Recently, I had a chance to work on Sangoma Board and FreeSWITCH ( an open source Telephony platform) which can also work as Interactive Voice Response system. The main challenge

Experience with Sangoma A101 and FreeSwitch

was to install the FreeSWITCH on Windows and get Sangoma board ( Model A101) work. Since I was new to Sangoma board low cost boards, I faced many problems due to lack of information availability on net for beginner.

Though Sangoma support was more or less satisfactory, but no where near the excellent support provided by Dialogic. So, few small things I learned about Sangom board which may be beneficial for people who have worked only with Dialogic board, but not Sangoma boards.

1. Sangoma drivers do not have any telecom protocol ( ISDN ) built into it. It always uses third party software for these protocols.

2. Sangoma boards are mainly made for Asterisk/FreeSWITCH and Linux systems. The basic card is cheap, but additional features like ‘echo cancellation’, NBE ( NetBorder Express – required for Windows) cost extra. So, if some one wants features equivalent of Donjin DN300 in A101, the cost will become more or less same!

3. Configuring the board is more difficult than developing the IVR system for beginner. I could not find any step by step guide for installing the card, run a demo application and test. This kind of utility is available in Dialogic, even in Donjin. Most of the time, Dialogic and Donjin support will help install the card, install the driver, do the configuration and run those demo application and hand over the system back to developer for installing their application. In case of Sangoma, we had to really, really struggled hard to get such support. At first, we were told that in order to get Sagoma board work in Windows, we will have to buy NBE as well as AFT series of board, A101 will not work! But after exchanging mail to and fro with main Sangoma support, we were finally advised to use FreeSWITCH in Windows and it worked for us!

4. Finally, Sangoma support had installed FreeSWITCH on our server and handed over to us. They compiled the recent FreeSWITCH source code along with freeTDM.

After installing FreeSWITCH, developing our IVR using JavaScript was quite easy actually. The were also good documentation about customizing “Dialplan” required to forward any incoming calls JavaScript codes.

So, I would say it was mixed experience for me with Sangoma boards. Using high level language like JavaScript has advantages as well as disadvantages too. But it surely is faster and much easier to learn then learning to write software in C++.

I hope to write a step by step guide to configure Sangoma A101 in Windows using FreeSWITCH some day.

What is IVR short code ?
IVR (Interactive Voice Response) short code can be defined as special telephone number which has ‘less’ digits in it than traditional full length telephone number. This kind of special ‘short code’ are mainly used for special purposes like customer care for telephone/mobile subscribers, VAS ( Value Added Service) access either SMS/MMS ( Short Messaging Service/Multimedia Messaging Service) or IVR based, public utility services etc.
With recent ‘VAS boom’ all around the world in telephone and mobile industry, short-code has become quite well known and many people want to ‘have’ it! While, short code is nothing but telephone number with less number of digits in it, it has become very special special as it is always associated with some special service as well as special charging for calling to it! For any VAS short code, it is charged premium while for any customer care service, it is free of charge for the caller.

Different countries have different policies about short code used for IVR access or SMS/MMS access. Normally any short-code has length of 5 digits and can have customisable suffix for specific service.

Advantages of a short-code

Short codes work as just like any telephone/mobile number. But the main advantage is, it is short so it is easy to remember.
Customer Care numbers are widely publicised by telephone companies and these short codes grow to become brand for any telecom company. Likewise, value added service providers also build their brand around the short-code for the services they provide and try to make them popular. For example, 56969 is SMS as well as IVR shortcode for Rajasthan Patrika Pvt. Ltd. in India which offers many value added SMS and IVR services through this short-code.

Apart from being short, these short-code also tend to be vanity number. For example 58888, 56969, 54321 which have some interesting pattern and easy to remember.

Sine it is short number, it is faster to dial or send SMS. Apart from advantages of being short and faster to dial, I could not think of any other advantages. Many people may think short code as vanity number and associate it with pride.

Disadvantages of Short Code

The main disadvantage of short-codes is it is completely telecom company dependent. Shortcodes are accessible within the home network only. In order to make the short-code accessible in all operators across the country, the short code needs to be configured in all the operators. For example, the short code 56969, it is registered by Rajasthan Patrika Pvt. Ltd. in all mobile operators in India in every circle. That is why it is available in all networks in India. This activity of registering the shortcode in all operators ( in India, we have more than 12 mobile operators) and the allocating software as well as hardware resources to for connectivity to each of them is quite cumbersome and time consuming affair. Most of the time, mobile operators may not allocate the desired short code or simply refuse to allocate one.

But a long code or traditional phone number is accessible by all telecom subscribers without having to register with multiple telecom operators.

Revenue Sharing
Normally, apart from the short codes used for customer care services, all other short codes are used for value added services and so charged premium. The revenue generated are shared by telecom operator and the value added service providers.

Short code for IVR
Short codes are being used IVR portals ( popularly known as voice portal) which provide many value added services or utility services. Popular TV reality shows, Contests generate lot of revenues from this kind of IVR short codes from participants. But this kind of short-code would be accessible across country and all telecom networks only when the short-code is registered with all telecom operators. This need huge initial investment in terms of hardware and software as well as recurring expenses.
So, those who do not want to invest more initially and start any IVR service quickly, they may opt for long code or normal telephone number with E1 connectivity form single operator. It may not be possible to charge callers premium and work like a VAS to generate revenue.

One of the advantages of using Donjin boards is easy portability of C++ codes written for Dialogic boards to Donjin boards! It should not take more than one hour for porting C++ codes from Dialogic boards to Donjin boards. Though, recently introduced KeyGoe series of products from Donjin has totally different API than of Dialogic, but its TDMA CTI boards like DN300, DN600 etc. uses ‘similar’ API as that of Dialogic boards!

Dialogic

Donjin Boards support Dialogic API

Here is the step I used for porting Dialogic C++ codes to Donjin boards in less than an hour:-

1. Header files
Donjin SDK DJNADK is freely down loadable from Donjin website. But it does not contain header files for its LIB files! So, in order to compile your C++ code for Donjin boards, you need to include header files from Dialogic SDK and include LIB files from Donjin. Normally, Dialogic SDK files are located in Program Files\Dialogic\inc while Donjin library files are located in DJNADK\Lib. So, one has to use the correct paths for Include files and then LIB files.
Normally, the below files are needed to be referenced for compiling C++ code for Dialogic Boards
a. libdtimt.lib
b. libdxxmt.lib
c. libgc.lib
d. sctools.lib
e. libfaxmt.lib
f. libsrlmt.lib.
which are found in Lib folder of Dialogic SDK.

And for Donjin boards, the following LIB files need to be referenced :-
a. DNDxxx.lib
b. DNGC.lib
c. DNFax.lib
d. DNSRL.lib
e. DNSCTOOLS.lib
These files are found in DJNADK\Lib folder.

Once these references are made correctly, C++ code gets compiled for Donjin boards !

2. Audio file
Dialogic uses VOX files as well as WAV files for playing back audio files on its boards. While Dialogic supports 8 Bit Mono with 11KHz sampling rate, Donjin supports only 8 Bit Mono with 8KHz sampling rate. So, if your Dialogic code plays WAV files with sampling rate 11KHz, you must change all WAV files to 8KHz sampling rate. It could be a real time consuming job as each files needs to be changed one by one. I have not found any tool/software to change format of WAV files in bulk or all files in a directory yet.

So, once these two processes are complete, your IVR application written in C++ could work in Donjin boards! Many people now days prefer low cost Donjin boards to save initial investment for their On-Premise IVR solutions. The developers who have written Dialogic IVR Application in C++, they can easily port their codes to Donjin CTI boards.

CRBT (Caller ring back tone), also popular as “Hello Tunes” in India, has generated almost 25% of total VAS ( Value Added Service) revenues. This kind of revenue pattern is same worldwide where CRBT has been a major revenue earner for any telephone service provider in its VAS portfolio.

So, what is the next of CRBT ?

If one observes closely about this music or song in mobile industry, one will observe certain trends. Here is how it worked.

Ring tone :
First, popular songs were used a ring tone which was played in the mobile handset as soon as any incoming call came. Not too distant in the past, it was a major revenue source for all involved in it, music producer, Mobile Phone service provider and VAS provider. But with lowering cost of high end mobile phone with MP3 players, and Blue Tooth enabled Mobile phone with easy computer connectivity has almost wiped out Mono Ring Tone from the market. Now, mono ring tones are heard only in very low cost mobile handsets.


RBT ( Ring Back Tone):
Once, Ring Tone revenue was going down, CRBT ( Caller Ring Back Tone) came along! Though the technical term remained RBT ( Ring Back Tone), it was christened with various names in India by various mobile operators, like Hello Tunes, Caller Tunes, PRBT ( Personalised Ring Back Tone) etc. In CRBT, a song was played to the caller by the called party and the called party could configure which song to be played. The called party could even configure specific song for specific phone number based on caller id detected.

So, song used to play on the handset, heard the handset owner in incoming call and could be configured by the handset owner. It graduated to be played on incoming call, but heard by the caller, configured by the called party ( equivalent of the handset owner), now the phone number owner! The caller has not control on the song or music he listens as RBT!

So the next generation CRBT is, listening to music as per your choice while you are calling a phone number! Technically, its implementation is same though!

UPDATE : Tata Docommo in India is already providing this kind of service where caller listens to a song every time hi makes an outgoing call as per his choice. It is powered by ‘Reverse CRBT’ from OnMobile, major VAS company in India.

Dialogic D/300 JCT-E1 is the entry level board for anyone who needs digital network interface with PSTN network as well as voice resources in the single board. I feel, D/300 board is most used entry

Dialogic D/300 JCT-E1 board

Features of Dialogic D/300 JCT-E1 board

1. Digital Network Interface

It has both E1(2.048MBPS) and T1(1.544 Mbps) support. It supports digital interface network protocols like ISDN PRI, R2MF. ISDN PRI is widely used protocol for connecting CTI board with PBX systems. With D/300 board, it is quite easy to configure it to work with many ISDN variants as per PBX from one country to another or manufacturer.

2. FAX support

Dialogic D/300 board supports both FAX and host based speech recognition. In certain cases, FAX and speech recognition may not work together. It also supports CSP ( Continuous Speech Processing) which offers IVR developers a powerful feature to develop many interesting IVR applications and features depending on the call progress. Speech recognition feature provides IVR developers to develop Speech-Enabled menu for any IVR application.

3. Scalability

Dialogic D/300 supports H.100 connector and it allows it to connect to other Dialogic boards. Inexpensive normal PC can be used to install this board and one PC may install upto 16 boards per machine if PCI slots are available.
With its support for GSM and G.726 bit exact voice coders ( standard for Voice Profile for Internet Messaging (VPIM) , one may develop unified messaging solutions based on D/300 board.

4. Programming

Dialogic D/300 board is supported by many RAD Tools for IVR development. It also supports widely popular GC ( Global Call Protocol) for making many high end, mission critical IVR applications.

Applications

D/300 board can be used to develop any kind of IVR applications. It offers many call related information available with digital network interface protocols. Call bridging available in ISDN PRI is one of the feature which is used to make Click To Call application can be developed.

D/300 board is also ideal choice for starting a voice portal service to start with 30 lines and then increase the lines.

DIalogic has some other models with higher capacity with same features. D/600 and DMV1200 have 60 voice and 120 resources respectively. One may move from D/300 to DMV/1200 without having to make any changes in the application.

Other CTI vendors like Donjin, Synway, Digium too have D/300 equivalent boards. For example, Digium TE122P/TE122B, Synway TEJ Series and Donjin DN300 are Dialogic D/300 equivalent boards with similar features.

Call transfer from interactive voice response system to human agent is one of the basic requirement and a must for IVR usability in any call center. While transferring an active call on an analog extension is very simple an straightforward, but the same may not be simple in case of ISDN PRI connectivity with PBX. The method of call transfer vary from one PBX manufacturer to another and that makes it more difficult to implement call transfer facility for a new CTI application developer.

Recently, I had faced similar difficulty while trying to implement call transfer facility on ISDN PRI with a PBX. It took many days, but later I realised that, it is mainly due to technical support from PBX manufacturer end only. In my opinion, PBX manufacturer should provide information how call transfer facility has to be used by a CTI application in their manual itself.

But I have learned a lesson and I thought I would share my experience how call transfer should be implemented using D300 Dialogic board or DN300 Donjin board with a PBX on ISDN PRI connection.

I would suggest these steps:-

1. Discuss with PBX technical support clearly that you are attempting to implement call transfer function on the ISDN PRI link, not on PABX extension line. Call transferring on PBX extension line is quite straight forward, put the phone on ON HOOK for 500 ms ( FLASH time, it may again vary from one PBX manufacturer to another), OFF Hook again, dial the number where call has to be forwarded, then disconnect (put of ON HOOK ) the call. I have faced this strange reply from PBX support as soon as I asked them about call transfer facility in their exchange. If they provide the correct information, then your life has become easy, if not then you would go through the same ordeals which I had gone through.

2. If PBX support provides you with standard call transfer method like SSCT( Single Step Call Transfer) or ECT ( Explicit Call transfer) etc. which are standard and known method, then you can let out a sigh of relief, if not keep your finger crossed! They may simply inform to send some messages on on D channel sequentially and then drop the call! Also, they may send you exact message format! Now you are on your own. But you are near to the solution.

3. If you have not learned how to capture ISDN trace for CTI boards, the time has come. Dialogis has very useful and powerful ISDN tool called ISDIAG.EXE ( normally found in Dialogic\Bin directory) which can be used to capture all messages on D channel and use PRITRACE.EXE in the same channel to decode the messages in simple, human readable English language. Also, capture every error messages while using gc_SndMsg() as well all events from PBX end. If by simply sending message using the Global Call protocol function gcSndMsg(), you are able to transfer call then you have been very lucky, if not continue reading next.

4. Once you have all these trace and logs, you would sit down and study them. I am sure, you would now understand what messages you are sending to PBX and why PBX is rejecting your message or not understanding! Major problem occurs sending Facility Message or Information message! At least, in my case, I was sending a Facility Message to PBX while asking PBX to hold the call, while it was expecting a simple Information message! Normally, you could not expect PBX support guys to give this information, hit and trial is the only method.

5. If you complete the above four steps, you would be able to use any features provided by PBX which is accessible on ISDN PRI! Finally, here is the piece of working code snippet for call transfer on ISDN PRI using a Dialogic D300 CTI board and Coral IRIS PABX (using Global Call Protocol and gc_SndMsg function):-

a. Send Hold call
IE_BLK info;

info.length = 3; // Length of data
info.data[0] = 0x2c; // Keyboard Facility code
info.data[1] = 0×01; // Length of facility
info.data[2] = 0×52; // ASCI R

GC_IE_BLK gc_info;
gc_info.gclib = NULL;
gc_info.cclib = &info;

rc = gc_SndMsg( ldev, crn, SndMsg_Information, &gc_info );

b. Send Number to be dialed Information
ie_blk.length = 6; // length of data
ie_blk.data[0] = 0×70; // called party number code
ie_blk.data[1] = 0×04; // IE length
ie_blk.data[2] = 0×81; // numbering plan number type
ie_blk.data[3] = 0×34; // digit “4″
ie_blk.data[4] = 0×32; // digit “2″
ie_blk.data[5] = 0×35; // digit “5″ 425 is th number where call would be transferred

rc = gc_SndMsg( ldev, crn, SndMsg_Information, &gc_info );

c. Drop the call normally
gc_DropCall(crn,GC_NORMAL_CLEARING,EV_ASYNC);

That is it! This would transfer your call to another PBX extension 425.

I thought I would share it here as I spend considerable time trying to implement call transfer with that particular exchange and on ISDN PRI interface. I got many help from various forums ( specially Dialogic Developer Network) and Dialogic support.

High-End digital CTI boards for a voice portal is not cheap! In fact some of them cost as much as a Honda Civic car in India. So, choosing a right CTI Board for your Voice Portal may not be easy, that is until you read this article .

So, how to choose a CTI Board for your voice portal application ? There are few parameters using which one should choose CTI Board for Voice Portal:-

1. Number of Expected Simultaneous Calls
Most important factor to consider! If you expect simultaneous calls to be not more than four, buy any four channel analog card. But than, most probably it will not remain a voice portal! Ideally, if one should choose a CTI board with 50% more capacity than the expected number of glutenous calls. if some one expects 10-15 calls, he should choose one 30 port Digital CTI board.

2. Time to connect a call
Time taken for connecting a call when an user call is important! It depends of the interface the CTI board supports. For example, CAS R2MF connects much slower than SS7 or ISDN PRI. One will hear “teent teent teent” sound after dialing the Voice Portal before connection. Normally quick and instant connection to Voice Portal is desired.

3. Interface With Switch Support
The type of interfacing with switch of Telecom Provider is important parameter to consider. Many Telecom Service Provider may not have CAS or ISDN protocols on their switch or they may not be willing to provide that connectivity to Voice Portal operators. So, one must consider about available protocol with Telecom Provider before deciding on a CTI board. Interfacing with various protocols is responsible time to connect of calls. SS7 cards are costlier than ISDN PRI or CAS R2MF boards, connects calls almost instantly. And some CTI Manufacturer like Dialogic does not have a combined board which has both Voice Processing and SS7 Signaling so far! This kind of arrangement has its own advantage over combined CTI boards. For example, one can have add more Media Processing (Voice Processing) Boards while keeping one signaling board in one system. Also combined (Signaling and Media Processing) CTI boards do not come in low density configuration, they normally have 8 E1 or more. 4E1 meaning at least 120 simultaneous calls!

[ad#image-large]

3. Scalability and IVR Software
This is another aspect one should not ignore. If the Voice Portal becomes popular, which is the main aim by any Voice Portal Operator, so the CTI Boards should have scalability feature where increasing number of simultaneous calls should not make existing hardware redundant, or make IVR Software obsolete. Separate boards for signaling and media processing is recommended for scalability. IVR Software should be designed such a way that, it can handle more voice resources without having to re-develop the IVR Software.

4. Support for the CTI Board
Availability of development support as well as quick 24 hours support for any problem in the CTI hardware is a must! Every minutes of downtime of Voice Portal, the operator loses revenue! So, one should choose a CTI vendor which can provide quick support or quick replacement of CTI hardware in case of CTI board failure! Also, CTI Vendor should have active forum to discuss various issues and should have good number of third party developers!

5. Cost
Cost is very important, well for everybody! But, one should not compromise of quality of CTI cards for cost.

My Favourite and Recommended CTI hardware:

1. For 30 to 120 simultaneous Calls.
Start with Dialogic D/300 JCT or DMV1200 card. They have ISDN, R2MF protocol, good support, quite tested CTI board. Many developers have written IVR Software on them! Dialogic has good support, active forum, large number of developers. But since they are the biggest CTI vendor, they may not provide you with a demo card for development of your Voice Portal!
These boards can be even used with Dialogic SS7 Signaling boards.

2. For 240 or more simultaneous calls
1. Synway SHD-240D-CT/PCI
Good Value for money, good support. Built in Signaling and media processing. Cheaper than Dialogic equivalent.
2. Dialogic SPCI4/SPCI2 and DNI2410
Awesome combination, great scalability, very good support and large number of developers as well as very active forum. Cost is almost double that Synway though!

Choosing a CTI( Computer Telephony Integration) card might not be easy for them who are going to deploy them for the first time or who want to develop their first IVR Application. I am trying to give a short introduction of CTI cards and some thumbs rule to pick your first CTI card.

Number of Port Or Number of Channels:
CTI cards come in number of channels or ports! It means number of telephone lines you can connect to the CTI card and number of calls your IVRS will be able to handle simultaneously. So for development purpose, you might be interested in a single port or single channel! But such is life, you will not find any single port CTI cards! The minimum number of ports/channels in a CTI card is normally 4! But then it makes sense! In a real life and real world situations, single port IVRS will be of no use! Even in development, if you could get a single port card to test, how could you make an application which would handle two or more channels/ports? So, you have to buy a 4PORT CTI cards! Be prepared, if you ask any vendor for a CTI Card, first thing they would ask is how many ports or channels! Smartly reply that you are looking for a 4Port card!

Analog or Digital
CTI card should be analog or digital ? well, this is little confusing, it actually means CTI card will connect to Telephone Exchange ( EPABX) using digital technology or analog technology ? By analog technology, we mean a telephone line ( which you connect to a telephone instrument) connectivity to the CTI card. How is it analog ? well, it uses TIP and RING connection, you know there is a call on the phone when there is a RING, and you UNHOOK, or lift the telephone headset to connect the call! You know the caller has disconnected when you hear engage tone! (I hope you know what is and engage tone). So, the analog CTI card detects the RING, unhooks the phone, and it detects ENGAGE TONE and it ONHOOK the phone to disconnect, puts the headset on the cradle! So all these activities are analog and takes time! And also there is limitation of number of channels! But in a digital connectivity, all those like RING, Disconnect Tone(engage Tone), everything comes in messages! And they use digital links. And all these jargons like E1/T1, ISDN PRI, R2MF, SS7 comes in! These are digital protocols to connect the CTI t the telephone exchanges! And the minimum number of channels in a DIGITAL CTI card is 30 voice ports! It uses a 2MBPS link. The number of voice channels will go up to 31 if you use a SS7 protocol on the same 2MBPS link (commonly known as E1 or PCM). So, the thumb rule is settle for analog card if you number of channels requirement is less that 20. Else go for a digital card.

[ad#image-large]

Development SDK and Tools
Most important part for developers is, what SDK(Software Development Kit) the CTI vendor provides and on what platforms! Normally one should opt for a CTI cards who has SDKs for MS Windows, UNIX/LINUX and also Sun Solaris! Though configuration for Analog cards are normally pretty easier than their digital counterparts! So, if you are a new in the field, technical support is a must from the vendor! And if you have to go for a digital card, ahem, you choose only a CTI Card vendor who has a very good technical support! Else you are asking for a trouble!

Price
Well, you will have to negotiate with the vendor!