numbers that are less than 1 digit.

C. CallerID is sent between the first and second rings but retrieved after detection of the send ring.

5. Why do two adjacent trunk modules sometimes receive the

same CallerID at the same time?

It is mostly due to permanent or transient short-circuiting between the two modules.

6. Why does a channel repeatedly receive the same CallerID?

The driver does not automatically clear the CallerID buffer. It is suggested that the application program is set to clear the buffer upon the retrieval of each CallerID, or the previous CallerID will be read.

7. Why does the record or trunk channel receive gibberish

when trying to retrieve CallerID?

This is often due to poor line quality or poor grounding.

8. Why do the trunk channels sometimes appear engaged,

without response, or crossed lines between two adjacent

channels (i.e. both lines appear to have received a call when

in fact only one has)?

1) Possible reasons for line-engagement or no-response on certain channels:

A. RJ11 jack might not be properly plugged in.

B. Outgoing phone line might have been short-circuited or faulty.

C. Modules might be loose, which resulted in bad connection

2) Line-crossing is usually a result of the 4 pins of the input RJ11 jack touching one another, thereby creating short-circuit among them.

9. How to properly configure tone analysis parameters for an

analog board?

Tone is a rhythmic audio signal usually used for indicating line status, such as

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calling/called party's hang-up, line-engagement, etc. A standard tone is often a single-sine wave signal of 450Hz. However, certain smaller PBXs or key telephone systems might use other frequencies or even dual-tones. The Synway driver platform endows each channel with 2 independent tone-detectors, allowing them to analyze 2 different tones at one time.

1) Correctly define frequency characteristics (i.e. center frequency and bandwidth) of the tone.

The Synway driver platform supports detection and analysis of single and dual tones of any frequency. Single tone mode requires the predetermination of a center frequency and a bandwidth. Similar settings can be used for handling dual tones whose frequencies are close to each other, so long as the center frequency and bandwidth covers that of the two. Dual tone mode, however, requires two sets of predetermined center frequencies and bandwidths.

2) Correctly set characteristics of the wave form of the tone.

A. Dial Tone:

This is a continuous tone. Set the shortest continuous time duration for the tone, which is generally above 1.5.seconds.

B. Busy Tone:

The periodic duty ratio is 1:1 (i.e. equal duration for tone and silence). Set busy tone period and busy tone count threshold (to prevent misjudgment, the driver concludes that the tone is indeed a busy tone only when the successive ring count exceeds the threshold parameter).

C. Ringback Tone:

The periodic duty ratio is 1:4 (i.e. tone-to-silence ratio is 1:4). Set the continuous time duration for the tone (high LEV) and for silence (low LEV).

Misjudgment of tones is usually due to incorrect configuration of the above parameters. Analyze the tones using ShTA.exe under the driver installation directory (Refer to ShTA_UserManual.chm under the same directory). If this tool failed to analyze the tones, record the unknown tones, analyze their characteristics such as frequency and wave form with editing tools like Cooledit, then enter the correct parameters into the configuration file accordingly. Thus the system can judge the tone accurately.

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10. Why does the analog trunk channel sometimes undesirably

pick up busy tones?

Normal voice, when under the influence of poor lines and poor grounding might generate noise. Illusive busy tones that resemble the characteristics of a genuine busy tone can sometimes be found among these noises. The consequent effect can be minimized by increasing busy tone’s count threshold to filter out the illusive busy tones.

11. Why does the application fail to invoke the function

SsmAutoDial to dial a number from the trunk channel?

The function SsmAutoDial only performs dial operation upon correct detection of dial tones on the analog trunk channel. Hence, the absence of dial tones, or failure to detect dial tones will result in the function call to be unsuccessful.

12. How does an analog trunk channel detect pickup from

remote phone in a dial-up process?

When the remote end picks up the call, the trunk channel state will switch from ‘pickup’ to ‘connected’ in a dial-up process. If the telephone line connected with the trunk channel has customized the polarity reverse service, the function SsmGetPolarRvrsCount can be invoked to judge whether the remote end has picked up or not. The return value 1 indicates the pickup of the remote end.

13. Why does the analog trunk channel sometimes fail to detect

hangup from remote end?

There might be two reasons for failure to detect remote hangup.

1) Fails to detect busy tones on the line. Refer to Question 7 “How to properly configure tone analysis parameters for an analog board?” in this chapter for busy-tone settings.

2) PBX has not sent busy tones.

14. How does the analog trunk module detect connection drop

on a line?

To detect whether the connection on a line is normal or not, the analog trunk channel can invoke the function SsmGetLineVoltage and examine the voltage obtained by. If

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the line is ok, the voltage will be normal too.

15. Is it possible for driver to make a misjudgment when PBX

plays prompt tones in a call from the analog trunk channel

to a mobile phone?

Yes, the driver may misconclude that the analog trunk channel is in a ‘connected’ state.

16. Why does an analog trunk channel switch from "idle" to

"ringing" only after two rings have been detected?

This has been set so in the item ‘AlwaysToRingingOnRingCntX’ of the configuration file ‘Shconfig.ini’, and can be modified for the same action to happen after one ring or multiple rings.

17. Why does an analog trunk channel take a longer time to

determine a ringback tone when calling out?

A trunk channel will not return ‘ringback’ values before a full-cadence ringback tone is detected. i.e. it will determine a ringback tone only after the first ring from remote end has ended or the second ring starts.

18. When a remote phone places a call to an analog trunk

channel, the phone cannot hear anything played by the

channel, or hears only a short section before it is cut off.

Why?

This happens if the trunk channel is set to stop playback upon detection of barge-in and the caller interrupts playback by talking. The option can be turned off.

19. What to do if a station on analog boards can hear echoes

from itself?

Echoes are cancelled at the end they are produced. Therefore, for echoes produced at the end of analog boards, you can cancel them by changing the board model to SHT-16C-CT/PCI/EC or SHT-8C/PCI/EC which supports echo cancellation in

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hardware.

20. Can multiple analog trunk channels share the same called

party number?

This can be done through an application to the telecoms office.

Station Channel:

21. What is the maximum length of a station phone line from the

station module on an analog board?

It should not exceed 4.8km.

22. Which phenomenon indicates the damage of the station

module?

If the station channel is constantly showing “off-hook” when the station module is not connected to a phone, or showing “idle” when the connected phone is off-hook and the power indicator is on, it could be that the connection between the module and the board is loose or the station module has been damaged in transport.

23. Can the external power supply for B-type boards be used for

C-type boards, and vice versa?

Yes, because B-type and C-type station boards have the same external power interface and interface parameters. This power supply is called Ringing Current & Battery Feed Power Supply.

24. When a station phone makes a call from a station module via

an analog trunk channel, the volume heard at the remote

phone sounds rather low. Why?

We suggest invoking the function SsmTalkWithEx to increase the volume for the station or analog trunk channel to a proper level.

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25. Two station channels and a trunk channel join the same

conference. When one of the station channels plays sound

with a station phone on-hook, other channels can still hear

the sound played. Why?

With the station phone being hung up, sound played by the station channel reaches other channels as echo from the phone, which is why other conference channels can hear the played sound.

26. Why does a station phone hear howling when it makes a call

from a station module connected through a trunk module?

When a station module is directly bussed with a trunk module, and the station phone is off-hook while the trunk module remains on-hook, dissimilarity of impedance levels will cause howling to happen in a call. This can be resolved by invoking a function to link the station channel with the trunk channel over bus, while both are off-hook.

27. Why is there much noise when the station phone connected

to an on-board station module is off-hook?

This usually occurs with poor contact, bad lines, improper grounding or even interference from the power supply.

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Chapter 4 SHD Series Boards

1. What is frame-synchronization signal?

Frame synchronization signal is used for determining whether the clock on the local receiving end is synchronized with that on the remote sending end. A normal local frame synchronization signal means that physical links from the remote sending end to the local receiving end are working fine.

2. What new features do C-/D-/E-type boards have, compared

with the old A-/B-type boards?

In addition to functions native to A-/B-type boards, C-/D-/E-type boards carry the following new features or improvements.

1) Enhanced stability of soft-faxing.

2) Used DMA technique for data reading and writing, minimizing host CPU cost for large-capacity voice recording and playback.

3) Improved echo-cancellation capability.

4) C-/D-type boards are compatible with PCI-X slot.

5) E-type boards support PCIe slot.

3. What does it mean if the synchronization indicator is off or

blinking?

An unlit synchronization indicator means that the frame synchronization signal on E1 is abnormal and has skipped detection by the board; a blinking indicator means the frame synchronization signal is unsteady.

These could be due to:

1) Reversed reception/transmission cable;

2) Incorrectly configured master/slave clock on a digital board;

3) Bad or improper board grounding;

4) Reception cable not connected properly;

5) CRC-4 not set properly.

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4. What causes an SS1 call to be abnormal?

1) Abnormal frame synchronization or multi-frame synchronization signal on local or remote E1;

2) Improperly configured trunk direction on a channel (it is required to configure a channel as outgoing or incoming trunk channel);

3) Improperly configured rule of number-reception for an inbound trunk channel;

4) The telecoms service provider does not offer the relative service or service data concerned are not ready.

5. What causes an ISDN call to be abnormal?

1) Abnormal frame synchronization signal on local or remote E1;

2) ISDN is not properly configured as network side or user side;

3) Incorrectly configured CRC check switch;

4) Incorrectly set TEI (terminal endpoint identification) value;

5) Incorrectly configured representation of channel identification information (number or timeslot diagram) for outgoing calls on ISDN link;

6) The telecoms service provider does not offer the relative service or service data concerned are not ready.

A good way to deal with such problem is to record the PBX information by our boards and then modify the on-board information to be consistent with that about PBX.

6. What does the SS7 server program Ss7Monitor.exe do?

Must we run it?

The server program Ss7Monitor.exe sets up MTP3 protocol for SS7 and posses signaling message distribution and system monitoring functions. The sever program must be run in order to handle SS7 signals. Detailed information about the SS7 server can be found in the section on signaling server in SynCTI Programmer’s Manual.

7. What causes the following issues when running SS7 TUP or

ISUP protocol?

Issue 1: SS7 server (Ss7Monitor.exe) is out of service.

Possible causes:

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1) Abnormal frame synchronization signal on local or remote E1;

2) Timeslot 16 not set by the remote PBX to transfer signaling messages;

3) Incorrectly configured IP address or parameters concerning signaling in the SS7 server program;

4) The telecoms service provider does not offer SS7 service or service data concerned are not ready;

5) Incorrectly configured local DPC or OPC;

6) Bad or improper board grounding;

7) You might have set the analog board to be master clock when using both digital and analog boards on a single computer. Note that the digital board should always provide the master clock.

Issue 2: SS7 signaling links appear to be in a state of circuit reset.

Possible causes:

1) Abnormal frame synchronization signal on local or remote E1;

2) Incorrectly configured CIC number of PCM;

3) The telecoms service provider does not offer the relative service or service data concerned are not ready.

Issue 3: SS7 server repeatedly receives SNT signal from remote end.

Possible causes:

1) When SS7 is not in service: MTP3 level of the PBX is not activated (mostly due to improper allocation of the CIC number of the PCM).

2) When in normal conversation: The PBX is checking if the lines are working fine.

Issue 4: CFL signal is received during an outgoing call.

Possible causes:

1) The message sent out by Synway board is inconsistent with that identifiable by PBX. For example, the remote PBX accepts only IAI messages with caller-IDs, but the sending party makes a call without setting caller-ID and the driver automatically sends an IAM instead of an IAI message.

2) The telecoms service provider of the remote end does not offer the relative service or service data concerned are not ready.

Issue 5: The SS7 board takes 1 to 2 minutes upon service set-up before it goes into an ‘idle’ state.

This is perfectly normal. The Synway’s SS7 board automatically sends a circuit-reset message to the remote end immediately after the service is set up. It then changes the

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relative channel to “idle” upon receipt of a circuit-reset message from the remote end. However, certain PBXs will wait for 1 to 2 minutes before they send out a circuit-reset message.

Issue 6: How many signaling link sets does an SS7 board support? How many signaling links does each set support? Are multiple DPCs and OPCs supported?

An SS7 board supports up to 48 signaling link sets. Each set supports up to 16 signaling links. At most 48 OPCs and 48 DPCs are supported.

Issue 7: When calling out in SS7 TUP or ISUP, the remote PBX sees the received calling-party number to have an extra ‘0’ at the end.

This can be resolved by setting the item ‘SetSTSignal’ to ‘1’ (the end pulse signal to be sent after sending calling-party number) in the configuration file ‘ShConfig.ini’.

Issue 8: Why not call in when the TUP or ISUP channel is IDLE?

Possible Causes:

1) The PBX did not route the incoming call to the corresponding trunk channel of the board. Check the information “MP3 monitoring terminator: MSU from remote end” on the Ss7Monitor.exe interface. If neither IAM nor IAI message is displayed, it indicates that the PBX did not send the call data to the board. In such case, you shall examine the user layer business data at the PBX end to see if they are correct.

2) The number-receiving rule for SS7 at the board end is not set accurately. Check the receiving and transmitting interfaces of Ss7Monitor.exe. For TUP channels, if you see from those interfaces that the board sends the UNN (Unallocated Number) or ADI (Address Incomplete) message after receiving the IAM or IAI message from the remote end, it indicates that there is something wrong in the settings of the number-receiving rule. For ISUP channels, if you see from those interfaces that the board sends the REL (Release) message with the cause indicator 80 or 81 after receiving the IAM message from the remote end, it indicates that there is something wrong in the settings of the number-receiving rule. In such case, you shall execute ShCtiConfig.exe and set the number-receiving rule according to your actual calling-party number.

Issue 9: Why not call out when the TUP or ISUP channel is IDLE?

Possible Causes:

1) The number-receiving rule at the PBX end is not set accurately. In such case, you shall examine the user layer business data at the PBX end to see if they are correct.

2) The called-party number is insufficient in length. You shall append digits to complete it.

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3) The remote end cannot recognize the callerID as the board did not send the ST signal following the callerID. You shall set the configuration item SetSTSignal=1 under the section [TUP] or [ISUP] in ShConfig.ini.

4) The PBX rejects or disconnects the call because some other parameters at the board end (e.g. the nature of connection indicator, the forward call indicator, the calling/called party number, the calling party’s category indicator, the transmission medium requirement, etc.) do not match those at the PBX end, or because of some other reasons (e.g. the calling-out feature is restricted by the PBX). In such case, a common solution is to modify the message sent out from the Synway board to be the same as those sent in to the PBX. For example, if the transmission mediums for ISUP are different at the board and the PBX ends, change it to comply with the PBX.

8. How to deal with call failure or PCM instability in connecting

digital board to optical transceiver?

1) Check if the board has a correct clock setting and if it is grounded properly.

2) Use the PBX to perform a self-loop testing on the board to ensure the good condition of lines.

3) Directly connect the board to the PBX and not through the optical transceiver, to find out if something is wrong with the connection of optical transceiver.

4) See if the impedance of board conforms to that of optical transceiver.

5) Check if the CRC-4 check switch is set properly.

9. Why there is so much noise and why do calls sometimes cut

off when using digital trunk boards?

This is mostly due to line fault in the digital trunks, for instance, loose contact between the head and jack or broken lines. It is suggested that the remote PBX perform a self-loop testing on both incoming and outgoing trunk lines when the board is properly grounded to check if the lines are working fine.

10. How does a digital trunk voice board receive numbers with

different prefixes?

Configure the number-receiving rule to prefix mode.

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11. Is it mandatory to set TS16 for signaling transfer on a digital

board which supports SS7?

The A-type, B-type and C-type SS7 digital boards support signaling transfer on TS1 or TS16, but not on both at the same time. The D-type and E-type SS7 boards can support signaling transfer on whichever timeslot except TS0 by proper configuration.

12. How to deal with the echoes produced by the device

connected to Synway digital boards?

Echoes are cancelled at the end they are produced. Therefore, for echoes produced at the end of digital boards, you can cancel them by changing the board model to SHD-120D-CT/PCI/EC or SHD-240D-CT/PCI/EC which supports echo cancellation in hardware.

13. How to deal with link breakup on digital trunk boards and its

induced call cutoff?

1) Check if the board has a correct clock setting and if it is grounded properly. Refer to Question 9 in Chapter 1 for the correct grounding method.

2) Use the PBX to perform a self-loop testing on the board to ensure the good condition of lines.

3) See if the impedance of board matches that of PBX.

14. What is probably the cause for appearance of alarm, PCM

asynchronization or unusable channel state on the end of

PBX when it connects to Synway E1 boards?

If the remote PBX alarms during its connection to Synway E1 boards that there is a block at the board end which causes failure in calls or the PBX shows with instable PCM synchronization or unusable channel state, this is probably because of improper CRC-4 setup.

CRC-4 is used to perform a cyclic redundancy check of multiframe signals, viewing if there is something wrong with the multiframe transmission. The driver sends the CRC-4 package through Timeslot 0 and the configuration item CRC-4[i] determines if the package is opened or closed. 0: Closed; 1: Opened (default). The driver does not do any thing about the CRC-4 package passed from the remote PBX. The effect of the

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CRC-4 check switch on E1 lines are as follows.

1) The remote PBX asks the Synway board to send CRC-4 but the board does not. Then the PBX gives the alarm information saying it fails to make normal calls. In such case, the channel seems to stay in idle or other states which are all not true,

2) The remote PBX does not ask the Synway board to send CRC-4 but the board sends. If the PBX works well, it means there is no effect on the PBX; if the CRC-4 is used to solve such problems as instable PCM synchronization and unusable channel state, it means there are some effects on the PBX.

15. What is the problem and what to do if the impedance of

digital board does not match that of PBX?

The following problems may appear:

1) PCM synchronization goes instable;

2) Channels on Synway boards stay in an unusable state;

3) Instability in links induces call cutoff.

See below to find proper ways to solve these problems in 3 different situations.

1) For 16E1 and C-/D-/E-type digital boards, modify the configuration item PcmLinkType[n] in Section [boardId=x] of the file ‘ShConfig.ini’ to reset the impedance. 0 means 120Ω twisted pair cable is used as PCM link, while 1 means 75Ω coaxial cable is used as PCM link.

2) For digital boards with impedance jumpers, simply change the jumping mode to switch the impedance between 75Ω and 120Ω. The default factory setting is 75Ω in which condition the jumper cap short-circuits two contact pins. Once pulling out the jumper cap, the impedance becomes 120Ω.

3) For other digital boards which neither have impedance jumpers nor allow impedance change by resetting configuration items, the impedance is determined by hardware. If there is a mark ‘120Ω’ on the label of board model on the back of board, it is a digital board with impedance of 120Ω; or otherwise, it is a digital board with impedance of 75Ω.

Refer to relevant hardware manuals of those digital boards with impedance jumpers for particular location and treatment of jumpers.

16. How to enable the Hide-A-Caller function for digital trunk

boards?

The Synway digital board sends the calling party number and the PBX once receiving

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the message sends only rings without the calling party number to the telephone. This is called Hide-A-Caller function. It is enabled in slightly different ways for various signaling protocols.

SS7 TUP

Modify the Calling Line Identity indicator before sending the IAI message. To be exact, invoke the function SsmAutoDialEx(ch , szPhoNum, wParam) and modify the parameter wParam, such as SsmAutoDialEx(0, 110, 0x04).

SS7 ISUP

Modify the Limit on Providing Address indicator in the calling party number before sending the IAM message. To do so, you may reset the configuration item DefaultIAM_CallerParam or invoke the function SsmSetIsupFlag.

Set under Section [ISUP] in the file ShConfig.ini: DefaultIAM_CallerParam=0x1401;

Invoke the function SsmSetIsupFlag(ch, nType, dwValue, pV) and modify the parameter dwValue, such as SsmSetIsupFlag(0, 1, 0x1401, NULL).

ISDN

Modify the Presentation indicator in the calling party number before sending the SETUP message. To do so, reset the configuration items under Section [ISDN] in the file ShConfig.ini (if the following configuration items can not be found in this file, they should be added manually).

PresentNumber =1: Sets the field in the signaling message that determines whether to display the calling party number. 0: not display; 1: display (default).

17. What problems may be caused if board clock is wrongly

configured? How to set it properly?

The incorrect clock configuration may result in the following problems.

1) A blank screen appears and the computer fails to start;

2) Fax data transmission may be interrupted, which brings about code slip;

3) An E1 link goes instable or becomes disconnected, which disrupts the call.

There are two clock settings: one is for the system involving all voice boards concerned, and the other is for PCM on digital station tap boards. They both should be performed properly.

1) Clock setting for whole board system

By using the configuration item WhoSupplySysClock in the file ShConfig.ini under Section [SystemConfig], you can set the ID number of the board that provides the clock for the whole system, with the value range of 0~N-1 or N.

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i. The value N indicates all board clocks within the system should be enabled. It is applicable to such situation that all boards within a system do not connect to each other over bus.

ii. The value -1 indicates the board is independently used as a slave board. However, to set this value, you should manually modify the item WhoSupplySysClock in shconfig.ini. Especially in such situation that you use boards from Synway and other manufacturers within a system, don’t forget to select a manufacturer’s board to provide system clock. If boards from different manufacturers all provide system clock, the computer probably can not start with a blank screen on it, or other abnormal phenomena may occur.

iii. 0~N-1 indicates the board specified by this parameter provides the clock, while other board clocks are all disabled. It is applicable to such situation that all boards within the system connect to each other over bus. In a system containing both digital and analog boards, you’d better choose the digital trunk board which connects with the PBX to provide the clock.

2) Clock setting for PCM on digital trunk boards

This clock setting is for a system involving digital trunk boards to get the synchronization of system clock and PBX clock. The Synway driver offers three modes for such clock setting: line-synchronization mode (master clock), free-run mode (master clock) and slave clock. Usually, if the PBX can provide a clock, the board which supplies system clock sets a PCM on it with the line-synchronization mode (master clock) and other PCM with the slave clock mode; if the PBX can not provide a clock, the board which supplies system clock sets a PCM on it with the free-run mode (master clock) and other PCM with the slave clock mode.

18. When you use a DTP series board monitoring the ISDN E1

line, if such problems as "what you have recorded is nothing

but silence", "a single call is divided and recorded into

several voice files", or "several calls are recorded into a

voice file" occur, what is the reason for them and how can

you deal with them?

There are two ways allowed for E1 devices to number the channels: using the continuous codes (1-30), and using the discontinuous codes (1-15, 17-31). Generally speaking, E1 devices use discontinuous codes to number the channels, that is, use Timeslot 16 to deliver signaling messages and the rest Timeslots 1-15 and 17-31 to deliver voice data. However, some particular devices (such as T1-to-E1 converter) may use continuous codes to number the channels, that is, use Timeslots 1-30 all to

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transport voice data.

Such situations as we mentioned above in the question will probably occur when the numbering method set by the Synway driver is different from that set by relative parameters for the monitored lines. As the Synway DTP series boards use discontinuous codes to number channels by default, if the device you use adopts continuous codes to number lines, the signaling messages and voice data on Timeslots 16-31 will become staggered. For example, the status displayed for Channel 18 is actually the status of Channel 19, but what you have recorded is still the voice data on Channel 18. In such situation, if there is a call on Channel 19 but no call on Channel 18, what you record is just silence but the status displayed is ‘Connected’ or ‘Talking’; if there are several calls on Channel 20 but only one call on Channel 19, what you record is only a call but it is divided into several voice files to correspond to the several calls on Channel 20; if there is only a call on Channel 19 but several calls on Channel 18, what you record are several calls but they are put together into a voice file to correspond to the only call on Channel 19.

To solve these problems, you can modify the setting of the configuration item SpyT1TransE1Line in the Synway configuration file ShConfig.ini, Section [SpyPcm], to change the numbering method. Once the issues mentioned above appear, change the value of this configuration item to 1.

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Chapter 5 SHN Series Boards

1. What VoIP protocols and voice transmission formats do

Synway VoIP boards support?

Protocol: SIP;

Voice transmission formats: G.711 A-law, G.711 μ-law, G.729A, GSM.

2. Which part of SIP protocol is supported by Synway VoIP

boards at present?

RFC3261.

3. Do VoIP boards support conferencing?

Yes. They include H.100 bus interface to enable conferencing with other boards.

4. Do VoIP boards support the access to IMS devices?

Currently, the VoIP boards support the connection to the CMCC IMS network.

5. Why are there a lot of echoes on the line when making an IP

call?

This can happen if there is network delay or echo cancellation at the remote end is poor.

6. The IP board driver is loaded successfully but the channel

state shows unusable. Why?

Usually, this can be resolved by doing the following steps and settling relative problems.

1) Check if the VoIP board is well connected to network. If the network connection fails, the channel state will transfer to “Unusable”.

2) Examine if the IP setting in the configuration program matches the actual IP address of the local PC.

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3) See if communication ports are occupied by other application programs.

Note: For SynCTI Ver. 5.3.2.1 and above versions, if case 2) or 3) occurs, the board will fail to be initialized and the system will report error. Therefore, the channel state will not transfer to “Unusable”.

7. Which purposes can VoIP boards be used for?

The VoIP boards can be used with other voice boards from Synway to help develop multiple VoIP platform systems, such as trunking gateway, VoIP call center and IP PBX.

8. What methods do Synway VoIP boards support for DTMF

transmission? What is the difference among them?

The SIP protocol supports in-band, RFC2833 and SIP Info to send DTMF.

The SIP protocol stacks adapted for the VoIP board use the RTP data packets to transmit voice data,

The RTP data packet consists of two parts: RTP header and RTP payload.

RFC2833: The RTP header carries DTMF information. This method provides high discrimination but may cause data loss in poor networks as the RTP data packet is delivered on the basis of UDP.

In-band: The RTP payload carries DTMF information. Because the information is encoded and needs to be decoded in hardware, this method provides a lower discrimination than RFC2833, and also it may cause data loss in poor networks.

SIP Info: The SIP message carries DTMF information. This method provides high discrimination but may cause data loss in poor networks as the SIP signaling packet is delivered on the basis of UDP.

9. What contributes to failure of initializing VoIP boards?

Reasons for failure to initialize VoIP boards based on SIP protocol:

Signaling IP or signaling port set incorrect

When an improperly configured signaling IP or signaling port is occupied by other application programs, the initialization of the VoIP board will fail.

Network card connected with no network or connection invalid

If the signaling IP is not a loop address and the corresponding network card connects with no networks or the connection goes invalid, the initialization of the VoIP board will fail.

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10. What causes call failure on IP channels? Reasons for call failure on IP channels:

Incorrect target address or port

Check the setting of IP address and port at the remote end. The default signaling port suggested by SIP protocol is 5060.

A party of the call is behind firewall or NAT

If a party of the call stays is behind NAT, it is necessary to perform NAT traversal with the help of IP server; if a party or both parties of the call stay behind firewall, it is necessary to check if the call has been blocked by the firewall.

Signaling messages incompatible

The call probably fails because incompatibility of signaling messages occurs during the call. Use capture software like wireshark to check if there is something wrong with signaling connection.

Others

Other reasons, such as network faults and IP server breakdown, can also cause call failure on IP channels. Use capture software like wireshark to check.

11. What is the SIP log for? How to set its levels?

The SIP log is an effective tool for positioning errors. The output logs are usually used

to analyze:

why the call cannot be established; why the sound cannot be heard or the sound effect is bad; the previous situation of the current call state machine and how it processes

call aberrance; the pressure of the current call; whether the A3 event has been thrown out properly.

We suggest you set the log level to Error while operating a program, set it to INFO1 while debugging a program, and set it to INFO2 when errors occur in a program.

12. What may contribute to failure of using VoIP boards for

registration?

This is probably because the registration port of the VoIP board or the registration server is occupied.

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13. When a B-type VoIP board is configured with proper

registration information to register a server on the

public network, the channel state shows registration

failed while the application is running, why? How to

locate the exact reasons?

1) Problems in board or driver

To confirm that there is nothing wrong with the board and the driver, first you should apply the default setting of the board and start the test program to check if all channels stay in the idle state. Then select Channel 0 to call the default IP address, fill in the DTMF column with 192.168.1.1 (i.e. the local PC) and invoke the function SsmAutodial. If Channel 1 rings, it means the board self-loop configuration is passed, both call and talk going well within the loop.

2) Registered account or SIP server unavailable

Install an SIP soft-terminal and use the registered account to see if you can register the server and call out successfully. If you can, it is sure that there is nothing wrong with the registered account and SIP server too.

3) The SIP server doesn’t pass the authentication information sent from the board.

4) Certain SIP servers only support a particular model of soft-terminals or IP phones.

14. What do you do if the DTMF digits received by SHN

series boards fail to be passed over bus to the

cooperating SHD/SHT series boards and

consequently fail to be sent out by them?

DTMF digits can be transmitted over IP by the following three methods.

In-band Mode: delivering DTMF by voice; Out-of-band Mode: delivering DTMF by signaling; RFC2833 Mode: RFC2833 is the standards-based mechanism used to send

DTMF digits in-band (RTP). In this mode, the DTMF signals are packed to be different from common voice data before being sent or received.

If you don’t use the in-band mode to send DTMF digits, the problem as described in the question will probably occur. There are two ways to solve this problem. See below.

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1)Modify the DTMF transmission mode for the IP device that communicates to the SHN series board. Change it to in-band mode. Then the DTMF signals will be delivered by voice and passed over bus directly to the SHD/SHT series board, and sent out straight by the SHD/SHT series board.

2)Use the program flow control solution. First, use the SHN series board to receive the DTMF signals that are sent in out-of-band or RFC2833 mode. Then invoke the function SsmTxDTMF on the corresponding channel of the SHD/SHT board that is connected with the SHN board by bus, to pass the DTMF digits and subsequently send them out.

15. How to deal with such voice problems as poor voice

effect, no voice pass, single voice pass, noise, etc,

appearing in normal calls on a VoIP board?

During the process a Synway VoIP board starts and establishes a call with other VoIP terminals, voice data and signaling messages are transmitted respectively. The former is over RTP protocol while the latter is over SIP protocol.

If the Synway VoIP board is able to connect a call with other IP terminals, it indicates the signaling interact between both parties is good. To find the reasons and solutions for no-voice-pass failure, refer to the 1st, 2nd, 3rd, 4th, and 5th of the below; to find the reasons and solutions for poor voice effect, refer to the 6th of the below; to find the reasons and solutions for poor speech quality, refer to the 7th, 8th, 9th and 10th of the below; to find the reasons and solutions for single-voice-pass problem, refer to the 3rd, 4th and 5th of the below.

See below for details.

1) If the IP address for voice transmission is unusable, the no-voice-pass failure may appear. Causes and troubleshootings are as follows: a) The board is not connected to a network cable or the modular plug is loose. You

can go in the operating system ‘Start Run’ and use the ‘ping’ command to check. b) There is ARP virus (address deceiving virus) in the network. As to the method of

detecting whether your computer has been infected by this kind of virus, you can search it on the Internet.

c) If the board is connected via a network cable to a network exchange on which the IP address assigned to the board is bound to other MAC address, it may cause the no-voice-pass failure. You can check the configurations of the network exchange to exclude this possibility.

2) The UMCT series IP board has a front panel and a rear panel. The network interface is located on the rear panel. So if the front and rear panels are not in position, it will lead to the no-voice-pass failure.

3) If one or both parties of a call have installed a firewall or the like, it may cause voice

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streams being blocked and lead to the no-pass or single-pass problem. Now you can use a capture software like wireshark to verify the reception of RTP voice streams.

Note: The RTP voice data packet of a B-type VoIP board transmits over the network interface. For how to capture the RTP packet, please feel free to contact our technicians.

4) If one or both parties of the call stay behind NAT, the poor capability in NAT traversal of either party may cause the single-pass problem or even the call failure. Now you can use a capture software like wireshark to capture signaling packets for analysis and check whether the following 2 IP addresses are public network addresses. The field Contact gives the signaling IP address while the field Connection Information gives the IP address to transmit the RTP data.

5) If a VoIP board seats in a LAN and an incoming party telnets to the LAN via a remote tool and establishes a call with the VoIP board, the field SDP in the Invite message which is sent by the incoming party indicates the IP address of the public network, and the VoIP board will reply by the IP in SDP but not by the original IP in the RTP packet, which will cause the single-pass problem and make the other party not hear from the VoIP board. To solve this problem, you can modify the configuration item AutoDetectRemoteRTPAddress to enable the remote RTP address self-adaptive feature.

6) The quality of network has a significant effect on the voice. If the voice effect is not satisfactory, you may use the following ways to adjust it: a) Broaden the network bandwidth. A too-narrow bandwidth may result in the network

congestion as well as a longer delay in data transmission. b) Use G.729 codec. G.729 corresponds to the rate of 8kbps while G.711 A-law and

G.711 μ-law correspond to the rate of 64kbps. Using G.729 can greatly reduce the

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bandwidth for a voice path. Actually, when the SHN-32A-CT/PCI board uses G.711 A-law or G.711 μ-law, the bandwidth for a voice path is 78.4kbps; when it uses G. 729, the bandwidth for a voice path is only 22.4kbps.

7) The mismatch of the voice CODECs between RTP packets on both sides results in the appearance of noise. In such situation, you can use a capture software to record the RTP data at both parties during the call and analyse the coding format to verify or eliminate this cause.

8) The inconsistency of the load of RTP packets on both sides may result in the appearance of noise. At present, our driver supports the load of 20ms or 30ms, which can be configured via the configuration items SizeG711A, SizeG711U and SizeG729.

9) In a system there is another board (for example SHD series board) besides the VoIP board and these two boards are connected with each other by CT-BUS, if the other board is uninitialized, noise may appear during the call on the VoIP board. So in case the other board is not used, you‘d better disconnect it from the VoIP board.

10) If it is the soft terminal that is talking to the VoIP board, the following two reasons may cause such voice issues: a) As the called party, sometimes eyebeam1.1 can not hear from the calling party

because of the incompatibility with Win7. Update to eyebeam1.5 to fix this problem.

b) The improper configuration of the sound card at the soft terminal may result in the appearance of noise, current noise, echo, etc. To fix it, modify the configuration of the sound card in the corresponding operating system. In WinXP: At the lower right corner of your desktop, click to choose ‘Volume’ > ‘Options’ > ‘Properties’ > ‘Mixer device’ > ‘Recording’, go ‘Recording Control’ and set ‘CD Volume’, ‘Line Volume’, ‘Stereo Mix’ to Mute. See below:

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In Windows 2003: At the lower right corner of your desktop, click to choose ‘Volume’ > ‘Properties’ > ‘Recording’, go ‘Recording Control’ and set ‘CD Volume’, ‘Line Volume’, ‘Stereo Mix’ to Mute. In Win7: Go to ‘Recording Control’ > ‘Microphone’, and set ‘MIC Boost’ to 0.

If all the possibilities mentioned above have been excluded, the cause may lie in the physical fault of the board. You can contact our customer service to replace the board for further tests.

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Chapter 6 ATP Series Boards

1. What should the input impedance on a recording module

be?

It should be: ≥1MΩ/500V DC; ≥10kΩ/1000Hz AC.

2. How to sample voices from a microphone or radio using a

microphone module or recording module?

Since microphones and radios have no electric voltage, the configuration item ‘IgnoreLineVoltage’ in file “ShConfig.ini” should be set to 1 to sample voices from these devices.

3. A recording channel is monitoring a trunk phone in parallel,

but the recording channel constantly appears to be off-hook

even if the trunk phone hangs up. Why?

This can happen if the on/off-hook voltages set in the recording module do not match the voltages in reality. To correctly identify on/off-hook statuses, set the on/off-hook voltages defined in the recording modules to the mean values of those in reality.

4. How does voice-activated recording work?

Voice-activated recording uses a function call by the driver to detect if there is voice activity on the line, and starts recording upon activity and stops for inactivity over a set duration.

5. Does the recording module support polarity reversal

detection?

Yes.

6. Can a trunk module be used as a recording module?

For SHT series B-type and C-type boards, yes, a trunk module can be configured to work as a recording module by modifying the configuration item

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‘SetAnalogChToRecCh’ based on the driver SynCTI 5010 or above. For SHT series D-type boards, a trunk module cannot be configured to work as a recording module.

7. Why does the recording system generate a 1K size recorded

file?

Possible reasons are as follows:

1) The computer is not installed with an MP3 engine but the application program records in MP3 format;

2) Such problem may occur if you use the SynCTI 5.0.2.0 driver and the MP3 engine provided with the 5.0.2.0 demo package for MP3 recording. In that case, we suggest you upgrade the driver and the MP3 engine to 5.0.3.0 or above versions. And the driver and the MP3 engine may as well be of the same version.

If you are sure such problem is caused not by the above reasons, update the driver to the latest version published on the website of Synway and try again. If the problem still exists, contact our technical support for help.

8. Where can I find FAQs about the ATP2400 recording server?

Refer to Appendix B Troubleshooting in ATP2400 Manual.pdf on the SUPPORT page of our website.

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Chapter 7 DST Series Boards

1. When the digital recording board is connected in parallel to

the digital phone, the channels stays disconnected or

between idle and disconnected. Why?

This usually happens when the line connection is poor, or when the board has not been correctly configured for the PBX model and the module type.

2. How to debug the digital recording board for monitoring?

What to do if the monitored PBX or phone models are not

supported by Synway driver?

The debugging procedures are as follows:

Step 1: Use ShCtiConfig.exe to do proper configurations according to the actual PBX and phone models as well as the corresponding firmware model for the PBX under the D-channel mode, referring to the latest version of the file DST Boards Supported PBX Models which may be downloaded for free from www.synway.net. Then click the button ‘Application’ to pass the configurations.

Step 2: Connect lines in parallel and keep the phone on-hook. Then use Test.exe to record silence, seeing whether there is noise within it. If any noise appears, contact Synway technical support department for help.

Step 3: Call in and out through the phone. Run Test.exe to check if any change in channel state and relative event output can be detected correctly. Once you find out some problems, use the signal tracking tool provided by Synway to seize the signaling messages and write down your performed operations step by step. Then submit them to Synway technical support department for help.

Step 4: Write the subsequent programs.

If you are monitoring some PBX or phone models that are not supported by Synway driver, please consult Synway technicians for the file DST Board Debugging Manual.pdf.

3. What are the supported digital PBX and phone models?

Refer to DST_PBX_Support.xls under the driver installation directory C:\Shcti for more

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information on supported PBXs and phones.

4. How to record one-way voice files while monitoring both

parties of a call?

The digital recording boards can adjust both inbound and outbound volumes. Just invoke the function DTRSetMixerVolume and set both of them to -7.

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Chapter 8 SynIPR Series

1. What protocols and RTP codecs do SynIPR support now?

Supported protocol:

For SynCTI Ver. 5.3.2.7: SIP, CISCO SCCP, AVAYA H323, SHORTEL MGCP, H323, PANASONIC MGCP, TOSHIBA MEGACO, SIEMENS H323, ALCATEL, MITEL, LG Nortel.

For other versions, refer to the section “1.23.9.2 Supported Protocol” in SynCTI Programmer's Manual.pdf.

Supported RTP Codec: G.711A, G.711U, G.729A/B、G.722.

Note: The supported protocols mentioned above are all standard. Any protocol, if encrypted, will be not supported.

2. Can SynIPR work under Linux?

Yes. For a 64-bit Linux operating system, you shall use SynCTI Ver. 5.3.2.3 or above versions.

3. Can SynIPR run on virtual machines?

SynIPR can run on virtual machines as long as the following conditions are met:

1) The USBKey can be recognized;

2) The network card is mapped;

3) The HASP has been installed correctly.

4. Can SynIPR work under Win8 operating system?

Yes. But you are required to use SynCTI Ver. 5.3.2.3 or above versions as it adopts winpcap 4.1.3. Win8 operating system does not support winpcap versions below 4.1.3.

5. What are the commonly used events to start/stop recording

in SynIPR programming?

Generally, the application will start recording upon the detection of the event E_RCV_IPR_MEDIA_SESSION_STARTED and stop recording upon the detection of the event E_RCV_IPR_MEDIA_SESSION_STOPED. Also, you can designate other events to start and stop recording according to actual requirements.

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6. Can SynIPR obtain callerID and calleeID?

Yes. If the monitoring protocol supports the call control events under the D-channel events, the callerID and the calleeID can be obtained via the parameters szCallerId and szCalledId in the structure tagIPR_CALL_INFO of the pvBuffer pointer in such call control events as DST_CALL_IN_PROGRESS, DST_CALL_ALERTING and DST_CALL_CONNECTED. If the monitoring protocol does not support the call control events, the callerID and calleeID can be obtained via the pointer pvBuffer in the LCD events under the D-channel events.

7. How to judge the call direction in SynIPR?

If the monitoring protocol supports the call control events under the D-channel events, the call direction can be judged via the parameter CallSource in the structure tagIPR_CALL_INFO of the pvBuffer pointer in such call control events as DST_CALL_IN_PROGRESS, DST_CALL_ALERTING and DST_CALL_CONNECTED.

8. What is the difference between the two monitoring modes

"Monitor by Station" and "Monitor All"?

“Monitor by Station” means to monitor designated stations, while “Monitor All” indicates to monitor all calls.

9. Can a SynIPR channel be bound with a designated IP

phone?

Yes. You can bind a SynIPR channel to an IP phone by the following method: define a Station and add such information as the IP address, MAC address and port of the designated IP phone to the Station. Then invoke the function SsmIPRAddStationToMap to add the Station to the map list.

10. What to do to increase/decrease the amount of SynIPR

channels or to prolong the time of authentication?

Please feel free to contact our salespersons for authentication upgrade.

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11. Why does the application detect the event

E_RCV_IPR_AUTH_OVERFLOW (authentication overflow)?

How to deal with it?

Possible causes:

1) A call has multiple sessions (such as a call with multiple parties or a call conference) and the total amount of sessions is larger than the authenticated number.

2) Due to network topology or PBX configuration, the monitoring port is transmitting or receiving messages of the current protocol other than the signaling messages. The driver mistakes such messages for active Stations. Thus, the total amount of Stations is larger the authenticated number.

For case 2), we recommend that you select the “Monitor by Station” mode to monitor the designated extension/IP address/MAC address.

12. Why does the recorded file come out silent?

Possible Causes:

1) Fail to invoke the function SsmIPRSendSession.

2) The mirroring port does not duplicate the RTP data, which leads to the loss of RTP packages.

3) The RTP codec of the monitored IP phone is not supported by Synway’s driver. You shall modify the RTP codec to a supported one.

To confirm the actual cause of the problem, you shall send such information as the API log of the application, the network data package captured by wireshark or similar software and the configuration files to our technicians to further analyze the problem.

13. Why does the recorded file only contain one party's voice?

Possible Causes:

1) The mirroring port only sends one party’s RTP package.

2) The two sides of the conversation use different RTP codecs either of which is not supported by Synway’s driver. In such case, you shall modify the configurations of the phone.

To confirm the actual cause of the problem, you shall send such information as the API log of the application, the network data package captured by wireshark or similar software and the configuration files to our technicians to further analyze the problem.

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Chapter 9 Fax Application

1. How can WORD or PDF files be faxed?

They can be converted to supported formats –*.tif (T4) files – before being faxed out.

2. What file formats does the fax board support?

Supports standard *.tif (T4) documents, which are featured as follows:

1) Color: Black and white.

2) Resolution: 204 * 196、200 * 200 (fine mode);

204 * 98、200 * 100 (common mode).

3) Fax standard: MH, MR, MMR (Note: MR and MMR are supported by SynCTI Ver.5.3.1.5 and above)

4) Pixels: image width of 1728, and no requirements for image height.

3. What contributes to failure of establishing a fax handshake

or carrier wave?

1) Bus cable is not connected or loose.

2) Wrongly configured bus master/slave clock, e.g. configured multiple master boards.

3) Fax channel is not connected to relative phone channel, or connected channel is not in ‘talking’ state.

4) Has not enabled fax channel in driver application.

5) Remote facsimile machine is in the process of sending when a send-task is started in the fax channel.

4. How to append fax file during a fax transmission?

You can append a document only after the transmission of the first file begins. Additionally, a file to be appended must have the same format as the file in transmission.

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5. Why does a received tif file sometimes show white text on

dark background when opened with ACDSEE?

This is due to the reversal bit of a tif file, which does not affect fax transmission/reception.

6. What are features of the Synway soft-faxing?

It has the following features:

1) Compression encoding: MH, MR, MMR supported

2) Error Correction Mode (ECM) supported

3) Fax protocol: V.17, V.29, V.27, V.34 supported

4) Baud rates: 33600bps, 14400bps, 9600bps, 7200bps, 4800bps supported

7. Is it possible to convert tif files without doing it manually?

The function ‘ShellExecute’ in Windows can be used to print Word and Excel documents in .tif format.

8. Can additional information such as company name be added

to the fax documents in transmission?

Yes. Fax headers can be added to .tif files.

9. Can we choose to fax certain pages of the document only?

Yes.

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Chapter 10 Voice CODEC Support

1. Which CODECs are supported for file recording/playback?

Play/record function supports the following CODECs at a sampling rate of 8000 times per second:

1) For standard wav files: Single-channel A-Law, μ-Law and 8-/16-bit linear PCM, IMA-ADPCM, Dialogic-ADPCM (VOX), MP3, GSM and G.729A are supported. For stereophonic recording which is invoked by SsmRecStereoToFile, A-Law and μ-Law are supported.

2) For plain files (i.e. files without header information): Single-channel A-law, μ-law, 8-/16-bit linear PCM, IMA-ADPCM, Dialogic-ADPCM(VOX), MP3, GSM and G.729A are supported.

3) For standard MP3 files: Single-channel MP3 is supported.

2. Which compression format does recording use, and at what

rate?

The following recording compression formats are supported:

1) IMA-ADPCM or Dialogic-ADPCM (VOX): Compression rate of 2:1, i.e. voice data are generated at 4kbps.

2) GSM: Compression rate of 5:1, i.e. voice data are generated at 1.6kbps.

3) MP3: Compression rate of 8:1, i.e. voice data are generated at 1kbps.

4) G.729A: Compression rate of 8:1, i.e. voice data are generated at 1kbps.

3. How to play WAV files recorded by a sound card with a

Synway voice board?

The files can be directly played by Synway voice boards if the file format is supported by our product. Otherwise, they have to be converted into a supported wav format with third-party software tools such as Cooledit.

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4. Why are noises sometimes heard in the process of file

playback?

This generally happens when voice data are corrupted or when incorrect play-format is used to decode an unformatted voice file. Another reason is the voice codec being used to play files is unsupported by Synway boards.

5. How to play or record GSM, MP3 voice files?

To play or record GSM, MP3 voice files, you should first modify the configuration item under Section [SystemConfig] in the file ShConfig.ini to GsmCodecEnable=1, and be sure to install necessary codec engines (The engine for GSM is usually carried by Windows operating system while that for MP3 could be acquired from the DEMO package released at our website).

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Chapter 11 Conferencing and Bus Applications

1. How to play background music in a teleconference?

Join the conference as an ‘always speaking’ participant with a channel, then play background music to the bus. This can then be heard by other members in the conference

2. Does teleconferencing occupy additional channels?

No. Synway voice boards have built-in distributed, interactive teleconferencing function that allows an unlimited number of conferences and participants within each conference. Background music playback, monitoring and recording of a conference can also be done without taking up additional resources.

3. How to record conversation of all conference members at

one time?

Turn on the mixer for one of the channels by the relative function call. The conversation of all conference members can be recorded by recording this mixed channel.

4. Why does howling occur when operation is carried out on a

bus or in a conference?

This generally occurs due to dissimilar impedance levels when using analog boards. Operations on the bus or in a conference are advised to be done only after the channels have gone into ‘talking’ state. When a certain channel is idle, refrain from operation on the bus or teleconference.

5. How can an outsider listen to the entire conference

conversation?

This can be done by calling the function SsmLinkFrom().

6. Up to how many conference groups and participants are

allowed in a conference?

There is no limit to either.

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7. Sometimes when a few channels from a digital board join the

conference as dynamic members, a speaker's voice cannot

be heard by other members or is only heard after a delay.

Why?

This usually happens when the channel on the board fails to detect a barge-in. If such an arrangement is needed, we suggest changing the barge-in sensitivity in the relative item in ‘ShConfig.ini’ to around 10.

8. In a teleconference, how does the board prevent DTMF key

tones in a channel from being heard by other channels?

The board supports the DTMF-clamping feature that can prevent DTMF key tones from overflowing to other channels.

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Chapter 12 About Linux

1. How to deal with the failure of running test program at

common user access to Linux?

It is a problem of access rights.

Solution: Go into the directory etc/udev/rule.d, modify shdpci.rules, and then restart the computer.

For example, to give the user name of Synway the access right to Linux, you only need to add the following 4 lines to shdpci.rules.

KERNEL=="shdpci-*", NAME="shd/pci9000-%n" GROUP="synway" MODE="0664"

KERNEL="shdpci-*", NAME="shd/pci9000-%n" GROUP="synway" MODE="0664"

KERNEL=="shdusb-*", NAME="shd/shdusb-%n" GROUP="synway" MODE="0664"

KERNEL="shdusb-*", NAME="shd/shdusb-%n" GROUP="synway" MODE="0664"

2. What to do if the driver reports "insmod shdpci.ko error"

during its installation under Linux?

You can troubleshoot the problem following the instructions below:

1) Check if the driver kernel information matches the system kernel information. (Generally, the load errors are caused by the inconsistency in kernel information);

2) If the driver kernel information do not match the system kernel information, please contact our technicians and offer the information of “Uname –a”, “cat /proc/version” and “lsb_release –a”.

3) Our technicians will provide you with a new driver installation package.

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Chapter 13 About Asterisk

Refer to the manual of AST series boards for the problems on AST series boards.

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Chapter 14 About UMCT

1. What to do if the switch shuts down with a blue

screen during the runtime?

Reasons and solutions for the blue screen of death during the UMCT runtime:

1) The main control board’s poor heat dissipation may cause a blue screen. If you are using the SSW-08 series UMCT switch, check if the main control board is SCU03 without a fan. If it is, replace it with SCU03-3 which includes a fan.

2) A bad memory bank may cause a blue screen too. Replace it with a good one to confirm or eliminate this possibility.

3) Poor contact between the voice board and the switch may also cause a blue screen. Replug the voice board to exclude this possibility.

4) Hard disk failure may cause a blue screen too. You can use such tool as HDTunePro.exe to test the HD.

How to use the tool HDTunePro in WinPE to check if the hard disk has bad tracks or not?

Preparations:

1. Make a WinPE startup U-Disk. 2. Copy HDTunePE.exe and shfolder.dll to the U-Disk.

Directions:

1. Click ‘Error Scan’. 2. Press ‘Start’. Note: Don’t tick ‘Quick Scan’. 3. Result: Red means bad tracks exist and you need to replace the HD; green is for ok.

2. What to do if the UMCT switch can't find the HD upon

startup?

The factory default value of ‘Integrated Peripheral → Onchip IDE device → on ChipSerial ATA’ in CMOS settings is SATA ONLY. Either the JP1 jumper on the main control board (staying at the 1-2 position by default) or the CMOS battery being moved, the CMOS parameters will reset, and the above value will change to Disable. In such situation, the UMCT switch may not find the hard disk upon startup. To solve this problem, you can open BIOS to reset the CMOS parameters. To be exact, go to ‘Integrated Peripheral → Onchip

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IDE device → on ChipSerial ATA’, you will see three options: Disable, Combined Mode and SATA ONLY. Choose the third option SATA ONLY.

If the hard disk is still not found at the setting of SATA ONLY, it may be something wrong with the hardware. Please send the machine back to our company to test.

3. Why does the SSW080B UMCT keep alarming upon startup?

SSW080B UMCT has redundant power supply. If only one power source is provided, the UMCT will keep alarming. Press the red button ‘BUZZER RESWT’ can remove the alarm.

4. What to do if the USB CD-ROM connected with UMCT cannot

be started normally?

Provided no failure occurs in hardware, usually this problem is caused by insufficient power supply using single USB interface. One solution is to connect both USB interfaces on the USB CD-ROM with the UMCT. Another option is to provide power using USB HUB, i.e. connect the USB CD-ROM to a USB HUB with power source using one USB line and use another USB line to connect the interface of the HUB with the UMCT.

5. How to set the auto-start-upon-power-on feature for the

UMCT series SCU03 boards?

Enter the bios interface and select the item “Integrated Peripherals”. See the figure below.

Set the option “SuperIO Device” to “ON”. See the figure below.

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Finally, restart the machine to validate the settings.

6. How to start the UMCT without keyboard?

Press ‘delete’ after starting up to enter CMOS. Enter the option ‘Standard CMOS Features’ and set the ‘Halt On’ configuration to ‘No Errors’ or ‘All, but keyboard’.

7. How to solve the problem of blank screen upon startup?

1. After plugging in the UMCT, the ‘standby’ indicator on the main panel stays green, indicating that the power line is normal. It goes out upon the startup of the machine; meanwhile the ‘power’ indicator stays green.

2. If the ‘HDD’ indicator does not flash, remove the boards (if there is any) on the UMCT and reboot the machine in order to eliminate the causes such as poor contact and hardware failure. If this still does not work, we basically conclude that the problem is caused by the main board. Please send the machine back to our company to test.

8. What to do if all parameters (such as Time) of the UMCT are

restored to factory settings after power-off and restart?

There are two possible causes as listed below:

1. The CMOS battery (see figure below) is out of power and required to be changed.

2. The jumper switch beside the CMOS battery (see figure below) is not set to “NORMAL”.

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Jumper Switch

CMOS Battery

9. Not being used for a time, the SSW080A UMCT fails to start

(the pwr indicator is on but the machine has no response

after you press the start button) and the PC fails to detect

the HD. Such problem disappears after you power off the

machine, wait for a short time and then restart it. What is the

cause?

It is because the power supply is in the protected mode due to overload or short circuit. In such case, the machine cannot be started until you power it off and wait for a time.

10. What shall you pay attention to while using UMCT intelligent

switches?

When you use a UMCT intelligent switch, we strongly recommend that you use the same power socket for the machine and the display lest the poor grounding cause large difference in voltage which may disable the machine or even burn down the elements.

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11. When the UMCT switch is restarted after abnormal power-off,

it can neither transmit nor receive data packets although the

network card driver is good. How to solve the problem?

A too quick restart of the device after abnormal power-off results in no reset of some modules and leads to such problem. Wait for at least 10s after power-off before you restart the UMCT switch.

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Chapter 15 Issues about Application

1. When debugging, the application is deliberately forced to

end and succeeding re-initialization fails. Why?

Forced interruption during debugging prevents the memory of the board driver from being properly uninstalled. To avoid error, we advise calling the function SsmCloseCti to uninstall the board before ending the program.

2. Do Synway boards support multi-thread programming?

Yes.

3. Can two applications supported by the same driver be run

on one system?

Yes, the driver supports multiple processes. However, different applications should run under separate directories, and the board cannot take function calls from different applications at the same time.

4. When the application generates a smaller buffer for

recording and uses a dual-buffer for playback, noise is

heard in playback. Why?

The default recording buffer size of the driver exceeds 800 bytes. When the application generates a buffer smaller than 800 bytes, modify the configuration file to turn on the small-buffer recording, i.e. set the configuration item RecordAndPlayUseAsIP=1.

5. Why does voice become intermittent during a playback with

DTMF signals available on the line?

The driver is set by default to stop playback upon detecting DTMF signals, and intermittency could be due to the board driver picking up DTMF during playback. To prevent this from happening, set the value of ‘DefaultPausePlayOnRxDtmf’ under [BoardId=x] to 0 in the configuration file ‘ShConfig.ini’.

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6. How to transfer an incoming call to an analog trunk channel

to a call-agent?

1) When the trunk channel detects ringing, the station channel invokes the function SsmStartRing() which causes the station phone to ring.

2) When the station channel senses that the station phone is off-hook, the corresponding trunk channel invokes the function SsmPickup() to pick up the call.

3) Invokes SsmTalkWith() and connects the off-hook trunk channel with the station channel through the bus. The trunk now talks with the station.

7. How to send CallerID via a station channel to a phone that

can receive CallerID?

1) For DTMF: The station channel can invoke the function SsmTxDTMF to send CallerID before the first ring.

2) For FSK: The station channel can invoke the function SsmStartRingWithCIDStr to send CallerID.

8. Which programming platforms does the driver work on?

The driver can work on all programming platforms that can directly invoke the standard Win32 interface, such as VB, VC++, .Net, DELPHI, C++BUILDER, PB, VFP, JAVA, etc.

9. What is the difference between the programming modes

CALLBACK and EVENT_POLLING?

EVENT_POLLING: Uses the polling mechanism and supports self-defined events.

CALLBACK: Supports the callback mode.

10. Why does the application sometimes not detect the change

in call or monitoring status?

Under the condition that the application uses the polling mode for programming and high frequency of interface refresh results in high CPU cost, such abnormal phenomena as failure to detect call or monitoring status may occur sometimes. Therefore, we suggest users optimize the application performance by using the event programming mode to reduce the cost of system resources.

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11. What is the difference between CIC and Ch?

1) In SS7, CIC represents Circuit Identity Code. It should be set based on both parties’ negotiation. The common principle for CIC setting is: for a 2.048kbit/s digital trunk, the 5 lowest bits of CIC represent the voice path time slot number, and the remaining 7 bits represent the number of PCM between DPC and OPC. To be exact, now the relationship between CIC and Ch is as follows:

When the value of CIC is set to 0000 0000 0010 0001, what it specifies are Timeslot 1 on PCM 1 and Channel 0.

2) SpyCic (i.e. Circuit in SpyPCM) indicates a specific time slot (circuit) in the SpyPCM (the monitored PCM). For SS7 signaling, SpyCic is just the CIC field in the TUP or ISUP message; for ISDN PRI and SS1 signaling, SpyCic is the time slot number in PCM.

Each SpyCic has two numbers: the physical number and the logical number. The physical SpyCic number means the SpyCic number in one PCM which corresponds to the time slot number in the same PCM as shown below.

Physical SpyCic Number 0~14 15~29 Corresponding Time Slot Number in SpyPCM

1~15 17~31

The logical SpyCic number is the unified number given to each SpyCic in all PCM involved in the whole application system, beginning with 0. All SpyCic functions provided by the driver take the logical SpyCic number as the input parameter. The mapping relationship between the logical SpyCic number and the physical SpyCic number is determined by the configuration items TotalAppSpyCIC and AppSpyCIC in Section [AppSpyCICTable] in the configuration file. The function SpyGetMaxCic can be used to acquire the total number of SpyCic in the application system.

Because one SpyPCM is bound to two physical PCM, there is an one-to-one correspondence between the logical SpyCic number and the channel number. Logically, one SpyCic is made up of two channels in the driver.

The functions SpyGetCallInCh and SpyGetCallOutCh are used respectively to obtain the logical number of the called party channel and the calling party channel bound to the SpyCic in the current call. The function SpyChToCic can be used to query the corresponding logical number of the SpyCic according to the channel logical number.

12. The application program fails to invoke the function

SsmRecToFile and the returned message says "Error

creating file! ". Why?

Possible reasons are as follows.

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1. Not enough free disk space to create the file;

2. Not enough free HD space to create the file;

3. Too many files already in the folder where to create the file;

4. The driver version is below 5.0.0.0 (Check the driver version and you are suggested to update those below 5.0.0.0 as they use the media function to create a file, which probably results in failure. The driver 5.0.0.0 and above versions change the way to create files. They create a file in CreateFile mode);

5. Use the MP3 format to record data into a *.mp3 file (Such problem may also occur in this case as the file is generated in read-only mode);

6. The fragmentation shown in the Disk Defragmenter is too high;

7. The entire HD has only one partition;

8. Norton AntiVirus, 360, SystemWorks & Internet Security Software and the like are installed in system.

9. Too many file handles are opened under Linux operating system.

If you are sure such problem is caused not by the above reasons, add the function SsmGetLastErrCode to the program to retrieve the error code returned by the driver (refer to MSDN to find more information of error codes) and give it to our technical support for help.

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Appendix A Technical/sales Support

Thank you for choosing Synway. Please contact us should you have

any inquiry regarding our products. We shall do our best to help you.

Headquarters

Synway Information Engineering Co., Ltd

http://www.synway.net/

9F, Synway D&R Center, No.3756, Nanhuan Road, Binjiang District, Hangzhou, P.R.China, 310053

Tel: +86-571-88860561

Fax: +86-571-88850923

Technical Support

Tel: +86-571-88864579

Mobile: +86-18905817070

Email: [email protected]

Email: [email protected]

MSN: [email protected]

Sales Department

Tel: +86-571-88860561

Tel: +86-571-88864579

Fax: +86-571-88850923

Email: [email protected]

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