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Kolmisoft 2006-2012 1 It is IMPORTANT to note that this manual is compilation of articles from our main data source: http://wiki.kolmisoft.com and current info in this manual can be outdated. In order to get newest info please use online wiki.

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Page 1: MOR 12 Manual

Kolmisoft 2006-2012

1

It is IMPORTANT to note that this manual is compilation of

articles from our main data source: http://wiki.kolmisoft.com and

current info in this manual can be outdated.

In order to get newest info please use online wiki.

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Contents Introduction ..................................................................................................................................... 6

Who is this product designed for ...................................................................................................... 6 Supported Business Models ............................................................................................................. 7 How to make first call ..................................................................................................................... 8

Concepts/Definitions ....................................................................................................................... 14 MOR Terminology ......................................................................................................................... 14 Billing Concepts ........................................................................................................................... 18 Billing Logic ................................................................................................................................. 19 Call explained .............................................................................................................................. 20 E.164 ................................................................................................................................. 21 Rounding ................................................................................................................................. 21 Multi-Tax system ......................................................................................................................... 21 What is a PIN? ............................................................................................................................. 24 ASR/ACD ................................................................................................................................. 25 Margin and Markup ...................................................................................................................... 25 What are callgroups and pickupgroups ............................................................................................ 26 Balance and Credit ....................................................................................................................... 27 Why amounts are without VAT? ..................................................................................................... 28 Local Calls ................................................................................................................................. 28 Prepaid Logic ............................................................................................................................... 30 Authentication ............................................................................................................................. 32 Fax over VoIP .............................................................................................................................. 34

Configuration from GUI ................................................................................................................... 36 ReCAPTCHA................................................................................................................................. 43 Payments configuration ................................................................................................................ 48 Default device settings ................................................................................................................. 54 Default user ................................................................................................................................ 56 Accountant permissions ................................................................................................................ 57 Logo change ................................................................................................................................ 59 Currencies ................................................................................................................................. 60 Translations ................................................................................................................................ 63 Global date format ....................................................................................................................... 64 Remove Manual link from admin GUI .............................................................................................. 65 Voicemail ................................................................................................................................. 65 Email settings .............................................................................................................................. 70

Number Manipulation ...................................................................................................................... 76 Localization ................................................................................................................................. 76 Provider Rules ............................................................................................................................. 82 DID handling by Localization ......................................................................................................... 85

Users ............................................................................................................................................ 93 User Details ................................................................................................................................ 93 Warning balance .......................................................................................................................... 95 User Blocking .............................................................................................................................. 96 Blocking ................................................................................................................................. 98 Hide Users ................................................................................................................................ 100 Why MOR does not allow to delete users ........................................................................................ 102 How to create User and Device ..................................................................................................... 103 How to change Admin details ........................................................................................................ 103 MOR forgot user password ........................................................................................................... 104 Minimum contract billing .............................................................................................................. 106

Tariffs .......................................................................................................................................... 107

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Directions and Destinations .......................................................................................................... 107 Destinations Groups .................................................................................................................... 108 Block some Destination to User .................................................................................................... 109 Day setup ................................................................................................................................ 112 Entering Rates for Wholesale Tariff ................................................................................................ 113 Rate import from CSV.................................................................................................................. 117 Wholesale Tariff .......................................................................................................................... 127 Entering Rates for Retail Tariff ...................................................................................................... 129 Advanced Rates .......................................................................................................................... 131 Custom Rates ............................................................................................................................. 134 Tariff Transformation ................................................................................................................... 135 Make Retail Tariff from Wholesale Tariff explained ........................................................................... 136

Devices ........................................................................................................................................ 138 H323 Device settings ................................................................................................................... 145 Extension lines ........................................................................................................................... 146 Ring several devices at same time ................................................................................................ 146 How to create User and Device ..................................................................................................... 147 Trunks ................................................................................................................................ 147 CLIs (Incoming CallerIDs) ............................................................................................................ 148 ANI/CLI ban system .................................................................................................................... 149 Duplicate call prevention .............................................................................................................. 152 Allow loss calls ........................................................................................................................... 152 Device groups ............................................................................................................................ 153 Call Flow ................................................................................................................................ 154 Which codecs should I choose for devices ...................................................................................... 160 Change Audio/Video Codecs priority .............................................................................................. 163 Authentication ............................................................................................................................ 164

Providers ...................................................................................................................................... 173 Authentication ............................................................................................................................ 175 Failover provider ......................................................................................................................... 178 Terminators ............................................................................................................................... 179 Provider with several IP ............................................................................................................... 179 Provider connection testing .......................................................................................................... 180 H323 Provider settings ................................................................................................................ 181 Provider with ANI ........................................................................................................................ 181 Configure Provider which can make calls ........................................................................................ 183 Configure Provider with dynamic IP ............................................................................................... 183 Simultaneous call limitation .......................................................................................................... 185 NO ANSWER/BUSY interpretation for providers ............................................................................... 188

Call Routing .................................................................................................................................. 209 LCR ................................................................................................................................ 209 LCR Logic ................................................................................................................................ 211 Call Routing by price ................................................................................................................... 213 Call Routing by priority (Manual LCR) ............................................................................................ 214 Call routing per destination basis .................................................................................................. 216 How to route call to specific Destination through specific Provider ..................................................... 219 LCR/Tariff change based on call prefix ........................................................................................... 220 Routing by Percent ...................................................................................................................... 221 Why call was not routed through other provider/trunk? ................................................................... 224

DIDs ............................................................................................................................................ 226 DID Management ........................................................................................................................ 226 Assign DID to Trunk .................................................................................................................... 236 Work flow for DID ....................................................................................................................... 238

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DID Billing ................................................................................................................................ 239 DIDs Usage ................................................................................................................................ 243 Personal DIDs ............................................................................................................................. 243 DID with VoxBone using his URI ................................................................................................... 248 Forward DID to External Number .................................................................................................. 249 How to charge DID on a monthly basis .......................................................................................... 249 DID handling by Localization ........................................................................................................ 250 DID Blocking .............................................................................................................................. 251 Dial Plans ................................................................................................................................ 252 Authorization by PIN ................................................................................................................... 253 Authorization by ANI ................................................................................................................... 256 Calling Card Dial Plan .................................................................................................................. 259 Callback Dial Plan ....................................................................................................................... 261 IVR Dial Plan .............................................................................................................................. 263 STATISTICS -> Various –> DIDs .................................................................................................. 263

Accounting .................................................................................................................................... 264 Invoices ................................................................................................................................ 269 Credit notes ............................................................................................................................... 280 Payments ................................................................................................................................ 283 Vouchers ................................................................................................................................ 286 Flat-Rates ................................................................................................................................ 289

Reporting...................................................................................................................................... 290 Active Calls ................................................................................................................................ 290 Last Calls ................................................................................................................................ 295 Users Calls ................................................................................................................................ 297 Users Finances ........................................................................................................................... 298 Loss Making Calls ........................................................................................................................ 298 Providers Statistics ...................................................................................................................... 299 Providers Calls ............................................................................................................................ 300 First Activity ............................................................................................................................... 301 Quick Stats ................................................................................................................................ 301 Simultaneous Calls stats .............................................................................................................. 304 Calls by Source ........................................................................................................................... 305 Login Stats ................................................................................................................................ 306 Profit Stats ................................................................................................................................ 307 Country Stats ............................................................................................................................. 307 Google Maps integration .............................................................................................................. 309 Balance in phone ........................................................................................................................ 310 Privacy to hide Destination ends ................................................................................................... 312

Troubleshooting ............................................................................................................................. 314 Action log ................................................................................................................................ 314 User actions ............................................................................................................................... 317 Call Tracing ................................................................................................................................ 317 Integrity Check ........................................................................................................................... 325 Hangupcause Codes .................................................................................................................... 327 Hangupcause Sounds .................................................................................................................. 329 Different code send to User compared to received from Provider ....................................................... 330 Call Info ................................................................................................................................ 331 SIP debug info ............................................................................................................................ 333 Asterisk CLI ............................................................................................................................... 334 MOR's Core version ..................................................................................................................... 336 Lots of calls made by System Admin user ...................................................................................... 336

Maintenance ................................................................................................................................. 337

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Upgrade MOR GUI ....................................................................................................................... 338 Lost MOR GUI admin password ..................................................................................................... 338 Where I can get my serial key from ............................................................................................... 339

Testing ......................................................................................................................................... 339 VoIP Bandwidth Calculator ........................................................................................................... 339

Payment gateways ......................................................................................................................... 340 Other functionality ......................................................................................................................... 351

IVR system ................................................................................................................................ 351 IVR Greeting for DID ................................................................................................................... 359 Voicemail ................................................................................................................................ 361 Dial Local ................................................................................................................................ 366 PBX Function External DID ........................................................................................................... 370 Ring-Groups ............................................................................................................................... 372 Fax2Email ................................................................................................................................ 375 Recordings ................................................................................................................................ 379 PhoneBook ................................................................................................................................ 381 Speed Dials ................................................................................................................................ 383 Backup system ........................................................................................................................... 384 Data import ................................................................................................................................ 386 CDR Import ................................................................................................................................ 394 CDR Rerating ............................................................................................................................. 398 Online registration ...................................................................................................................... 400 Online registration customization .................................................................................................. 403 Quick Forwards ........................................................................................................................... 403 Quickforwards rules..................................................................................................................... 406 Transfers with MOR ..................................................................................................................... 408 GUI on Mobile Devices ................................................................................................................. 409 PBX connection to MOR ................................................................................................................ 410 MOR API ................................................................................................................................ 421 Cron actions ............................................................................................................................... 426

MOR Addons ................................................................................................................................. 429 Calling Cards Addon .................................................................................................................... 429 Callback Addon ........................................................................................................................... 469 Auto-Dialer Addon ....................................................................................................................... 482 Reseller Addon ........................................................................................................................... 488 Reseller Pro Addon ...................................................................................................................... 494 Manual for Resellers .................................................................................................................... 497 SMS Addon ................................................................................................................................ 509 Recordings Addon ....................................................................................................................... 511 Mobile Number Portability Addon .................................................................................................. 516 Payment Gateway Addon ............................................................................................................. 518 Monitorings Addon ...................................................................................................................... 543 Call Shop Addon ......................................................................................................................... 554 Skype Addon .............................................................................................................................. 572 Provider billing addon .................................................................................................................. 583

Tutorials ....................................................................................................................................... 586 Usual configuration sequence ....................................................................................................... 586 How to configure 911 service ....................................................................................................... 591 How to make several connections from one IP ................................................................................ 593 How to forward calls when device is offline ..................................................................................... 597

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Introduction

Kolmisoft VoIP Softswitch with Billing and Routing MOR is an advanced system that enables

alternative telecoms to provide flexible voice services, effectively manage service offerings

and grow their voice businesses due to the systems stability and high traffic scalability.

Kolmisoft System MOR integrates the core AAA functions Authentication, Authorization, and

Accounting with vital functions that facilitate the effective management of all billing/routing-

related processes:

Customers

Services

Finance

Services distribution

Reporting

Traffic partners

User rights

On the business side, Kolmisoft System MOR makes possible the deployment of services with

high-quality and flexibility within any VoIP business model; on the technical side the system

maintains seamless performance at high and rapidly increasing traffic volume. The optimal

combination of functions, the adaptability to every VoIP business model and rate-plan

settings, as well as its stability, make Kolmisoft System MOR the absolute billing/routing and

customer care solution for Voice Over Broadband/ IP businesses worldwide.

Who is this product designed for

Kolmisoft System MOR is designed for premium performance and scalability; therefore, it is

suitable for VoIP or Broadband service providers who:

plan for a full-scope VoIP business deployment

need to maintain medium, high or very high voice traffic volumes

differentiate on a first-class customer service

demand high reliability from their billing system

enjoy rapid business development and have outgrown their legacy billing systems

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Supported Business Models

Kolmisoft System MOR is adaptable to telephony service providers' business models and

custom requirements for service configuration. Voice-over-IP and Voice-over-Broadband

providers can configure different service types within all of the following basic business

models:

Wholesale traffic origination/termination

Retail

Prepaid

Postpaid

Calling Cards

Corporate Clients

Residential Clients

Internet services

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How to make first call

This small guide will show how to make your first call after

system installation.

Create a tariff

Tariffs are a set of rates (prices) to different destinations.

Go to SETTINGS -> Billing -> Tariffs and click New tariff

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Enter the name (you will use it to recognize tariff (price list) in variuos system places). Choose

currency in which you will pay for providers/suppliers. Select what type of tariff do you want

and click Create.

In this example tariffs have rates/prices from your Providers. These prices are paid TO

Providers when they terminate calls FROM your system to PSTN. This is self-cost for you, e.g.

you buy calls using these prices, as it is written next to the blue information icon.

Rates

However if you choose other type than Providers you will have to either import rates from

CSV or fill them in manually by clicking Rates -> New rate. The most important fields in

Rates are Prefix and Rates.

Create a provider

Go to SETTINGS -> Billing -> Providers and click New provider

Select Technology, from a dropdown list assign Tarrif you have created, select the server and

click Create.

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Create a LCR

Go to SETTINGS -> Billing -> LCR and click New LCR

Enter the name. From a dropdown list select the order. If you select order of providers by

price the cheapest way to dial a destination will be used. If you select by priority the call will

be routed by a predetermined order. Routing by percent is similar to routing by priority. After

you selected click Create.

You will see that there are no providers assigned to this created LCR.

Click Providers.

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Assign created provider from the dropdown list and click Add.

Please note that Save changes button is only for failover providers, not for providers in

general.

Adding a user

To add a user go to SETTINGS -> Users. Click Add user.

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Assign created LCR and click Create at the bottom of the page.

Add a device for user

To add a device go to SETTINGS -> Users and in the users list click the devices icon ( ). In

the opened window click Add new device.

Description - it is recommended to change it so that it will be easier to recognize it anywhere

in the system.

PIN - you can leave it as it is or enter a new one.

Type - choose SIP type.

Click Create.

Softphone

To make a real call, you will need a softphone for a quick test.

You can use your own if it supports the SIP or IAX2 protocols.

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For the sake of simplicity, we will show how to configure Idefisk 1.37 , which is very small and

easily configurable.

First, download it from our server (http://www.kolmisoft.com/packets/idefisk137

_installer.exe) and install it. (It is for Windows OS.)

It can be downloaded for Linux (http://www.zoiper.com/zclinux.php) and for MacOS

(http://www.zoiper.com/downloads/free/mac/communicator/ZoiperCommunicatorFree-

1.04_installer.dmg).

Use Idefisk configuration manual to configure Idefisk with your <SERVER_IP> and

username/password of test user's device.

NOTE: that User 101 is by default Blocked to insure maximum security to your system.

Unblock it before making a test call.

Dialing

Next, enter a phone number into your softphone and try to dial.

NOTE: enter the number in international (E.164) format. Do not use any international prefixes

- use country code + area code + number.

For example, a Lithuanian number is 37063042439, where 370 is the country code, 6 is the

area code (MOB network), and all the rest is the number itself.

NOTE: USA/Canada numbers should start with 1 (which is the USA/Canada country code).

If you hear nothing, your softphone is not registered to the server.

If you hear a message from Kolmisoft, you do not have an account on our server

(contact us (http://www.kolmisoft.com/contacts) to get one).

If you hear ringing, everything is okay.

Call logs

After the call, you can check your call in the system.

Go to: STATISTICS -> Calls -> Last Calls and you will see your call.

It will be already billed.

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Concepts/Definitions

MOR Terminology

1. Accountant [http://wiki.kolmisoft.com/index.php/Accountant_permissions] – a type of user

which has more rights than simple user in MOR, but still not enough to fully manage all

configuration of MOR system. His rights can be adjusted at any time.

2. Action log [http://wiki.kolmisoft.com/index.php/Action_log] – detailed report about user

activity.

3. Addons [http://wiki.kolmisoft.com/index.php/MOR_Addons]– a special feature group for

MOR which adds additional functionality.

4. Admin [http://wiki.kolmisoft.com/index.php/Users] – most powerful user account in MOR

(just like root in UNIX systems).

5. Aggregate [http://wiki.kolmisoft.com/index.php/MOR_Manual#Reporting] – advanced

report on calls.

6. API [http://wiki.kolmisoft.com/index.php/MOR_API] - application programming interface is

an interface implemented by a software program to enable interaction with other software,

similar to the way a user interface facilitates interaction between humans and computers.

7. Asterisk [http://en.wikipedia.org/wiki/Asterisk_(PBX)]- PBX free software, multi platform,

published under license GPL per Spencer Mark of the Digium company.

8. Auto Dialer [http://wiki.kolmisoft.com/index.php/Auto-Dialer_Addon] – this addon allows

to call an array of numbers automatically and when answered plays back a prerecorded

message to the receiving end of the call.

9. Backup [http://wiki.kolmisoft.com/index.php/Backup_system] - a backup or the process of

backing up refers to making copies of data so that these additional copies may be used to

restore the original after a data loss event.

10. Browser title – most top of web browser.

11. Callback [http://wiki.kolmisoft.com/index.php/Callback] – in telecommunications, a

callback (also written as call-back) occurs when the originator of a call is immediately called

back in a second call as a response.

12. Call tracing [http://wiki.kolmisoft.com/index.php/Call_Tracing] – used for checking the

setup of MOR in order to be sure user will be able to dial out to specified destination number.

13. Calling cards [http://wiki.kolmisoft.com/index.php/Calling_Cards_Addon]– an addon to

MOR billing which adds Calling Card functionality.

14. CDR – call data (detail) records are computer records produced by a

telephone exchange containing details of a call that passed through it.

15. Click2Call [http://wiki.kolmisoft.com/index.php/Click2Call] – the technology which virtually

always refers to a means for a web consumer to ask a website operator to call her. In

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other words, "clicktocall" would be more accurately be referred to as "RequestACall,"

as in "Please give me a call (here is my phone number)."

16. Default device [http://wiki.kolmisoft.com/index.php/Default_device_settings] - when you

configure default device's parameters, next time when you try make new device for someone,

the newly device will have ―default device's‖ parameters already.

17. Default user – when you configure default user's parameters, next time when you try to

make new user, he already will have them.

18. Destination – destination helps identify country and city and destination number stands

for final number which will be sent to provider.

19. Devices [http://wiki.kolmisoft.com/index.php/Devices] - represents endpoint which can

generate (originate) or receive (terminate) calls. As example device can represent: VoIP

phone, VoIP softphone, PSTN phone (connected through FXS port), PBX, softswitch, etc.

20. Dial Plan [http://wiki.kolmisoft.com/index.php/Dial_Plans] - a dial plan establishes rules

how to handle some call.

21. DIDS [http://wiki.kolmisoft.com/index.php/MOR_Manual#DIDs] - Direct Inward Dialing is a

feature offered by telephone companies for use with their customers' PBX systems, whereby

the telephone company allocates a range of numbers all connected to their customer's

PBX. As calls are presented to the PBX, the number that the caller dialed is also given, so the

PBX can route the call to the desired person or bureau within the organization.

22. Direct link or Direct URL [http://wiki.kolmisoft.com/index.php/What_is_direct_URL] – a link

to specified place in WEB.

23. Direction [http://wiki.kolmisoft.com/index.php/Directions_and_Destinations] - usually

refers to Countries. Could also be satellite networks. They describe direction where the call

will be routed.

24. Domain – usually meaning of second level domain, for example: example.com (com is first

level and example is second level)

25. Domestic – meaning to describe ―calls inside country‖ or in other words ―local calls‖.

26. FAX [http://wiki.kolmisoft.com/index.php/Fax2Email] - a fax (short for facsimile) is a

document sent over a telephone line. FAX implementation in VOIP is very bad and currently it

is still not possible to send/receive fax in VOIP at 100% succession rate.

27. Firewall – is a part of a computer system or network that is designed to block

unauthorized access while permitting authorized communications. Poorly configured firewall

is very often the cause to failed calls.

28. Frontpage text – first page in MOR GUI displayed for successfully logged in user or admin.

29. Graph – function which generates visual data sets.

30. HGC [http://wiki.kolmisoft.com/index.php/Hangupcause_Codes] – substitution for

―hangup cause code‖, those codes used to identify status of all made calls through MOR.

31. Integrity check [http://wiki.kolmisoft.com/index.php/Integrity_Check] – a place with

various properties of billing system which tells where potential problems could exists.

32. Invoice [http://wiki.kolmisoft.com/index.php/Invoices] - a document issued from a vendor

for payment, detailing the goods or services the vendor has provided to the buyer, the

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cost(s), and any VAT due. The document may be called a 'Sales Invoice', from the point of

view of the vendor, or a 'Purchase Invoice' by the buyer.

33. IVR [http://wiki.kolmisoft.com/index.php/IVR_system] - Interactive Voice Response (IVR) is

a technology that allows a computer to detect voice and dualtone multi-frequency signaling

(DTMF) keypad inputs. Most often usage of IVR in VOIP is for Virtual menus.

34. Last Calls [http://wiki.kolmisoft.com/index.php/Last_Calls] – place in MOR GUI where you

can see detailed list of last billed calls.

35. LCR [http://wiki.kolmisoft.com/index.php/LCR] – stands for Least Cost Routing is the

process of selecting the path of outbound communications traffic based on cost (cheapest

price), order (always route calls to A, but if A fails, then route to B, if B fails, route to C etc.) or

percent (used for load balancing, for example: send 50% of call to provider A and rest 50% to

B).

36. Localization [http://wiki.kolmisoft.com/index.php/Localization] – technology used by MOR

which transforms received number to E.164 format number.

37. LOGO [http://wiki.kolmisoft.com/index.php/Logo_change] – your company's logo.

38. MOR [http://www.kolmisoft.com/billing-and-routing/mor-voip-billing-features] –

softswitch with billing and routing made by Kolmisoft Ltd.39. Number digits - how many

decimal digits will see all system users. You can change this setting in Settings.

40. Originate - initialize call.

41. PBX [http://en.wikipedia.org/wiki/Private_branch_exchange] - small version of the phone

company's larger central switching office. A PBX is a private telephone switch. It is connected

to groups of lines from one or more central offices and to all of the telephones at the location

served by the PBX.

42. PhoneBook [http://wiki.kolmisoft.com/index.php/PhoneBook] – place containing an

alphabetical list of telephone numbers.

43. Prefix - refers to the first digits of a telephone number. These digits are a way to identify

the area that the phone number is located in.

44. Profit [http://en.wikipedia.org/wiki/Profit_(accounting)] - amount expressed as a value that

is left over from sales once all expenses have been deducted.

45. Provider [http://wiki.kolmisoft.com/index.php/MOR_Manual#Providers] – meaning for

VOIP provider. Usually company sells voice traffic.

46. Rate [http://wiki.kolmisoft.com/index.php/Tariffs] – a price for destination.

47. ReCAPTHA [http://www.google.com/recaptcha] – a mechanism which helps to protect

websites from automatic registrations.

48. Recording [http://wiki.kolmisoft.com/index.php/Recordings] – voice file where you can

hear conversation of your customer and remote end (where he called).

49. Registration [http://wiki.kolmisoft.com/index.php/Online_registration] – place in MOR

where new users could register themselves without help from MOR admin.

50. Rerating [http://wiki.kolmisoft.com/index.php/CDR_Rerating] – recalculation of the call's

price which is already recorded.

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51. Reseller Addon [http://wiki.kolmisoft.com/index.php/Reseller_Addon] – this addon

incorporates basic reseller functionality support into MOR system which adds opportunity

MOR system owner have reseller support.

52. Retail tariff [http://wiki.kolmisoft.com/index.php/Advanced_Rates] – with retail tariff there

is possibility get more control of how call will be billed, let say make first 15 seconds of call

free.

53. Router [http://en.wikipedia.org/wiki/Router] - a device that forwards data packets from

one local area network (LAN) or wide area network (WAN) to another. Poorly configured

device also leads to voice and traffic problems.

54. Service - product which can be sold to the client. That can be various types of services. For

example: monthly fee, line rent, phone rent, internet plan.

55. Site – your MOR website.

56. SMS Addon [http://wiki.kolmisoft.com/index.php/SMS_Addon] – this addon adds SMS

sending option through and from MOR.

57. SSH [http://en.wikipedia.org/wiki/Secure_Shell] – secure shell for completely controlling

server from remote location.

58. Subscription - users can subscribe for some services. When they do so – they get

Subscription for some service.

59. Supplier [http://wiki.kolmisoft.com/index.php/Providers] – see meaning of Provider.

60. Terminate – end call in specified end.

61. Terminator [http://wiki.kolmisoft.com/index.php/Providers] – a final destination point

where call be terminated.

62. Trunk [http://wiki.kolmisoft.com/index.php/Trunks] - other server/PBX which can be

connected to the MOR system.

63. User [http://wiki.kolmisoft.com/index.php/Users] – your customer.

64. Voicemail [http://wiki.kolmisoft.com/index.php/Voicemail] - Voicemail (or voice mail,

voicemail, vmail or VMS, sometimes called message bank is a centralized system of

managing telephone messages for a large group of people. The term is also used more

broadly, to denote any system of conveying voice message, including the answering machine.

65. VOIP [http://en.wikipedia.org/wiki/VOIP] - general term for a family of

transmission technologies for delivery of voice communications over IP networks such

as the Internet or other packet-switched networks.

66. Voucher [http://wiki.kolmisoft.com/index.php/Vouchers] – this function allows to enter

voucher number over phone and top up user's balance.

67. Web callback [http://wiki.kolmisoft.com/index.php/WEB_Callback] - technology where a

person can enter his or her telephone number in a form on a web site.

68. Wholesale tariff [http://wiki.kolmisoft.com/index.php/Wholesale_Tariff] – in wholesale

tariff you can set price for each prefix, while in retail you cannot, but retail has its own

advantages.

69. ZAP/Zaptel series of drivers for telephony hardware devices (later renamed to DAHDI).

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Billing Concepts

This section deals with the billing system. It breaks down the actual call-flow from user to

Provider(s).

The billing system is quite complex and there are a lot of concepts to grasp.

There are a number of main categories of billing in MOR. The most important are:

Providers

Least Cost Route (LCR)

Directions

Destinations

Rates

Tariffs

Users

Devices

Providers - In the old PSTN world, these were called Trunks. They are your routes to the

outside world (out of your VoIP network). Providers can be of several types: Zap, SIP, IAX2,

and H323. The type determines which technology is used to connect to the Provider.

Least Cost Route (LCR) - LCR describes how calls may go through several Providers and the

order in which they go. The order can be determined by the call route's price, quality or

manual priority.

Directions - This term refers to countries, or perhaps to satellite networks. It simply describes

the direction in which the call is routed.

Destinations - Destinations are the specific steps that make up the course of the call. Let's

say we are dialing number 37068547771. From this number, MOR knows that the Direction is

Lithuania, and the Destination is the mobile network Bite. Destinations are determined by

Prefix, and each destination has a unique Prefix. The Prefix in this example is 370685.

Rates - These are the prices for the Destinations. They have many properties, which

determine they way they are calculated.

Tariffs - These are groups of Rates. Sometimes they are called Price Lists.

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Users - This term refers to the actual user of the system – the client who dials out or receives

calls. ―User‖ may represent a single person or a whole company, but it is main unit in the

system. Most of the settings are associated with Users.

Devices - Each user can have many devices. A device represents the end-point to which voice

traffic should be sent. For example, a device might be a:

VoIP phone

VoIP softphone

PSTN phone (connected through FXS port).

Billing Logic

In order to use MOR successfully for dialing out and billing, some rules must be followed.

The best way to show this is by example.

User A dials a number, trying to reach User B.

The system then takes all Providers that belong to that LCR.

The system takes the Tariff of every Provider.

For every Provider system takes their Tariff.

From every Tariff (based on Prefix), the Rate is taken.

If the LCR type is 'Price', then all the Providers are sorted to identify the Provider with

the lowest Rate to that Destination. This Provider becomes the first Provider.

Using the first Provider, the system tries to dial the number.

If the call is successful, the system does the billing and finishes the session.

If the call is unsuccessful, the system tries to dial the next Provider.

The system goes on calling until the call is successful or no Providers are left.

When the call ends, the system bills it.

This example provides only a basic illustration of how the billing system works. The whole

process in MOR is much more complex.

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Call explained

Duration - the time from when the caller enters the number until the end of the call.

Billsec - the time from when the callee answers until the end of the call.

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E.164

What is an E.164 format number? In short, it's a number without an international prefix that

starts with the country code. For further details, check Wikipedia

[http://en.wikipedia.org/wiki/E.164].

Every destination inside MOR should be in the E.164 format. That is one of the most

important rules of MOR. If the destination is not in the E.164 format, it can't be billed

correctly.

All DIDs in MOR should also be in this format. Do not include international prefixes for DIDs.

Enter them in E.164 format.

In Australia, for instance, the local number 08 6311 3000 would be converted under E.164 to

61863113000. More information on how MOR works with E.164 can be found at Number

Manipulation.

Rounding

Rounding in MOR GUI is done only when showing the final result.

Values are not rounded in the database. Why not? For example: 0.027 + 0.027 = 0.054.

Rounded, this comes to 0.05. But if you are rounding each step, then you get: 0.03 + 0.03 =

0.06. This figure is not so accurate as 0.05.

Multi-Tax system

All prices and rates in MOR are without TAXES. That's a rule. Each user has a TAX percent. The

user is charged by this TAX percent when he makes payments or views his rates.

The Multi-Tax system provides for several taxes in the final reports for users.

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Compound Tax and Simple Tax

Let's say we have taxes T1, T2, T3 and T4. And some amount without tax, X.

When Compound tax is checked, multiple taxes are calculated in following way:

Amount with Compound Taxes = (((X + T1%) + T2%) + T3%) + T4%

If Compound tax is disabled, then:

Amount with Simple Taxes = X + (T1% + T2% + T3% + T4%)

Example for Compound/Simple tax calculation

We have initial data:

SUBTOTAL 3.99

TAX1: 18%

TAX2: 15%

Calculating when Compound tax is enabled:

Amount with Compound tax = (SUBTOTAL + TAX1) + TAX2

in digits:

(SUBTOTAL + TAX1) = 3.99 + 18% = 3.99 + 3.99 / 100 * 18 = 3.99 * 1.18 = 4.7082

(SUBTOTAL + TAX1) + TAX2 = 4.7082 + 15% = 4.7082 + 4.7082 / 100 * 15 = 4.7082 +

0.70623 = 4.7082 * 1.15 = 5.41443

Amount with Compound tax = 5.41443

Calculating when Compound tax is disabled (Simple tax):

Amount with Simple tax = SUBTOTAL + (TAX1 + TAX2)

in digits:

SUBTOTAL + (TAX1 + TAX2) = 3.99 + (18% + 15%) = 3.99 + 33% = 3.99 + 3.99 / 100 * 33 =

3.99 + 1.3167 = 3.99 * 1.33 = 5.3067

Amount with Simple tax = 5.3067

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Example for Canada

As an example we can take Canada, which has GST and PST taxes that should be shown on

invoices, reports, and so on:

Open SETTINGS –> Setup –> Settings –> Tax

Enter Total Tax name (i.e. Tax)

Check Compound tax

Enter Tax 1 name (i.e. GST) and value (i.e. 5.0%)

Check Active? near Tax 2

Enter Tax2 name (i.e. PST) and value (i.e. 7.0%)

Click Save changes

Taxes are applied to all newly created Users:

and Calling Cards:

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If you change taxes, you can assign this to all Users/Calling Cards by pressing the buttons

below:

Apply to all Users

Apply to all Calling Cards

Apply to all Vouchers

What is a PIN?

Description

A PIN is a Personal Identification Number used to authenticate Devices or Calling Cards in

MOR.

The followings are the usual scenarios where a PIN is used:

• A user dials an Access Number (DID).

• The system asks the user to enter a PIN.

• The user enters the PIN.

• The system checks the database for the PIN.

• If the PIN is found, the system asks for the Destination.

• The user enters the Destination.

• The system calls the Destination.

• The call is connected.

How to get a PIN

For Devices, it can be done in the Devices window.

Calling Cards

An example of how a PIN is used in Calling Cards can be found here: Calling Card logic.

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ASR/ACD

ASR - Answer-Seizure Ratio

The Answer-Seizure Ratio (ASR) is calculated by dividing the number of successfully answered

calls by the total number of calls attempted, which are known as "seizures". Since busy signals

and other rejections by the called number count as call failures, the calculated ASR value can

vary depending on user behavior. (http://www.voip-info.org/wiki/view/ASR)

60-70% is considered a very good ASR in the VoIP world.

ACD – Average Call Duration

The Average Call Duration (ACD) is calculated by taking the sum of billable seconds (billsec)

of answered calls and dividing it by the number of these answered calls.

Margin and Markup

If we have BUY and SELL prices, then:

PROFIT = SELL - BUY.

Margin = PROFIT / SELL x 100.

Markup = SELL / BUY x 100 - 100.

In other words:

Margin = PROFIT's part in the SELL price, or the part profit makes up in the sell price.

Markup = the increase from the BUY price to the SELL price, or how much the SELL

price has increased from the BUY price, in percent.

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What are callgroups and pickupgroups

Letting someone else answer a call

In the mgcp, SIP, IAX, and Asterisk Skinny Channels and in the zapata channels, you can

define call and pickup groups for phones. Note that call pickup typically only works WITHIN a

technology, e.g. from a SIP phone to a SIP phone, and not ACROSS technologies, e.g. from a

SIP phone to an IAX phone.

The basic functionality is this:

A call is placed in one or several call groups.

If a phone belongs in a pickup group that matches one of the call's call groups, the

user of that phone may pickup the incoming call by dialing *8#

In v1-0, the feature code seems to be *8# on zap channels, and *8 on SIP channels:

*8# does not seem to work from a SIP phone (unless your phone is configured to

interpret # as the ENTER key)

Groups are numbered in the range 0-63.

You define call and pickupgroup per device, in this way:

callgroup=1

pickupgroup=1-9,13

A common misconception

Callgroups are not intended to call a group of phones. All a callgroup entry does is help

define who is permitted to perform a pick-up for particular calls.

Call pick-up is the function that allows a user to pick up his phone, enter a predefined

number, and take a call from another phone that is currently ringing. It is useful in situations

when somebody has called your colleague and you do not want to jump up from your desk

and run to his desk in order to answer his phone. Instead you pick up your own phone, enter

the predefined number, and take the call on your own phone.

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Balance and Credit

Description

There are two fields that describe the money situation for the user: balance and credit.

Balance is the amount of money you owe the user.

Credit is the amount of money you allow the user to be in debt to you.

The balance can be increased or decreased using Manual Payments.

Credit can be increased or decreased in Users' Details.

These fields act differently and have a slightly different meaning for postpaid and prepaid

users.

Postpaid users

The balance can be any value:

< 0 – the user owes you money for your services. He will get an invoice at the end of

the month, and when he pays that invoice, his balance will increase.

> 0 – the user has overpaid you (you owe the user).

== 0 – nobody owes anything to anyone (this is usual when the user is new or has just

paid his invoice and has not yet made further calls or used other services).

Credit for a postpaid user can be:

> 0 – the user can make calls and use other services until he reaches credit, e.g. if his

credit = 10 USD, he can make calls (and use other services) to a total amount of 10

USD. His balance starts at 0 and goes down to -10 USD. When the balance reaches -10

USD, the user will not be able to make any more calls.

== ∞ - the user can make unlimited calls and use any service without restriction. All

the charges will be included in the invoice.

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Prepaid users

The balance for prepaid users should ALWAYS be more than zero. Otherwise, the prepaid

user will be unable to dial-out.

Credit for prepaid users is ALWAYS zero.

Note: balance and credit are set for users, not devices. For example, if a user has several

devices and his balance/credit does not allow him to make any more calls, he will be unable

to use any of his devices.

Why amounts are without VAT?

VAT stands for 'Value added TAX'. All prices and rates in MOR are without VAT. That's a rule.

each user has VAT %. By that VAT % he gets charged when he makes payments or views his

rate. Info how to configure VAT for specified solutions is available in Manual.

Local Calls

Description

A Local Call is a call made when a device which is connected directly to the system dials the

extension of another device connected directly to the system.

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Billing

MOR does not bill Local Calls.

MOR does not serve local calls when users are using extensions, so no reports are available to

see such calls.

If one user dials another user using DID, then it is possible to see reports about such calls.

The same applies to billing local calls. If one extension dials another using DID, then it is

possible to bill such a call.

Local Calls are billed using DID Rates, not Tariffs.

Calling to Local Device

Use Extension (of another Local Device) to dial another Local Device. Extension can be found

in this Device's Edit window.

Make sure that Device is registered to Asterisk server before dialing.

sip show peers

in Asterisk CLI shows this information.

It is also visible on softphone or VoIP phone screen usually (if this Device is registered to the

Asterisk server).

See also

Sip poke noanswer: Peer 'XXX' is now UNREACHABLE!

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Prepaid Logic

Explanation

Prepaid Logic applies only to Prepaid users.

In /etc/asterisk/mor.conf value, frozen_time describes the maximum time for calls by prepaid

users.

When a user makes a call, his balance is frozen by the amount which is necessary to make the

maximum length of the call (frozen_time).

If the user talks for less time, his part of the balance is unfrozen and the correct amount is

deducted.

If the user talks for a time which is equal to frozen_time, then his call ends automatically after

the frozen_time minutes are up.

Example

/etc/asterisk/mor.conf frozen_time = 3

A call minute to Lithuania MOB costs 0.2 EUR/min.

A prepaid user has 8 EUR in his balance.

When the user makes his first call to Lithuania MOB:

MOR freezes his balance of 6 EUR (30 min * 0.2 EUR/min).

Now the user has 8 EUR in his balance with 6 EUR frozen (2 EUR left free).

The call limit for this call is 30 minutes.

When the user makes a second call to Lithuania MOB (the first call is in progress):

MOR freezes the remaining 2 EUR from the user's unfrozen balance.

Now the user has 8 EUR frozen balance and 0 left as unfrozen.

The call limit for this call is 10 min (2 EUR / 0.2 EUR/min).

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When the user makes a third call to Lithuania MOB (the first and second calls are in progress):

The call is dropped because there is no free balance to cover the call.

When the user hangs up his first call, let's say after 12 minutes:

His balance is unfrozen by 6 EUR.

2.4 EUR (12min * 0.2 EUR/min) is deducted from the user's balance, so now the user

will have a total balance of 5.6 EUR (8 - 2.4), where 2 EUR is still frozen by the second

call.

Now the user can make a fourth call while the second is still in progress, and the

maximum length of the fourth call would be (5.6 - 2) / 0.2 = 18 minutes.

If the user hangs up his second call, say with a length of 9 minutes and there is no fourth call,

we have this situation:

The balance is unfrozen by 2 EUR.

The call price is 9 * 0.2 = 1.8 EUR, so the user's balance will be 5.6 - 1.8 = 3.8 EUR, and

none of this amount is frozen.

If the user talks for 10 minutes in the second call, this call is automatically ended by the

system.

This logic prevents any loss to the system owner. It applies stricter rules, but this pays off in

the long run.

Special case

When only one simultaneous call is allowed for the device and/or user (Call Limit = 1), then

frozen_balance is not used.

Example #1: If User/Device is limited to only one call at a time, he can use all his remaining

balance for the call (if the user is prepaid).

Example #2: Call will last till balance runs out.

NOTE: Two hours is the hardcoded call limit for all calls. Even if the balance allows longer

calls than two hours, the call will still be limited to two hours.

If the user has Call Limit not 1, but his one device (D1) has Call Limit == 1 and another device

(D2) Call Limit not 1, then device D1 will be able to make 1 simm. call two hours in duration,

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and ONLY if the balance is sufficient to cover for frozen_time. Device D2 calls will be limited

to frozen_time.

Calling Card calls are always limited to one call - that is, a calling card can make only one call

(for whole available balance) at a time.

Questions

Q. What would happen if a user has a balance lower than the cost of a "maximum duration"

call?

A. The user will be allowed to talk only for the time that is covered by his balance. Example:

maximum duration = 2 hours. If a user has enough balance for one hour, his call will be

limited to one hour. After one hour, his call will end (the system will terminate it) and you will

not take any loss.

See also

Hangupcause Codes

MOR authorize: Balance too low for more simultaneous calls!

Authentication

There are two types of authentication in MOR: IP authentication, and using login and

password.

IP authentication means accepting calls from a given IP automatically, without requesting any

additional login data.

To use login and password authentication, the sender needs to register his phone or pbx to

MOR before sending calls, otherwise MOR will reject them.

Note: IP and login/password authentication cannot work together for the same device

or provider.

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IP Authentication

This example shows how to configure device to use IP Authentication. Important parts are

marked in RED:

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Fax over VoIP

Fax over VoIP is not reliable and does not work most of the time.

It is possible to get a nearly 100% rate in rare circumstances

but there is a very very low chance of this.

The broad explanation lies in a technology mismatch

between FAX and VoIP, as explained in Steve Underwood's

article here [http://www.soft-switch.org/foip.html].

Steve Underwood developed the SpanDSP package, which

helped to implement FAX support into Asterisk, Freeswitch and other open-source VoIP

platforms.

The T38 protocol is supposed to work reliable over VoIP, but the problems with this protocol

are following:

1. Different hardware vendors do not follow standards and T38 implementation on

various hardware is mostly broken.

2. Asterisk does not support T38 very well.

See also

Send Receive Fax over T38

Fax is not received

I have a problem with Fax

Configuration

mor.conf

This file is in /etc/asterisk/ and describes some values for the app_mor.so application, which

is the main MOR core working in the Asterisk environment.

hostname, dbname, user, password, port, sock – All these values describe

connections to the database.

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min_frozen – Minimum credit left to allow simultaneous calls. If the user's balance is

lower, he will not be able to make simultaneous calls (default 10); Not used starting

from MOR 0.7.

frozen_time – How much balance to be frozen in terms of minutes if the user is

prepaid (default 30). If the user makes several simultaneous calls and is prepaid, we

need to "freeze― some amount to be safe. A more detailed explanation is in Prepaid

Logic.

show_notice, show_warning, show_error – Sets debug level for the app_mor.so

application in Asterisk CLI.

zap_change – Change ZAP into something else in Asterisk to permit the use of some

custom-made channels based on Zaptel.

server_id – Unique number for server identification.

active_calls – Tells app_mor.so to update info to DB related to calls currently on the

system.

dial_out_settings – Global settings for Dial() command in Asterisk (rTt or similar).

serial – Serial key for activating app_mor.so.

use_lcr_partials - Should MOR be able to use LCR by destinations? Default 1.

global_call_timeout - Global timeout for calls in seconds, default 7200 seconds (2h).

check_banned_clis - Should MOR check for banned CLIs? Default 1, Enables/Disables

ANI/CLI ban system (available from MOR 8).

use_flat_rates - Should MOR use Flat-Rates, default 1.

use_speed_dials - Should Speed Dials be enabled.

server_ip - external server IP, default 127.0.0.1 (Used in P-Asserted-Identity).

play_hgc_audio - Should we play audio with error message when call fails? (default 0

if not set).

o If it is equal to 1, then sound files will be played for error codes 200 and higher

o If it is equal to 2, then sound files will be played for ALL error codes (sound files

for these error codes are not included into MOR) (This option is available

starting from MOR 10.)

In order to activate changes made to this file, you need to restart Asterisk or reload

app_mor.so

Reload app_mor.so

1. Login to Asterisk CLI

2. Make sure there are no active calls using command: show channels. Wait till they end

or kill them using soft hangup command.

3. Issue: module unload app_mor.so

4. Then: module load app_mor.so

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See also

How to restart Asterisk server

Configuration from GUI

Where I can find this settings menu?

You can find this menu in SETTINGS –> Setup –> Settings

Global

SETTINGS –> Setup –> Settings –> Global

Company – name of the company.

Company email – email address of the company.

Version – version number shown on the bottom left corner of GUI.

Copyright Title – shown on the bottom left corner of GUI after Version.

Browser Title – browser title for MOR GUI.

FrontPage Text – some text to place on front page.

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Registration

SETTINGS –> Setup –> Settings –> Registration

Enabled? – do we want to let users register online automatically?

Hide registration link in Login page? - tick this option you don't want that users

would be able to register online.

Default country

Asterisk server IP – what IP user should connect his softphone/server?

Send email to user after registration – just yes or no.

Send email to admin after registration – just yes or no.

ReCAPTCHA configuration

Terms and conditions

Allow to create user and device with matching username and password – just yes

or no.

See also

Default user

Invoices

SETTINGS –> Setup –> Settings –> Invoices

Invoice configuration is divided into Prepaid and Postpaid.

Settings

Number start – how the invoice number should start.

Number length – the length in digits AFTER Number Start – e.g. Number Start is not

counted into this length. This value is active only for first number type.

Number type – select more appropriate number format.

Invoice period start – when invoices are issued, default is first day of the month.

Show calls in detailed invoice – yes or no.

Show balance line - should we show balance line in invoice or not?

Round finals to 2 decimals - round or not total in Invoices to 2 decimals.

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Use short file name

Instead billsec show user_billsec

Invoice Page Limit - invoice will be generated till this page limit. Additional page is

added with an explanation that page limit was reached.

Details

Address format – changes field positioning in the invoice for the address.

Address line 1-4 – address info.

Bank details line 1-5 – bank details info.

Invoice balance line - line for balance (e.g. "Your current balance is")

Invoice to pay Line

End title – a title for the end of your invoice.

Additional

Show additional details on separate page - enables additional custom page at the

end of the invoice.

Additional details - the information to be displayed in a custom page at the end of

the invoice.

WEB Callback

SETTINGS –> Setup –> Settings –> WEB Callback

Active? – enabled or disabled.

MaxRetries – how many times to try to redial if the call fails.

RetryTime – in what intervals (in seconds) to try to redial.

WaitTime – timeout value for the call, or how long to wait till the call is answered.

WEB Callback CID - what CID to show on outgoing call.

Web Callback Server - which server from the servers list to use for webcallback.

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Emails

SETTINGS –> Setup –> Settings –> Emails

Sending enabled – if this is checked, you will be able to use Mass Mailing and the

system will send emails upon user registration.

SMTP server – hostname of the server.

Port - port of SMTP server.

Login – username to your SMTP server.

Password – password to your SMTP server.

Batch size – value for Mass Mailing, oe how many emails to send at once.

From - email address to put into From: field in email.

Test email sending - save settings and press this to test email sending. Do not forget

to save settings first!

P.S. Remember to set admin's e-mail in admin's personal settings. Without this

test email sending will not work!

See also

Mass Emails

Email variables

Various

SETTINGS –> Setup –> Settings –> Various

Days for DID to close - how long to keep DID closed before making it free.

Agreement number length - number length in digits.

Change Zap - Sometimes this is used to change "Zap" to something else in GUI, e.g.

to "PSTN", mainly because not everybody knows what Zap stands for. PSTN is more

common.

Device PIN length - length for automatically generated PIN codes for devices.

Device range MIN - from which extension automatic extension assigning starts.

Device range MAX - till which extension automatic extension assigning goes.

o For example, from this interval: [Device range MIN, Device range MAX] MOR will

select extensions for newly created Devices.

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User wholesale tariff enabled - enables user wholesale tariffs.

Allow devices to check balance over HTTP

Allow devices to check rate over HTTP

Play IVR for unauthenticated call

Banned CLIs default IVR - default IVR to route banned numbers.

Do not allow duplicate device user names

Do not allow users to change their email addresses

Allow short passwords in devices

Do not allow prepaid user balance drop below zero

Logout link - a user will be redirected to this link after pressing "Logout"

Change ANSWER to FAILED if HGC not equal to 16 for Users - changes the

Hangupcause for users in call reports from Answered to Failed if Hangupcause is not

16.

Tax

SETTINGS –> Setup –> Settings –> Tax

Settings are explained in Multi-Tax system

Google Maps

SETTINGS -> Setup -> Settings -> Google Maps

Fullscreen - should we show Google Maps in full-screen (for example, remove the

menu from the template)?

Reload Time - how often to refresh in seconds

Width - width for the GMaps window in pixels

Height - height for the GMaps window in pixels

Key - GMaps key.

o Instructions on how to get it here.

o More about Google Maps integration.

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Backups

SETTINGS –> Setup –> Settings –> Backups

Configuration is explained here: Backup system

API

SETTINGS –> Setup –> Settings –> API

Allow API - enable/disable API.

Allow GET API - enable/disable GET in API - use only for testing!

API Secret Key - secret key for API. This will be used as a password for signing queries

to MOR API, check MOR API documentation for more information

XML API Extension - .NET does not follow standards (surprise!); enable this setting if

you are using it to get XML starting with HTTP tags.

Allow API Login Redirect

See also

MOR API

Functionality

SETTINGS –> Setup –> Settings –> Functionality

FAX

Fax device enabled - enable/disable Fax device in MOR.

Email2Fax From Sender - There is a field "From:" in an email form. When MOR sends

Fax to Email, "Email2Fax From Sender" value is puted in "From:" field.

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CSV

CSV Column Separator - how columns are separated in imported/exported CSV files.

CSV Decimal Separator - how decimal parts in numbers are separated in

imported/exported CSV files.

Archive CSV file when size reaches - if CSV file reaches a certain size, the system will

archive it to save time to download it (0 means to archive always)

Active Calls

How many maximum Active Calls to show in GUI - how many Active Calls to show

in GUI

Active Calls refresh interval - Active Calls refresh interval in seconds

Show Active Calls for Users - tick if you want to allow users to see Active Calls

Show Servers in Active Calls

Tariffs/Rates

Show rates for users - tick if you want to allow users to see rates

Show rates without Tax - tick if you want to see rates without TAX

Show Advanced Rates for users - tick if you want to allow users to see Advanced

Rates

Privacy

SETTINGS –> Setup –> Settings –> Privacy

Settings are explained here: Privacy to hide Destination ends

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Visual

SETTINGS –> Setup –> Settings –> Visual

Items per page - the number of items to display per one page

Default date format - System default time format

Number digits - how many digits to display after a comma

Hide Quick Stats - hide Quick stats from start page

Hide device passwords for users - do not display device passwords for users

Hide HELP banner - hide HELP banner in GUI

Hide "I want to" help links

Hide "Manual" link - hides "Manual" link

Show logo on first page - unselect to hide logo on the start page

Show Full Source (CallerID Name and Number)

Hide payment options for postpaid users

Show advanced Device settings

Hide non-completed payments for user - hides non-completed payments (the ones

which are started in MOR, but are not completed in a payment gateway website)

Show only main page - this is a very advanced feature for those who want to hide

default MOR menu tree on the left so that they could build their own menu using

advanced techniques without modifying MOR GUI sources.

Hide Recordings for all users - hides recordings for all system users from menu and

device settings.

Show only main page - this is a very advanced feature for those who want to hide

default MOR menu tree on the left so that they could build their own menu using

advanced techniques without modifying MOR GUI sources.

Hide Recordings for all users - hides recordings for all system users from menu and

device settings.

ReCAPTCHA

A CAPTCHA [http://recaptcha.net/captcha.html] is a program that

can tell whether its user is a human or a computer. You've

probably seen them — colorful images with distorted text at the

bottom of Web registration forms. CAPTCHAs are used by many websites to prevent abuse

from "bots," or automated programs usually written to generate spam. No computer program

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can read distorted text as well as humans can, so bots cannot navigate sites protected by

CAPTCHAs.

More info [http://recaptcha.net/captcha.html]

Implementation in MOR

reCAPTCHA is disabled by default in MOR.

To enable it, go to SETTINGS –> Setup –> Settings –> Registration.

Here mark reCAPTCHA enable and press on Get reCAPTCHA keys

[https://admin.recaptcha.net/accounts/signup/?next=]

and follow pictures how to register and get keys:

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Here enter your domain name, or your IP if you do not have a domain - for example:

http://22.33.44.55 (put your IP here! Do not leave support.kolmisoft.com or 22.33.44.55 <-

these are just examples!)

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Copy-paste these keys into MOR configuration:

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Now, on the registration page, the user should enter valid words to complete registration.

And in the registration page, the user will see captcha:

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Payments configuration

Where can I find Payments configuration menu?

You can find Payments configuration menu by navigating to:

SETTINGS -> Setup -> Settings -> Payments:

Vouchers

Vouchers enabled – yes or no.

Number length – how long a voucher's number should be; make it at least 10 digits

long.

Disable time (minutes) – if a user fails several times to enter the correct voucher

number, the period for which he should be prevented from trying again.

Attempts to enter – how many times a user can try to enter the voucher number.

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PayPal

PayPal Enabled – yes or no.

PayPal Email – account in PayPal where you want to get payments.

Default currency – select a currency which matches your account's supported

currency. If they do not match, you will get errors.

User pays transfer fee – check whether the user should pay a transfer fee; otherwise

you as the system owner will pay it (default).

Default amount – amount to show for customer when he tries to enter payment. User

can change this amount.

Minimal amount – the minimum amount a user can send you.

Test – if this is checked, all amounts will be equal to 0.01. Good for testing with a real

account.

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WebMoney

WebMoney Enabled – yes or no.

Default currency – select a currency which matches your account's supported

currency. If they do not match, you will get errors.

User pays transfer fee – check whether the user should pay a transfer fee; otherwise

you as the system owner will pay it (default).

Default amount – amount to show for customer when he tries to enter payment. User

can change this amount.

Minimal amount – the minimum amount a user can send you.

Test – testing mode.

WebMoney SIM MODE - testing mode in WebMoney system. Connects to the test

server instead of a real one.

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LinkPoint

Enabled – yes or no

Test – testing mode

Allow HTTP requests – yes or no

LinkPoint StoreID – enter LinkPoint StoreID

LinkPoint Currency – select currency

Default Amount - amount to show for customer when he tries to enter payment. User can

change this amount.

Min. Amount - the minimum amount a user can send you.

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Cyberplat

Cyberplat Enabled – yes or no.

Cyberplat Test – testing mode.

Cyberplat Currency – select a currency which matches your account's supported

currency. If they do not match, you will get errors.

Default amount – amount to show for customer when he tries to enter payment. User

can change this amount.

Minimal amount – the minimum amount a user can send you.

Transaction Fee - a transaction fee that user pays for a transaction

Shop IP - a shop IP

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Ouroboros

Enabled – yes or no.

Merchant Code – a merchant code

Secret Key – a secret code.

Default amount – amount to show for customer when he tries to enter payment. User

can change this amount.

Minimal amount – the minimum amount a user can send you.

Currency – select a currency which matches your account's supported currency. If they

do not match, you will get errors.

Language – in which language Ouroboros should display the web page.

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Default device settings

About

These settings will be applied to manually created devices.

This means that if you enter some special settings here and later go on to create a new device

for a user, the device will have these settings preset.

Where to find Default device settings?

Go to SETTINGS -> Setup -> Settings -> Default device:

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Default values of Default Device

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Default user

What is Default User settings?

These are the default settings which are applied to all new users.

Where to find Default User settings?

Default user settings can be found in SETTING -> Setup -> Settings -> Default User

What are the most important Default User settings?

The most important Default User settings are:

LCR - set default LCR for new users

Tariff - set default Tariff for new users

Balance - the amount you enter here user will get to call for free. Warning - if you have

registrations enabled from MOR login page - this setting can lead to significant

money losses if this option is set high.

For more details about available options in default user settings please refer to user details.

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Accountant permissions

Description

By default Accountants have only read-only rights.

It is possible to allow them to make some changes to the system by changing their

permissions.

Accountants belong to some Accountant Groups with different permissions.

This makes it possible to create separate Accountant types, such as:

Financial accountant - which can manage Tariffs/Rates, etc.

Network engineer - can manage Devices.

Network administrator - can manage Users.

... and so on. Many variations are possible.

Accountant Groups

SETTINGS -> Setup -> Settings -> Accountant permissions

Create groups here:

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Create permissions:

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Create Accountants and assign them to an Acc. Group.

Logo change

What is a LOGO?

LOGO (shortcut of "logotype") is a picture that represents your company. For more

information see logotype.

Where to change a LOGO?

Go to SETTINGS -> Setup -> Settings -> Logo

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How to change a LOGO?

You can do this by clicking Choose file, navigating to the file in your operating system and

pressing Upload logo.

See also

Custom Reseller Login page LOGO

Currencies

SETTINGS -> Setup -> Settings -> Currencies

Here is a list of supported currencies.

MOR operates in one and only one currency.

Default Currency

It is necessary to decide in which currency the software will operate and set such currency as

the default before starting to work with the software.

That means that all operations inside MOR will be done in this default currency.

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You can fin a video which demonstrates how to change the default currency here

[http://wiki.kolmisoft.com/index.php/Currencies]

Currencies window

In the Currencies window, we see a list of currencies.

The first one is the default and the most important currency.

Exchange rate is equal to Default Currency DIVIDED BY some other currency, e.g. EUR

Exchange rate when Default Currency is USD is equal (in 2009-03-24) 0.7351, e.g.

USD/EUR.

Last Update is for informational purposes.

Active? - shows if this Currency is used in GUI (to create Tariffs, Payments, view Stats,

etc).

Update? - if checked, this currency (its exchange rate) will be updated from

finances.yahoo.com at midnight.

Yahoo - update this currency (its exchange rate) from finances.yahoo.com right now.

Edit - this option is accessible when Update? is OFF. You can then enter Exchange

Rate manually for this currency.

Update currencies from web - updates all currencies marked as to be updated from

the web.

Add/Remove Currency

There is a main set of currencies that cannot be removed from MOR.

You can add own currency - just make sure it is not in the list already.

To add a currency, fill data at the bottom of the Currencies table and click the Add icon.

For a newly added currency, you will need to set the Exchange Rate manually. Updating from

Yahoo is not possible.

A manually added currency can be deleted, that is, removed from MOR.

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Default supported currencies

These currencies are in MOR by default and can be updated automatically from

finances.yahoo.com.

USD United States dollar

EUR Euro

XAF Cameroon CFA Franc BEAC

ALL Albanian Lek

DZD Algerian Dinar

ARS Argentinian Peso

AWG Aruban Florin

AUD Australian Dollar

ATS Austrian Schilling

BHD Bahraini Dinar

BDT Bangladesh Taka

BBD Barbados Dollar

BEF Belgian Franc

BZD Belize Dollar

BMD Bermuda Dollar

BTN Bhutan Ngultrum

BOB Bolivian Boliviano

BWP Botswana Pula

BRL Brazilian Real

GBP British Pound

BND Brunei Dollar

BGN Bulgarian Lev

KHR Cambodian Riel

CAD Canadian Dollar

CVE Cape Verde Escudo

KYD Cayman Islands Dollar

BCEAO CFA Franc

XOF CFA Franc

BEAC CFA Franc

XPF CFP Franc

CLP Chilean Peso

COP Colombian Peso

KMF Comoros Franc

CRC Costa Rican Colon

HRK Croatian Kuna

CUP Cuban Peso

CYP Cypriot Pound

CZK Czech Koruna

DKK Danish Krone

DJF Djibouti Franc

DOP Dominican Peso

NLG Dutch Guilder

XCD East Caribbean Dollar

EGP Egyptian Pound

SVC El Salvador Colon

EEK Estonian Kroon

ETB Ethiopian Birr

FJD Fiji Dollar

FIM Finnish Markka

FRF French Franc

GMD Gambia Dalasi

DEM German Mark

GHC Ghanaian Cedi

GIP Gibraltar Pound

GRD Greek Drachma

GTQ Guatemala Quetzal

GNF Guinea Franc

GYD Guyana Dollar

HTG Haitian Gourde

HNL Honduras Lempira

HKD Hong Kong Dollar

HUF Hungarian Forint

ISK Iceland Krona

INR Indian Rupee

IDR Indonesian Rupiah

IEP Irish Punt

ILS Israeli Shekel

ITL Italian Lira

JMD Jamaican Dollar

JPY Japanese Yen

JOD Jordanian Dinar

KES Kenyan Shilling

KWD Kuwaiti Dinar

LAK Laos Kip

LVL Latvian Lats

LBP Lebanese Pound

LSL Lesotho Loti

LTL Lithuanian Litas

MGF Malagasy Franc

MWK Malawi Kwacha

MYR Malaysian Ringgit

MVR Maldives Rufiyan

MTL Maltese Pound

MRO Mauritania Ouguiya

MUR Mauritius Rupee

MXN Mexican Peso

MNT Mongolian Tugrik

MAD Moroccan Dirham

MZM Mozambique Metical

MMK Myanmar Kyat

NAD Namibian Dollar

NPR Nepal Rupee

ANG Netherlands Antilles

Guilder

NZD New Zealand Dollar

NIO Nicaraguan Cordoba

NGN Nigerian Naira

NOK Norwegian Krone

OMR Oman Rial

PKR Pakistani Rupee

PGK Papua New Guinea Kina

PEN Peruvian Sol

PHP Philippines Peso

PLN Polish Zloty

PTE Portuguese Escudo

QAR Qatari Rial

CNY Renminbi Yuan

ROL Romanian Leu

RUB Russian Ruble

SBD Solomon Islands Dollar

STD Sao Tome & Principe

Dobra

SAR Saudi Arabian Riyal

SCR Seychelles Rupee

SLL Sierra Leone Leone

SGD Singapore Dollar

SKK Slovak Koruna

SIT Slovenian Tolar

ZAR South African Rand

KRW South Korean Won

ESP Spanish Peseta

LKR Sri Lanka Rupee

SHP St. Helena Pound

SDD Sudanese Dinar

SRG Surinam Guilder

SZL Swaziland Lilangeni

SEK Swedish Krona

CHF Swiss Franc

SYP Syria Pound

TWD Taiwan New Dollar

TZS Tanzanian Shilling

THB Thai Baht

TOP Tonga Isl Paanga

TTD Trinidad Dollar

TND Tunisian Dinar

TRL Turkish Lira

UGX Ugandan Shilling

UAH Ukraine Hryvnia

AED United Arab Emirates

Dirham

VUV Vanuatu Vatu

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VEB Venezuelan Bolivar

VND Vietnam Dong

WST Western Samoa Tala

ZMK Zambia Kwacha

ZWD Zimbabwean Dollar

Translations

If you want to change the translation of the MOR GUI, you can click on the selected flag in

the upper right corner of the window.

In order to manage Translations of the MOR, you should go to SETTINGS –> Setup –>

Settings -> Translations.

Here you will see a window with all the available translations.

In the first column, you can Drag & Drop a translation to its new place. This column is

dedicated to managing the order of translations.

The second column is for activating/deactivating translations. Simply click on the desired

translation to activate/deactivate it. When disabled, it is grayed out.

If you want to see changes instantly, click the icon at the bottom. The page will be reloaded

and you will not need to logout/login to see the changed translation order in the upper right

corner of your window, where translations are represented as flags.

The first translation is the DEFAULT translation. If you want to change the default translation,

Drag&Drop the other translation to the very top of the translations list.

The default translation is the translation that is active when the user comes to the page for

the first time.

You can find a video which shows you how to change the default translation here

[http://wiki.kolmisoft.com/index.php/Translations]

See also

MOR GUI Translation

Change flag for translation

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Global date format

Global date format can be easily changed in SETTINGS -> Setup -> Settings -> Visual.

To change the date format select it from Default Date Format drop-down menu.

P.S. Don't forget to save settings by pressing the Save changes button

See also

GUI configuration

More info about date formats on Wikipedia

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Remove Manual link from admin GUI

What is a Manual link in MOR?

Manual link opens a MOR MANUAL in Kolmisoft wiki page. This link can be found on the

right bottom:

How to disable a Manual link ir MOR?

1. Go to SETTINGS -> SETUP -> Settings menu:

2. Select Visual tab and select an option Hide "Manual" link.

3. Press Save changes button

The MANUAL link should not be visible now.

Voicemail

Description

Voicemail (or voice mail, voice-mail, vmail or VMS, sometimes called messagebank) is a

centralized system of managing telephone messages for a large group of people. The term is

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also used more broadly, to denote any system of conveying voice messages, including the

answering machine. [http://en.wikipedia.org/wiki/Voicemail]

Voicemail is automatically sent to a user's email if the email address is specified in the device

details.

An audio recording in WAV format is attached to the email.

One minute of audio takes approx. 100 Kb.

Configuration

This section can be found in SETTINGS –> Billing –> DIDs –> PBX Functions –> VoiceMail:

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VoiceMail Extension (local) - describes which local extensions a local user can dial to

retrieve his voicemail (the system will ask him to enter his voicemail password, which

can be set in the device's details).

If you want to setup VoiceMail (VM) on a different server, then the following settings are

important:

VoiceMail Server active - turns VM server on/off. When on, all VM requests will go to

distant VM server; when off, VM will be handled locally on Asterisk.

VoiceMail Server connection/device - describes connection to VM server.

VoiceMail Retrieve Extension (local) - which extension to dial on a distant VM server

to retrieve voicemail for an extension.

Example:

Let's say we have our server on IP 23.45.67.89 connected with SIP to our Asterisk.

We create a separate user/device (vm_device) to describe this connection. Device's

type is SIP, IP = 23.45.67.89; set other settings also (codecs/etc).

VoiceMail Retrieve Extension (local) = 999.

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First action is when VM is recorded

So now when the VM server is ON and the user with device 201 gets a VM, the call is

redirected like this:

Dial(SIP/vm_device/201)

This means that the server on 23.45.67.89 will receive a call to extension 201. The server

SHOULD BE configured so that it will process VM correctly when it receives such a call.

Second action - retrieving VM

When user wants to retrieve his VM on a distant machine he dials VoiceMail Extension

(local).

He is redirected like this:

Dial(SIP/vm_device/999#201)

Again, the distant server should know how to process such a request.

Let's say we have a distant VM server based on Asterisk.

So in our example its configuration in /etc/asterisk/extensions.conf should look like this:

[incoming]

# to retrieve VM

exten => _999#.,1,VoiceMailMain(${EXTEN:4})

exten => _999#.,2,Hangup

# to record VM

exten => _X.,1,VoiceMail(${EXTEN})

exten => _X.,2,Hangup

Retrieve VoiceMail from outside over DID

In PBX functions create new function with type: check_voicemail.

Assign this PBX Function to a DID.

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FROM string

To change the Voicemail email sending FROM field, edit the /etc/asterisk/voicemail.conf file

line:

;fromstring=The Asterisk PBX

to:

fromstring=SOME OTHER LINE

... and reload Asterisk.

serveremail

This setting can be used to identify the source of a voicemail notification message.

;serveremail=asterisk

to:

serveremail=SOME OTHER LINE

OR:

;[email protected]

to

serveremail=YOUR COMPANY@YOUR COMPANY.net

... and reload Asterisk.

Leave only one of those settings, not both.

Change UNAVAILABLE message

Dial Voicemail access number (default *97) and press 01; then, after the signal, dictate your

UNAVAILABLE message.

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MWI - Message Waiting Indication

In telephony, a Message Waiting Indicator (MWI) is a telephone calling feature that

illuminates a LED on selected telephones to notify a user of waiting voicemail messages. It

works on most telephone networks and PBXs. (Wikipedia)

MWI is supported automatically starting from MOR 9.

Most phones support this feature.

See also

No sound on Voicemail or Playback

Voicemail does not work

Voicemail is sent with wrong time

Email settings

Please check Configuration from GUI for information how to setup email sending.

Emails

What are Emails?

This function is used to send emails to your clients. It can be used to send commercials, notes

about service unavailability or anything else – maybe Christmas greetings!

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Where I can find Emails?

This function can be found in SETTINGS –> Setup –> Emails:

Here you see emails which are templates for emails about various system messages. The 'test'

email is created for demo purposes for mass-mailing.

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Next is how to Edit email. Creating email and editing it are very similar processes. Here is

how to edit:

The Name and Subject fields are simple. The most important field is Message text. Here you

enter message for the user. Also, you can use the following variables here:

<%= server_ip %> - Server IP where your users should register their SIP/IAX2 devices.

<%= device_type %> - Used on user registration - shows the device type the user has

selected.

<%= device_username %> - Used on user registration - shows the device's username.

<%= device_password %> - Used on user registration - shows device's password.

<%= login_url %> - GUI login url.

<%= login_username %> - User's username to login to GUI.

<%= login_password %> - User's password to login to GUI.

<%= username %> - same as login_username.

<%= first_name %> - First name of the user.

<%= last_name %> - Last name of the user.

<%= full_name %> - First name and last name of the user.

Click on Change to save any of these variables.

Full list of MOR supported email variables can be found in Email variables page.

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In the main window (SETTINGS –> Setup –> Emails), you can press on icon to select to

which recipients to send this email:

Additional Notes

Emails sending must be enabled to use this functionality

Full list of MOR supported email variables can be found Email variables page.

Email variables

Every email

<%= server_ip %> - server IP where your users should register their SIP/IAX2 devices.

<%= device_type %> - shows the user's primary device type. (The device is set as the

primary device after user registration).

<%= device_username %> - shows the user's primary device username. (The device

is set as the primary device after user registration).

<%= device_password %> - shows the user's primary device password. (The device is

set as the primary device after user registration).

<%= login_url %> - GUI login url.

<%= login_username %> - user's username to login to GUI.

<%= username %> - same as login_username.

<%= first_name %> - first name of the user.

<%= last_name %> - last name of the user.

<%= full_name %> - first name and last name of the user.

<%= balance %> - user's balance.

<%= nice_balance %> - formated user's balance.

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<%= warning_email_balance %> - amount below which user should be warned with

email.

<%= nice_warning_email_balance %> - formated amount below which user should

be warned with email.

<%= currency %> - default system currency's short name (USD).

<%= user_email %> - user's email address.

<%= company_email %> - company email address.

<%= email %> - company email address.

<%= company %> - company title.

<%= primary_device_pin %> - PIN of user's default device. The fault device is created

the moment the user registers, so this variable can be used universally.

After registration

<%= login_password %> - user's password to login to GUI.

<%= user_ip %> - user's IP address.

Cyberplat payment report

<%= currency %> - payment currency.

<%= amount %> - payment amount.

<%= date %> - payment date.

<%= auth_code %> - authorization code.

<%= transaction_id %> - payment transaction ID provided by Cyberplat.

<%= customer_name %> - payment customer name provided by Cyberplat.

<%= description %> - payment details provided by Cyberplat.

Calling Card purchase

<%= cc_purchase_details %> - cards list (Number, Pin, Price).

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Example

This template:

Hello, <%=full_name%>

Your balance in MOR has fallen below <%= nice_warning_email_balance %> <%=

currency %> and now is <%= nice_balance %> <%= currency %>.

Please login to <%= login_url %> and replenish it.

Yours sincerely,

KOLMISOFT

will result in an email similar to this:

Hello, John Smith

Your balance in MOR has fallen below 10.00 USD and now is 5.67 USD.

Please login to http://your_site/billing and replenish it.

Yours sincerely,

KOLMISOFT

Illegal usage

You will get an error when you try to use non-existent variables.

Also, you cannot use something like this: <%= "www.kolmisoft.com" %>

See also

Email Configuration

Mass Emails

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Number Manipulation

Localization

Description

Localization transforms received number to E.164 format number.

The main idea of localization is: No matter how the user dials the number (destination),

when localized it should be in E.164 format.

Localization has nothing in common with numbers you send to providers. It operates with

numbers received from the caller.

Depends on the location of the user (or his device), it is often important to allow users to dial

differently formatted numbers to reach a destination. For this purpose, each device has a field

Location, which describes the physical location of the device and provides rules for how th3

device can dial a destination. In other words each device IS in (or belongs to) some Location

where specific dialing rules apply.

Location describes where a caller IS - not where he is calling to!

By default, a device has the location Global. Global rules apply to ALL devices. If a user

belongs to some other location and his dialed destination has some rule for his destination,

then global rules are not activated – they have a lower priority in location rules.

You can access Localization settings in SETTINGS –> Billing –> Functions –> Localization.

In the main window you can see available Locations:

At first there is only the Global location. You can't delete this location. It's the default for all

devices.

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You can add a new Location by entering this Location's name in the field at the bottom of the

table:

and clicking :

In the Rules column, click the icon to access Location's rules:

The following settings should be explained for rules:

Name – just for informational purposes.

Cut – what prefix should be cut from the number.

Add – what prefix should be added to the number.

Min Len – the minimum length of number to which the rule is being applied.

Max Len – the maximum length of number to which the rule is being applied.

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Tariff - which Tariff to apply when this rule is applied. It is used in special cases, for

example here. Leave this empty in most cases.

LCR - which LCR to apply when this rule is applied. It is used in special cases, example

here. Leave this empty in most cases.

When you have entered all the necessary values (Cut or Add can be left empty), click .

For example:

The normal Lithuania MOB prefix is: 3706.

The user's device is located in Vilnius, Lithuania (displayed as "Lithuania Vilnius").

By current Lithuanian regulations, the local customer can dial 9-digit numbers starting

with 86 to reach Mobile subscribers.

To describe this situation, we create the following rule:

Rules can be enabled or disabled by clicking the icon or in the Enabled? column.

- This icon allows you to edit the details of the rule.

- This icon allows you to delete the rule.

Now, in the main Location window:

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click on in the Devices column to access Devices in Global Location:

Here all Devices with a selected location are listed. Choose other locations for the selected

Device and click on icon to activate this change. The device will be assigned to the selected

Location.

NOTE: It's possible to change a device's Location in the Device settings (SETTINGS –> Users

–> Devices).

Now our device belongs to Location "Lithuania Vilnius". And if we dial a mobile number in the

Lithuanian format, for example 863042438, MOR cuts 86 (leaving 3042438) and adds 3706.

The result is: 37063042438, which is E.164 compatible and can be billed correctly by MOR.

NOTE: make sure that after localization your number is always E.164 compatible. Call Tracing

can help you do this.

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Special cases

Cut everything till Hash

With rule Cut: *# it is possible to cut everything till # (# included) in the dialed destination:

For example:

Dialed number: 1aaaaaaaaa#37063042439

After Localization: 37063042439

This function can be used in fancy MNP (Mobile Number Portability) schemes and perhaps

elsewhere.

NOTE: The Add value still can be used to add any value after Cut is used on the Destination.

Reseller localizations

Starting from MOR 10 reseller is not using admin's global localization rules. Instead now he

can copy the rules and modify them according to his needs. All new created resellers have

those global rules copied automatically from admin.

IMPORTANT: Fields "Change User Tariff to this Tariff" and "Change User LCR to this LCR" will

not be copied.

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Default location does not act as Global location on reseller's account. Only the rules set in the

device's settings will be applied.

Route to DID

Admin and Reseller PRO users are can route calls to a DID according to localization rules.

Change owner of call depending on incoming prefix

It is now possible to change the owner of the call depending on the incoming tech prefix.

1. Open the localization menu to find the "Change Call's Owner" field.

2. Enter the incoming prefix that needs to be cut in the "Cut" section.

3. Afterwards, choose a user from the drop down list and the device to which the call

should be assigned:

This feature is available also for resellers.

Admin can assign the call to ANY user on the system (even to those that do not

belong to admin).

See also

Number Manipulation

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o Provider Rules

o DID handling by Localization

o Number Manipulation Examples

Provider Rules

Provider Rules transforms an E.164-compatible number to any number format the

Provider requires.

Description

Different providers expect different number formats from you. You can send them whatever

they want using Provider Rules.

Provider Rules can be reached by clicking the icon next to a provider in the Providers

window:

We see a similar window as the one in Localization. The rules to manage Provider Rules are

identical to those for Localization management, which are covered in number manipulation

page.

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Let's say our provider expects to get a number with the technical prefix 89765#. We can do

this in following way:

That's it – very simple. Now every number that is sent to this provider will be transformed by

adding our created tech. prefix.

For example, if 37063042438 should be sent to this provider, this rule will be activated and

the number 89765#37063042438 will be sent to provider.

REMINDER: You can check whether Provider Rules are applied correctly in Call Tracing

Special cases

Cut everything till Hash

With rule Cut: *# it is possible to cut everything till # (# included) in the dialed destination:

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For example:

Dialed number: 999#37063042439.

After Localization: 37063042439.

This function can be used in fancy MNP (Mobile Number Portability) schemes and perhaps

elsewhere.

NOTE: Add value still can be used to add any value after Cut is used on the Destination.

CallerID overwrite for Provider

Sometimes Providers only accept calls with CallerID.

It is possible to overwrite any CallerID before sending a call to such a Provider.

In Provider Rules (for this Provider) set up the Rule:

In this example, all calls going to this Provider will have CallerID = 9999999999

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Destination same as received

If you want to send a number to a Provider exactly as it was received, set Cut = '-' (minus sign

as shown in the screenshot).

That way, the same number that comes to MOR also leaves MOR:

See also

Number Manipulation

o Localization

o DID handling by Localization

o Number Manipulation Examples

DID handling by Localization

When a call comes in to your system from the Provider, it should be localized to E.164 format

for DID search. In other words, as DIDs are in E.164 format, whatever comes from the provider

should be localized to E.164 to find the correct DID.

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To do this, each Provider also has a Location. This means that Location rules can be created

and applied for the Provider. You can do this in the Provider settings window:

For example, you have DID 37052058393 and the provider sends you 0037052058393. You

need to strip this international prefix with the following rule:

Don't forget to apply the correct Location to the Provider!

Special case: In some countries when E1 is connected to the server, only the four last digits of

the number reach MOR. Localization should be used here to add the remaining part of the

number.

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See also

Number Manipulation

o Localization

o Provider Rules

o Number Manipulation Examples

Number Manipulation Examples

Example with Localization and Provider Rules for CallerID(Source) and Destination

Lets say 868555666 is calling to 868777888. The first number is the source (CallerID), and the

other number is the destination. Localization source rules are used to modify the CallerID

which comes to MOR, and provider source rules are used to modify the CallerID that will be

sent to the provider. They aren't used all the time, only when owner of system wants them to

be.

Take, for example, a provider that only accepts calls that are nine numbers in length and that

begin with 8. MOR works only with E.164 standard callerID's if you want to run billing on

them.

First of all, we need to make a Localization source rule for our situation:

Cut 8, add 370, Min/Max 9/9.

So now the caller's number is 3708555666, not 868555666. MOR identifies the caller as being

from Lithuania (by country code 370), which means he can bill him.

Now we need to send a call to 868777888, but our number must be in the old format

(868555666), so we need to apply a suitable provider source rule:

Cut 370, add 8, Min/Max 11/11.

That's it - the caller's final number will once again be 868555666.

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Examples

Example 1 - Italy

The client has the following situation:

My customers are mainly in Italy. To call Italy, they are not going to add the 0039 country

code, while they do put 00 and the country code when they want to call other countries.

Then, on the other side, my provider aspects numbers to Italy without the 0039 country code,

while to other countries it does aspect the country code before the called number.

NOTES:

The numbers that clients dial are 10 digits in length. For more info about the Italian

number format, see: http://en.wikipedia.org/wiki/%2B39

'+' is not used. It should be changed to '00'.

To handle such a situation, the following actions should be taken:

Location 'Italy' should be created.

All devices from Italy should be assigned to this Location.

Rules in Location Italy:

o Name: 'Local Italy', Cut: (empty), Add: 39, Min: 6, Max: 10.

o Name: 'International', Cut: 00, Add: (empty), Min: 11, Max: 16.

Provider Rules for Provider through which call will be terminated:

o Name: 'International', Cut: (empty), Add: 00, Min: 10, Max: 100.

Example 2 - USA/ Canada

The client has the following situation:

A local number is dialed in following formats: 1-XXX-XXX-XXXX or XXX-XXX-XXXX.

International numbers: 011 - country code - area code - number.

Localization rules:

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Now the call should be sent to the provider which accepts numbers in following format:

USA/Canada calls should start with 1.

International calls should start with 011.

Provider Rules for this Provider should look like:

Example 3 - Brasil

Local number dialed in format 0XXXXXXXXXX.

International: 00 + country code + area code + number.

Example 4 - Spain

Local number dialed in format 9XXXXXXXXXX (includes area code + number).

International: country code + area code + number.

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Block wrong short numbers with Localization

To block ALL short numbers (length [1-8]) you need to:

1. Create Empty LCR.

2. In Localization, add the following rule:

NOTE: Such a rule will route calls to Empty LCR, where they will be dropped.

If you want to allow some short numbers to pass, you need to create more granular rules to

handle your situation.

This example is very strict and is only for demo purposes to illustrate the principle.

Make sure you understand Number Manipulation to proceed with rules for your case.

See also

Number Manipulation

o Localization

o Provider Rules

o DID handling by Localization

o Number Manipulation Examples

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LCR/Tariff change based on call prefix

The main purpose of this function is for calls to be routed through a provider or providers

based on call prefix.

You have to create a provider, tariff, LCR (where the specified provider has to be), and a user

with his device. And most important, the localization rule or rules. It should be applied per

device basis (Device settings -> Location), making sure the specified device has the correct

rule applied. It must be noted that specified rules have higher priority than global rules.

1) Create a new tariff, provider, LCR, user and device.

2) Assign the newly made provider to the new LCR.

3) Create a new localization rule, named for example "test1".

4) Assign the device to location "test1".

5) Now edit the "test1" rule. It should be done last with normal localization rules, but

when you are creating a localization rule based on tariff or LCR, assign the specified

Tariff or LCR to correct place.

You can find more info on how to make rules in Number Manipulation.

Example

Let’s say we have a provider that provides three quality routes: bad-cheap/normal/good-

expensive.

The bad-cheap route is selected when dialing an E.164 number with prefix 00.

The normal route is dialed when dialing the normal E.164 number.

The good-expensive route is selected when dialing the E.164 number with the prefix

000.

To accomplish this, we need to:

Create three providers with the same connection settings:

o Bad_Provider

o Normal_Provider

o Good_Provider

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Create three Tariffs for providers:

o Provider_Tariff_Bad

o Provider_Tariff_Normal

o Provider_Tariff_Good

Enter/Import appropriate rates to these Tariffs.

Assign these Tariffs to providers.

Create three LCRs and assign appropriate providers:

o Bad_LCR - assign Bad_Provider here

o Normal_LCR - assign Normal_Provider here

o Good_LCR - assign Good_Provider here

Create three Tariffs for users:

o User_Tariff_Bad

o User_Tariff_Normal

o User_Tariff_Good

Enter/Import appropriate rates to these Tariffs.

Create Location BadNormalGood with the following Rules:

Assign User_Normal_Tariff and Normal_LCR to User.

Assign all the user's devices to Location BadNormalGood.

That's it.

Now when the user dials E.164 numbers, he will be using the normal route; when 00, the bad

route; and when 000, the good route.

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Users

User Details

Where to find user details?

1. Navigate to users list

SETTINGS –> Users

2. Select the marked icon

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3. You will get into the following menu:

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In this window you will see a lot of user settings.

There are several groups: General, Blocking, Warning Balance Email, Invoices, Payments,

Details, Taxes, Registration Address and Recordings.

General

Username – username for user to login to MOR GUI.

Password – password for user to login to MOR GUI.

User type – by default it is "user".

LCR – LCR indicating which of the user's devices will be used to dial out.

Tariff – prices for the user.

Balance – the user's balance.

Credit – the user's credit, showing how low the user's balance can go.

Minimal Charge for Calls - minimum charge user pays for voice calls.

Minimal Charge for Calls Date - date since when the previous option is valid.

Postpaid/Prepaid – type of user.

Not Hidden/Hidden – is this user hidden?

Primary device – the primary device for this user.

Allow loss calls - should loss calls be allowed for this user?

Call Limit - how many simultaneous calls the user can make?

Warning balance

Description

This function allows a number of actions to be taken when a user's balance drops lower than

the set value.

You can edit Warning Balance in User Details.

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Check Active to activate this function.

Upon activation, the user will receive an email when his balance drops lower than the balance

set in the field after: Send email when balance drops lower than.

Select when to send such an email:

Only once when balance drops lower than.

Every day at: (select at which hour to send email to user).

Additionally, you can play an audio message whenever the user makes a call.

Check Play before every call and select the audio file to play.

Use this with caution - not many users will like this!

User Blocking

Definition

When a User is blocked, he is unable to make outgoing calls.

NOTE: the User is still able to receive calls. Unassign DID for him to keep him from receiving

incoming calls.

Direct blocking

Mark User Blocked or Not Blocked to instantly block/unblock him.

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Unconditional blocking

This option sets the date when user should be blocked.

If date is in the past, this option is not active.

Block check is done every night at 00:00:00.

NOTE #1: if you want to block the user in the future, set this date in Block at BUT do not mark

this user as Blocked (if you do so, the block will take effect instantly). The system will mark the

user as Blocked at your selected date.

NOTE #2: If you want to block the user today, just mark the user as Blocked. DO NOT enter

today's date into Block at field - the system will not block it if you do so, because it will not

check until tomorrow 00:00:00.

Conditional blocking

Under this option, select day of the month when to perform check.

The check is done at night at 00:00:00.

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On the day the user is checked, the following questions are asked:

Is he POSTPAID ?

Is his BALANCE < 0 ?

Has he UNPAID INVOICES ?

If all conditions are met, this user is blocked.

See also

Monitoring Addon

Blocking

Various options for User Blocking.

Warning Balance Email

If you want to send an email notification to a user (or take another action) when his balance

drops lower than a certain amount, check this setting and enter the desired amount.

More details: Warning balance

Invoices

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Generate Invoice - should we generate an invoice for this user when we choose to

generate invoices for a period?

Invoice, Detailed, By CallerID, By Destinations and PDF/CSV show various types of

invoices which should be sent to the user as an email attachment.

Show zero Calls - if the call price = 0, should such calls appear in the invoice or not?

Payments

Cyberplat - check this option if you want to allow the user to use this payment

gateway.

Details

Self-explanatory.

Taxes

Here tax parameters are entered for the user.

The tax system is explained in Multi-tax system page.

Registration Address

Self-explanatory.

Recordings

This section is available when the Recordings Addon is installed in the system.

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Hide Users

Why should I hide users?

As it is not allowed to delete users who have at least 1 call, after some time such 'non-active'

users accumulate and we need to do something about them to have a clear control panel.

For this reason, we just Hide them. They remain active - nothing changes for them - except

that we do not see them in the GUI.

How to hide a user?

1. Go to users list:

Go to SETTINGS –> Users

2. Click the marked icon:

3. Click OK when the confirmation appears

Done, user should be hidden now.

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Additional Notes

The same icon can be found in user's details. It works the same way.

Also you can hide user by selecting an option Hidden in user's details in General

section

Where to find hidden users?

Go to SETTINGS –> Users –> Hidden

How to unhide a user?

Navigate to Hidden Users list and press the same icon again as in a hide user example

See also

Users

o Create a User and a Device

o Why MOR does not allow to delete users?

o User Details

Block User

o (SECURITY) How to change Admin username and password

Providers

o Hide a Provider

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Why MOR does not allow to delete users

MOR does not allow users to be deleted who have made at least one call (or has at least

1 payment) in the past.

WHY?

The answer is simple - because it would ruin the billing.

P.S. Because of this reason you are also not allowed to delete providers and devices.

Simple example

Let's say we have 2 (TWO) users U1 and U2 and 1 (ONE) provider P1.

During the month, one user U1 dialed 20,000 calls and the other user U2 dialed 10,000.

For the sake of simplicity, 1 call = 1 EUR.

So user U1 should pay you 20,000 EUR and user U2 should pay you 10,000 EUR. Total 30,000

EUR.

You sell with 100% margin, so that way you have to pay 15,000 EUR to your provider P1 for

the calls.

And 15,000 EUR is your profit.

Imagine now what happens if you delete user U1 from the system with all his calls.

We have this situation:

During the month user U2 made calls worth 10,000 EUR and you see that you should pay

provider 50% of that = 5,000 EUR to your provider P1.

BUT!!!! Provider P1 sees that he sent 30,000 EUR worth of calls and not 10,000 as you say. And

these calls are worth 15,000 EUR.

So you will be at -10.000 EUR (MINUS 10.000 EUR) after one month!

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What to do?

Hide users if you don't want to see them any more in your system. Note: currently it's not

possible to hide devices.

A description how to hide providers can be found in Hide Provider page

How to create User and Device

Video

This video demonstrates the process WITHOUT setting additional settings. Make sure you

read and understand User Details and Device Settings.

This video can be found here:

http://wiki.kolmisoft.com/index.php/How_to_create_User_and_Device

See also

Users

Devices

How to change Admin details

Video

This video demonstrates how to change the Admin username and password (for greater

security).

This video can be found here:

http://wiki.kolmisoft.com/index.php/How_to_change_Admin_details

See also

Users

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MOR forgot user password

What is Forgot Password functionality

This MOR functionality resets current user password and sends it to his email.

Where to find this functionality?

This functionality can be found at MOR login page:

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How to use this functionality?

Click Forgot Password and the following input box will ask you for an email.

If the entered email matches with the one that already exists in the system - the password will

be sent.

Requirements

"Forgot password" functionality is visible only if:

1. Email settings are entered

2. Email sending is enabled

3. You have MOR 9 or newer

For more information about emails in MOR please read here:

http://wiki.kolmisoft.com/index.php/Configuration_from_GUI#Emails

"Forgot password" functionality can be used only if:

4. User has an email set in his user details

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5. Email must be unique in the system

6. Email settings settings must be correct

Minimum contract billing

Minimum contract billing is the feature that lets you set a minimum monthly charge for

certain clients.

How to set minimum contract billing

To set this functionality go to SETTINGS -> Users. Click on the chosen users name and you

will see this window:

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Select Enabled? as shown in the picture. Now you can set the minimum charge and select

since when the setting will be valid. If you can't see those fields it is possible that your chosen

user is prepaid. This feature is only available for postpaid users.

Specialties

If clients total billing is below the set charge he will have to pay the minimum amount

anyway

If clients total billing is above the set charge he will have to pay the whole amount

Minimum time period per which you can set Minimum contract billing is 1 month

Minimum contract billing starts to be valid only since the first day of the month. It

means if today is February 15 and you set Minimum charge for calls date current years

February, client will start to be billed for it since 1st of March anyway.

Can only be applied for postpaid users

Can only be applied for voice calls

An invoice includes the line with the difference between the minimum amount and the

total amount.

Tariffs

Directions and Destinations

About

Directions refers to countries, or perhaps to a satellite network. The term merely describes

the direction in which the call is routed.

Directions can be found in SETTINGS –> Billing –> Directions. You can see all Directions

with their details in the main window.

Destinations - these are more concrete concepts of the calls course. Let's say we are dialing

number 37068547771. From this number MOR knows that the Direction is Lithuania, but the

Destination is the mobile network Bite. Destinations are determined by Prefix. Each

destination has a unique Prefix. The Prefix from the example is 370685.

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Destinations can have several types/codes:

FIX – fixed numbers

MOB – mobile numbers

NGN – non geographical numbers

IMPORTANT NOTE: When adding new destinations, make sure you include them in some

Destination Group. Otherwise you will get corrupted invoices if the user uses a retail tariff.

QUESTION: How do you import destinations from a file?

ANSWER: You can import them using the Tariff import script. Just create a dummy

tariff, add dummy rates, and import. Destinations will be created automatically. Later

you can delete the tariff. Just remember to enter prefixes in E.164 format.

NOTE: You can get a CSV file with all Destinations in the Tariffs window. It is suitable

for rate entering and importing back to MOR.

Destinations Groups

In order to make life easier and to manage prices for destinations, destinations are grouped.

With grouped destinations, one price applies for a group of destinations, rather than for

destinations separately. (A group can contain one destination also.)

Destinations groups are for users, not for providers, because providers need to have very

exact rates for each destination.

You can edit Destination Groups settings in SETTINGS –> Billing –> Directions –> Groups.

NOTE: A destination can belong to only one group at a time.

HINT: If you want to split a Destination Group, remove all Destinations from it, delete the old

Destination Group, and create new ones. Then add the removed Destinations to the newly

created Destination groups.

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Block some Destination to User

Block Destination to User

If you need to block a special prefix (Destination), follow these steps:

1. Create an empty LCR (without Providers in it) - "Empty LCR".

2. In the LCR used by the user, use Call routing per destination basis to add the prefix

you want to block.

3. Assign "Empty LCR" to this prefix.

Now, when the user dials a number with this prefix (Destination), the call will be routed to the

empty LCR, which is to say it will be dropped.

Block Destination to all Users

If you need to block a special prefix (Destination) to all Users, follow these steps:

1. Create an empty LCR (without Providers in it) - "Empty LCR":

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2. In Localization Global Rule, add a rule for your prefix and set it to route to Empty LCR:

Now, when the user dials a number with this prefix (Destination), the call will be routed to the

empty LCR, which is to say it will be dropped.

(In this example we are blocking UK numbers starting with 44, change in your system

accordingly).

Bulk management

Starting from MOR 10 you can reassign multiple destinations to other direction by specifying

a regexp pattern:

1. Go to SETTINGS –> Billing –> Directions

2. Press Bulk management

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3. Enter a prefix regexp which destinations should be reassigned, select a group to assign to

and specify type:

4. Press Reassign

5. Now MOR shows you which destinations are going to be reassigned:

6. If you want to proceed - press Reassign at the bottom of the window again and the

destinations will be reassigned

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MOR supported regexp patters

Supported regexp symbols:

% - matches any symbol any number of times

[1,2,3,4]% - matches destinations beginning whit 1, 2, 3 and 4

Examples

370% - will match numbers 370xxxxxxxxx

370[1,3,9]% - will match numbers 3701xxxxxxxx, 3703xxxxxxxx and 3709xxxxxxxx

Day setup

MOR lets you set different dates for different types of days. There are two types of days: work

days (WD) and free days (FD).

Saturday and Sunday are counted as Free days by default.

Mondays to Fridays are Work days.

National holidays and other days that do not meet the default settings should be set

manually.

This can be done in SETTINGS –> Billing –> Tariffs –> Day setup:

Here you can add, edit or delete days.

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Entering Rates for Wholesale Tariff

Description

This method is used to enter rates one by one by hand. It is primarily used for editing rates

because it's very time-consuming to enter thousands of rates by hand. CSV import is also

available for this task.

Click on Rates:

In this window you can view already entered rates:

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Click on Details to get more details:

Here you can Split to Work Days and Free Days, delete a rate or Edit it:

Here you can add a different End Time for different rates in a time period and change other

values that are simpler than Advanced Rates.

NOTE: if there are no rates for some destinations, the user will not be able to dial these

destinations using the provider to which the tariff is assigned. This is usual gotcha when call

does not go through – always use Call Tracing if you have problems dialing-out!

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Adding new Rates

It is possible to add Rates using GUI in three different ways. It is also possible to import Rates

using CSV.

Here we show three ways of adding Rates over GUI:

New Rate

This way lets you add many rates for selected countries by first letter - for example, for all

Destinations starting with A:

Enter Rates and click the Update button to add the desired Rates.

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Quickly add a new Rate

This way allows you to quickly add a Rate. Fill in the fields (most important

are Prefix and Price). Make sure the software shows you which Destination it is (if it says that

there is no such Destination, you need to create it) and click Add Rate:

Add new Rate

This method is similar to previous one except that it allows you to enter rates only to a

selected country (Direction). If a country has all rates already set, it will not let you choose

that country:

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User wholesale tariff

This tariff is used by users and is no different from the provider wholesale tariff. All the

functions are the same. This tariff type can be enabled or disabled in Settings.

When a wholesale tariff type is assigned to the user, he will not be able to see all rates online.

He will be able to download them in PDF or CSV format.

NOTE: you can't have Custom rates for a user if he is using the Wholesale tariff.

Rate import from CSV

File format

File should be CSV (comma separated values).

Necessary fields in CSV line:

Prefix

Price

No other fields are necessary.

First line can be header.

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Value Separator and Decimal Separator MUST be set correctly in SETTINGS -> Setup ->

Settings -> Functionality:

CSV Column Separator

CSV Decimal Separator

File and CSV separator settings should match. If they do not, you will get errors when trying

to import file.

Good CSV rate file examples

Bad CSV rate file examples

Where can I get an example file with available prefixes?

Go to SETTINGS -> Billing -> Tariffs and click on Destinations CSV.

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Description

It is very tedious work to add several thousand rates by hand, so you can import them from a

CSV file.

There are seven steps for importing rates from a CSV file:

1. File upload

2. Column assignment

3. Column confirmation

4. Analysis

5. Creating destinations

6. Updating rates

7. Creating new rates

We will cover each of these one by one.

IMPORTANT NOTE: When importing, prefixes should be in E.164 format!!! For example,

no international prefixes!

File upload

Select Import from CSV for the Tariff you want to import rates.

Then click on Browse and select CSV file with rates. Click Upload file:

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Column assignment - wrong separators detected

Sometimes the system is able to detect that CSV separators are wrong and suggests you to

change them:

Column assignment

The file is uploaded and the first five lines of the file are available to check:

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This helps us in the next step, to assign CSV file's columns to MOR's variables. We have the

following window:

Here we must assign columns from the CSV file. Prefix and Rate are mandatory fields. All

other fields can be empty. Connection Fee, Increment and Min. Time have their defaults. If

you wish, you can enter values manually for them. Manually entered values have the

highest priority. If you omit them, then MOR inserts values from the CSV file. If there is

no assigned column for these fields, then default values will be added to database (DB).

There are a group of values under the name USA specifics. These are only for USA/Canada.

Users from these countries need to use these fields with their CSV files.

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In the following example, we select fields which match our CSV file. Also entered are some

manual values for Increment and Min.Time. We count time by 30s increments and minimum

time for the call is 60s, with these settings:

This way of selecting CSV rows and assigning them to MOR values lets us import various CSV

file formats.

Rate Day Type

It is possible to select to which Day Type rates apply before importing them.

So in two steps it is possible to import separate rates for Work Days and Free Days, or (in a

single step) to import rates for All Days at once.

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Destination name update on rate import

When this option is selected destination name is updated from csv file during rate import

Column confirmation

The next step is to review our selection and confirm that everything is correct. We can go

Back or forward to the next step:

Analysis

It will take some time to accomplish this step. MOR checks many things from DB and CSV

files: it compares rates, searches for duplicates, fixes small errors and so on. After the analysis

is completed, we get something like this:

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The most interesting line is 9 bad destination(s). We click on the List to check them. In the

new window, we can see which destinations are bad and we can decide what to do with them.

Is it possible that there are no directions for them, MOR doesn't know to which country to

assign this destination, or maybe these destinations have duplicates in CSV file? You have to

fix this if you want to import the whole file. Here you can stop and fix your CSV file, or you

can ignore these errors and continue.

We continue the demo by clicking Proceed.

The following steps are separated to avoid excessive demands on processing power if a rate

file is huge. This keeps the browser from crashing and allows it to successfully import

160.000+ destinations from one CSV file (this has been tested).

Creating destinations

MOR creates new destinations here. These destinations are necessary for the rates, which will

be added in step 7. This window is just informative, no decisions here:

MOR creates only Destinations for prefixes which are not yet in MOR's DB and determines

whether it is possible to find out the Direction (Country) for these new prefixes (Destinations).

This may be done in two ways:

1. The system checks the available prefixes for the best match and selects found Direction as

a basis for the new prefix (Destination).

For example: if you try to import the rate for prefix 370630 and this prefix is unknown to

MOR, MOR checks the longest matching prefix - 3706 in this case - and finds out that it is

Lithuania MOB. So it creates a new Destination with Prefix 370630 in which the Direction will

be Lithuania and the subcode MOB (mobile network).

2. If Country code is available in the imported file, MOR assigns this Direction to the newly

created Destination.

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For example: if again you try to import 370630 and it is unknown to MOR, but you have LTU

in same CSV line for Country Code field, MOR will create such a Destination with Direction =

Lithuania.

Updating rates

Here MOR updates rates with the values from CSV files. These rates already exist in DB. Just

click Proceed:

Creating new rates

Finally, MOR creates all rates and we are all done!

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NOTES

Rate import is VERY CPU-intensive.

If there are a lot of rates, please be patient and let MOR finish its job. It may take some time

to count everything and come back to life.

At some point it may seem that browser is not responding. That is not the case – it is just

working very hard, importing your very own rates from the CSV file.

If you get a timeout error, try to import the file in smaller pieces.

Destination to create from CSV

When importing rates from the CSV file, you can check which Destinations will be created

automatically.

You can view them in a separate window when you click List in step 4 in the field

"Destinations to create".

Such destinations do not exist in MOR's database but MOR will create them when importing

the rate file.

More info: Integrity Check

Example Videos

Examples how to import tariffs from CSV can be found here:

http://wiki.kolmisoft.com/index.php/Rate_import_from_CSV#Example_Videos

See also

Good CSV rate file examples

Bad CSV rate file examples

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Wholesale Tariff

What is a Wholesale Tariff?

This tariff is a price list which you get from your provider, later you can transform this tariff

into retail or user wholesale tariffs. Without a transformation this tariff cannot be assigned to

user.

Min Time

Min(imal) Time is the minimum time that will be billed if a call is answered.

For example:

Min Time = 10s

If the actual length of a call is <= 10s, billsec = 10.

If the actual length of a call is > 10s, then billsec = actual call length.

Actual call length Call time which will be billed

1 10

2 10

... 10

9 10

10 10

11 11

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Increment

An Increment is the amount of time in which a call is billed. It is counted in seconds.

For example, assume that:

A particular call lasts for 15 seconds.

The rate for 1 min is (for the sake of simplicity) 1 EUR/min.

Then:

If the increment is 1, then the call length will be counted as 15s and the call price will

be 0.25 EUR (1/60*15). As the increment is 1, the call can be 1, 2, 3... in length. 15s is

the exact length of the call.

If the increment is 10, then the call length will be counted as 20s and the call price will

be 0.333 EUR (1/60*20). As the increment is 10, the call can be 10, 20, 30... in length.

20s is nearest to 15.

If the increment is 20, then the call length will be counted as 20s and the call price will

be 0.333 EUR (1/60*20). As the increment is 20, the call can be 20, 40, 60... in length.

20s is nearest to 15.

See also

Tariffs

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Entering Rates for Retail Tariff

In order to manage rates for Retail Tariff, click the Rates next to Tariff:

Here you can enter Rate and Round by values for every rate. If you finished entering what you

want, click Update at the bottom of the page. All rates will be updated.

View shows you the Destinations which belongs to this particular group.

Click on and you will be able to edit rates in Advanced mode.

Here you can click on Split to Work Days and Free Days to split the rate into two parts - for

Work Days and Free Days.

Now we have control over different types of days: Work Days and Free Days. We can enter

different rates for these days.

We can also set different rates for different times in the day. For this, we need to create hour

intervals. To do so, select a time in the column End Time and click on .

In order to manage Rate, click on the Details icon .

And here you should add/edit/delete Advanced rates as described in Advanced Rates.

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Advanced Rates for Users

It is possible to show Advanced Rates for Users.

Make sure they will understand them before showing these Rates to Users.

It is necessary to turn on the settings:

In SETTINGS –> Setup –> Settings –> Functionality check Show rates for users and Show

Advanced Rates for users settings

Click Save changes

Then Users are able to see the rates in their portal:

and their details:

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Advanced Rates

Description

Advanced Rates are used ONLY for Retail Tariffs.

Please see video below where to find and edit advanced rates.

In order to manage rates for destinations more easily, destinations are grouped, so that one

rate applies for a group of destinations, not for one destination separately. (A Group can

contain only one destination if necessary.) Destinations groups are for users, not providers,

because providers need to have very exact rates for every destination.

Each group of destinations consists of several destinations described by prefixes. As an

example, let's take Albania MOB. Here we have three destinations described by prefixes:

35538, 35568 and 35569. Each of these destinations is Albania MOB. Usually users do not care

which MOB network they are calling, and for the less popular countries providers usually put

one price on all destinations in that direction. It's the same here: We can put one price on all

these three destinations grouped as "group Albania MOB".

Each rate for a destination group may have many rate details. This means: a destination

group may have one rate, but this rate has many rate details (sub-rates). Each rate detail

describes one interval of the call. If the first rate detail is RD1, the second is RD2, and so on,

all calls can be expressed as RD1 + RD2 + .. + RDn. The number is finite. The rate detail can

be seen as the time intervals for all call durations. The time interval could also be zero length.

It is explained further below.

Each rate detail consists of the following parts:

FROM – from which second the rate detail starts.

DURATION – the length of this rate detail (interval).

TYPE - minute or event.

ROUND BY – in what intervals to round the time.

RATE – the price per minute in this rate interval.

The type 'minute' describes an interval that lasts some time. This time is billed by rate and

rounded accordingly. An Interval starts at FROM and lasts for DURATION seconds. The type

'event' describes a zero-length interval, which we call EVENT. This might, for example, be a

connection fee. If the type is 'event', the rate detail has no DURATION (it's always zero) or

ROUND BY fields.

NOTE: The same rules apply to Custom Rates for users.

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Examples

Example #1

The call should have 0.2 connection fee (regardless of the currency) and all other time must

be billed by 6-second intervals with a rate of 0.1 per minute. Such a rate should be described

like this:

FROM DURATION TYPE ROUND BY RATE

1 0 event 0.2

1 ∞ minute 6 0.1

Example #2

The first 30 seconds of the call cost 0.1, the next time till 5 minutes is billed at rate 0.05

rounded by 30 seconds, and the remaining time is billed by rate 0.02 and rounded by 1

second.

FROM DURATION TYPE ROUND BY RATE

1 30 minute 30 0.2

31 300 minute 30 0.05

301 ∞ minute 1 0.01

The same example can also be described like this:

FROM DURATION TYPE ROUND BY RATE

1 0 event 0.1

1 30 minute 30 0

31 300 minute 30 0.05

301 ∞ minute 1 0.01

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Example #3

The rate for the call is 0.01 per minute rounded by 1s. The maximum price for call is 0.1.

FROM DURATION TYPE ROUND BY RATE

1 600 minute 1 0.01

601 ∞ minute 1 0

Example #4

The rate for the call is 0.01 per minute rounded by 1s. The first 15 seconds of the call are not

billed.

FROM DURATION TYPE ROUND BY RATE

1 15 minute 1 0

16 ∞ minute 1 0.01

Import from CSV file

It is not possible to import (upload) Advanced rates from the CSV file directly.

Here is a workaround:

1. Create Wholesale Tariff.

2. Import Wholesale rates to the newly created Wholesale Tariff.

3. Transform Wholesale Tariff to Retail Tariff.

4. Delete Wholesale Tariff.

Where to find advanced rates

Video that shows where to find advanced rates can be found here:

http://wiki.kolmisoft.com/index.php/Advanced_Rates#Where_to_find_advanced_rates

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Additional Notes

Allow users to see advanced rates

See also

Tariffs

Destinations Groups

Custom Rates

Custom Rates are rates that differ from a default user's tariff.

For example: users A and B both have the tariff 'GOLD', but customer A has a rate to Russia of

0.5 and customer B has a tariff to Spain of 0.6. 'GOLD' rates for these destinations are 0.7 and

0.8. Put new values for each customer in their Custom Rates and the new rates will be used

instead of the default tariff rates. This makes it possible to assign different rates for different

customers, without creating a new tariff for everyone.

Most of the time, customers are only interested in calling a few destinations. Nobody needs

complete A-Z destinations. With Custom Rates, it is possible to adjust rates to the relevant

destinations.

You can reach Custom Rates by clicking the yellow coins icon in the Users window.

From the drop-down list, select the Destination Group for which you want to create a Custom

Rate and click the green Plus Add icon.

NOTE: Custom Rates are available for users whose main Tariff type is Retail.

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Tariff Transformation

Wholesale tariffs can be transformed to user retail tariffs. This can be done in the main tariffs

window by clicking on Make user retail tariff or Make wholesale Tariff (these links

appear when a tariff has at least one rate):

Enter the values you want to add to change from a wholesale tariff to a retail tariff. Let's say X

is the wholesale rate. There are three possible ways of making the change:

1. Enter value Y into the Add amount field. The retail rate will be X + Y.

2. Enter value Z into the Add percent field. The retail rate will be X + (X/100 * Z).

3. Enter value Y into the Add amount field AND Z into Add percent field. The retail rate will

be (X + Y) + ((X + Y) / 100 * Z).

NOTE: When performing this transformation, the largest value from the wholesale tariff is

taken for the Destination group in the retail tariff (by prefix). For a more detailed explanation,

check Make Retail Tariff from Wholesale Tariff explained.

HINT: This function can be used when importing a user retail tariff. First you create a fake

wholesale Tariff, then import Rates into it, and finally transform to the user retail Tariff with

values 0/0.

See also

Make Retail Tariff from Wholesale Tariff explained

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Make Retail Tariff from Wholesale Tariff explained

Retail Tariff from Wholesale Tariff is done by the following algorithm:

1. The highest Rate in Wholesale Tariff for Destinations in Destination Group (DG) is selected.

2. This rate is modified by adding a value or percentage to it.

Example

Imagine we have a Wholesale Tariff with the following rates for the USA:

Prefix 1, rate 0.01.

Prefix 12, rate 0.02.

Prefix 123, rate 0.03.

And we have Destination Group USA with Destinations (1;12;123) in it.

Now, if we want to create a Retail Tariff by adding 0.005 to the Wholesale tariff, we will have a

price for USA Destination Group = 0.035.

For example, the highest rate from Destinations is selected and modified.

Inner mechanics

The wholesale tariff line will have these values:

connection_fee

min_time

increment_s

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When transforming to Retail, these Advanced Rates are created:

Based on which

variable FROM DURATION TYPE ROUND BY RATE Comments

connection_fee 1 event connection_fee

This record is only

created when

connection_fee > 0.

min_time 1 min_time minute min_time rate

This record is only

created when min_time >

0.

increment_s min_time

+ 1

-1 (means

'infinity') minute increment_s rate

This record is always

created.

Why User can't check his Rates

Because it is turned off in Settings.

Show rates for users is disabled by default. Enable it to show the Rate menu option for

resellers/users.

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Devices

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Each device can have many settings. The screenshot shows settings for an SIP device. These

settings are similar to those for other device types as well.

There are several groups of settings: General, Authentication, CallerID, Network Related,

Groups, Voicemail, Codecs, Allowed Addresses, Advanced, Tell Options, Debug and

Recordings.

General

Accountcode – the unique ID of the device in the system.

Description – for informational purposes.

PIN – device PIN for authentication/authorization.

Device group – to which group the device belongs (not used most of the time).

Type – what type of device it is.

Extension – a short number by which this device can be reached; must be unique in

the system.

DTMF Mode – in which protocol phone button presses are sent over network. The

available options are: inband, info, RFC2833, and auto. Choose which one your

provider uses (RFC2833 is used most often).

Works not logged – does the user need to log into MOR GUI to be able to dial out?

(This feature is mainly used for Call Center environments; keep the default in most

cases).

Location – the default is Global. Choose the appropriate location based on

Localization needs.

Timeout – how long to try to reach the device for, and when to hang up if the device

does not answer.

Trunk – No/Yes/Yes with ANI – Is this device Trunk with/without ANI?

Call Limit – how many simultaneous calls can a user make?

Ringing Timeout - allows to limit the ringing duration in seconds. Leave 0 for

unlimited

Authentication

For dahdi devices:

Channel – which channel (or channel group) to use on PRI/BRI/PSTN card. Channels

and groups should be configured in zapata.conf.

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For all other device types:

Username – username you enter in your device.

Password – password you enter in your device.

Authenticate by IP – should the device be authenticated by IP (rather than by

username/password)?

CallerID

A CallerID consists of two parts: Name and Number. The "Number" part is transferred by

default by all technologies (GSM, PSTN, SIP etc), but the "Name" part is transferred only by

some. The number you see on your mobile phone when someone is calling you is the

"Number" part.

Name – the "Name" part of the CallerID.

Number – the "Number" part of the CallerID.

NOTE: if you leave these fields empty, the user can set the CallerID by himself. Usually it is not

advisable to allow the user do it on his own.

Number from DID - use the DID as the CallerID (only available when the device has

DIDs), this option just sets CallerID Number to be equal to DID. Next time

you will edit Device's settings, first option - Number will be checked.

Control by DIDs (CID from DIDs) – only available when the device has DIDs.

This setting lets you control what a CID user can enter in his devices. CID numbers should be

from the set of the device's DIDs. They are mainly used when the user's PBX is connected over

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Trunk and many DIDs are routed to this Trunk. Calls coming out from this PBX must have a

CallerID Number from the set of DIDs assigned to this Trunk (PBX). If "CID from DIDs" is

checked, the system checks whether the device's CID number is from DIDs assigned to this

device. If no DIDs are assigned, this value is ignored. In order for this setting to be active, CID

Name/Number fields must be empty to allow the user to enter any CID number he wants. If

"CID from DIDs" is checked and user enters any CID (not from his DIDs), the system will

change his CID by his first available DID (selected from database by lowest ID).

Control by CIDs - system checks incoming CallerID (Device's CallerID which is set on

the Device). If CallerID matches one of the Device's CIDs - then such CallerID will be

allowed. If it does not match - it will be changed to the selected CID.

Control by Destination - This is advanced option which works by such algorithm:

1. system checks Destination and checks DIDs assigned to calling Device. Then it finds

'nearest' DID to the Destination by subtracting DID from Destination in numerical form

(Example, DID: 11111, Destination: 11112, |DID-Destination|=|1111-11112|=1) If Device

has several DIDs - the lowest value is selected and this DID is set as CallerID.

2. If Device has no DIDs - then User DIDs are used.

3. If Device does not have DIDs and User has no DIDs, then CallerID set in Number

field will be used.

4. If nothing is set here - CallerID coming from Device will be used.

Network Related

Host – the device IP or hostname; the value can be "dynamic", which means that

device should register (it should be like this most of the time).

IP Address – should be the device's IP, or disabled if the device is "dynamic".

Port – the port used to connect to the device (the defaults are 5060 for SIP, and 4569

for IAX2). If host = dynamic, this field is updated by Asterisk when the device registers.

It is the port from where the call is coming (from the device).

Last time registered – when device was last time registered to the Asterisk server.

Media control – can reinvite/transfer. Do you want your server to stay in media path

between your clients and your provider? Please note that this option will work only if it

is supported from both, device and provider, sides and it allows to handle SIP packets

only. Moreover, no codec mismatch between device and provider is available. Default

value is No. WARNING: This option can cause incorrect billing.

NAT – the available options are: yes, no, never, route. For a detailed explanation of

these settings, refer to here [http://www.voip-info.org/wiki/view/Asterisk+sip+nat].

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Qualify – how often to "ask" the device about availability. More details here

[http://www.voip-info.org/wiki/view/Asterisk+sip+qualify] for SIP devices and here

[http://www.voip-info.org/wiki/view/Asterisk+iax+qualify] for IAX2 devices.

Groups

Call Group – to which Call Group this device belongs.

Pickup Group – which Call Groups this device can pick up.

Voicemail

Email – where to send received Voicemail.

Password – the digital password the user enters when he calls the Voicemail number

to hear his messages.

Codecs

Choose the codecs your provider uses.

NOTES:

When no fields are checked, all codecs are available – for example, settings in sip.conf

or iax.conf are effective.

If the Provider and the Device do not have similar codecs, no call can be established.

Allowed Addresses

IP, MASK – permits traffic from this IP only. You can find a detailed explanation here.

[http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+permit-deny-mask]

If you do not clearly understand what these settings are used for, leave them with default

values.

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Advanced

Fromuser/Fromdomain – used when calling TO this peer FROM Asterisk. If you're

using _register=>_ with another SIP proxy, this setting can come in handy since some

SIP networks only allow users in the right domain with the correct user name.

Trustrpid – defines whether or not Remote-Party-ID is trusted. It's defined in

http://tools.ietf.org/id/draft-ietf-sip-privacy-04.txt.

Sendrpid – defines whether a Remote-Party-ID SIP header should be sent. The default

setting is "no". This field is often used by wholesale VoIP providers to provide calling

party identity regardless of the privacy settings (the From SIP header).

Insecure [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure]

o port: ignore the port number where request came from.

o invite: don't require authentication of incoming INVITEs.

o port, invite: don't require initial INVITE to authenticate and ignore the port

where the request came from.

T.38 support - Asterisk does not have good T.38 support, so use this option just for

testing.

SIP 302 Reinvite support - SIP Reinvite support for the device. Disabled by default.

Progressinband:

o yes - when "RING" event is requested, always send 180 Ringing (if it hasn't been

sent yet) followed by 183 Session Progress and in-band audio.

o no - send 180 Ringing if 183 has not yet been sent, establishing an audio path.

If the audio path is established already (with 183), then send in-band ringing

(this is the way Asterisk historically behaved because of buggy phones like

Polycom's).

o never - whenever ringing occurs, send "180 ringing" as long as "200 OK" has

not yet been sent. This is the default behavior of Asterisk.

Video support – does your provider support Video over IP? More info here

[http://www.voip-info.org/wiki/view/Asterisk+video].

Allow duplicate calls - the default setting is "no".

Language - sets IVR language

Use ANI (if available) for CallerID: - When the call comes the information about who

is calling is found in CallerID field. This field is used to determine who is calling. When

the call comes through PRI/SS7 channels - then additional information is available who

is calling in ANI field (in call's data channel) so sometimes CallerID might be empty or

anonymous, but the caller can be found in ANI field. This option allows to use ANI field

as CallerID to determine and recognize who is calling.

Incoming Call CallerID Presentation - sets CallerID Presentation. More information

can be found here [http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerPres]

and here [http://www.voip-info.org/wiki/view/Asterisk+cmd+CallingPres]

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Change Failed Code To - if call fails change Hangup Cause Code to this value. This

works only for outgoing calls from device. Not for Incoming.

Forward DID - it allows to forward call to DID which is assigned to Authorization by

PIN or Calling Cards dial plan. After user enters PIN of any device or card, call gets

connected with destination.

Transport - lets you choose protocol(s) for data transfer. Appears only if device is SIP

and when Asterisk 1.8 is enabled. Default value: udp.

Tell Options

Tell balance – should MOR tell the user his balance every time he tries to dial? The

default setting is "no".

Tell time - should MOR tell the user his remaining time every time he tries to dial? The

default setting is "no".

o Time is told in MINUTES and SECONDS. Currently it is not possible to tell ONLY

in minutes.

Tell remaining time when left – when some time is left, MOR will tell the remaining

time to talk (in seconds).

Repeat remaining time when left – repeats the remaining time when some time is

left (in seconds).

Debug

Process SIPCHANINFO - shows SIP channel info in Asterisk CLI and saves this information

on database.

Save Call Log - saves whole call log on database. Log can be overviewed by click in

Last calls.

NOTE: debug should be enabled only if you are experiencing any problems. It should be

disable in any other cases, because it stores lot of information on database.

Recordings

This section is available when Recordings Addon is installed in the system.

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See also

Grace time - available starting with MOR 9

H323 Device settings

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Extension lines

In device settings, you can select Extension lines. It will show you Asterisk extension lines:

This information is for people who have a deeper understanding of Asterisk internals.

Ring several devices at same time

Currently MOR does not DIRECTLY support a configuration to ring several devices at the

same time.

Such functionality can be implemented using: PBX Function External DID

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How to create User and Device

Here [http://wiki.kolmisoft.com/index.php/How_to_create_User_and_Device] you can see a

video demonstrating how to create a user and a device WITHOUT setting additional settings.

Make sure you read and understand User Details and Device Settings

See also

Users

Devices

Trunks

"Trunk" refers to another server/PBX that can be connected to the MOR system.

When calling such a device, we send the full number that has already been dialed.

For example, when dialing a simple device, we would use:

Dial(SIP/some_device)

If this device is a Trunk, we use:

Dial(SIP/some_device/${EXTEN}).

This lets Trunk decide to which device (connected to the Trunk) to route the call.

If you want to make a device Trunk, just mark it as Trunk in the device details: No/Yes/Yes

with ANI.

Trunk with ANI

It is sometimes necessary not to bill a whole trunk as one client, but to bill separate clients

connected to the server/pbx (trunk) separately.

In order to do this, you should mark a device as Trunk with ANI.

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Now, when a device connected to the Trunk dials through MOR, MOR checks the calling

device's CallerID. If the CallerID is present for some device in the MOR, MOR recognizes this

device as a valid device on the MOR and assigns the call to it instead of to Trunk.

See also

PBX connection to MOR

o MOR and Trixbox

CLIs (Incoming CallerIDs)

CLI or CallerID is a number from which a user can dial MOR. These values are used when the

user is dialing into MOR through Trunk/Provider with ANI (Automatic Number Identification).

It is important to understand that CLI is not the CallerID that is put on a call that goes out of

the system.

CLI is the CallerID for a call that is COMING INTO the system.

It is mainly used to AUTHENTICATE the caller.

You can reach the main CLI window in device settings by selecting CLIs:

IMPORTANT! CLIs should be in E.164 format!

It is possible to ban CLIs. This is explained ANI/CLI ban system topic.

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ANI/CLI ban system

This function makes it possible to ban unwanted CallerIDs.

SETTINGS –> Users –> Devices –> CLIs

When no IVR is selected, the call ends with the CONGESTION tone.

When an IVR is selected, it is played for the user who tries to dial.

As example IVR might say: "Sorry, but your account is blocked because..."

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Blocking CallerID because of simultaneous calls

To set this functionality go to main devices edit window and in Advanced block

choose Yes in Block callerid if (number) simultaneous calls come from it:. Add a number of

simultaneous caller IDs. Default value is 2:

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This scheme shows how this functionality works:

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Duplicate call prevention

Sometimes, devices have poorly implemented SIP protocols and try to initiate the same call

twice at the same time.

The "Duplicate call prevention" option is implemented to avoid such a situation.

It kills both calls and prevents the malfunctioning device from dialing.

Allow duplicate Calls is available for Providers and Devices in the Edit window under

Advanced settings.

Its default setting is No. If you are not sure what you are doing, leave it that way.

Allow loss calls

Allow loss calls can be put on a per user basis:

This setting allows you to decide whether you want to allow a user to use a provider if the

provider's rate to a destination is lower than the user's rate.

By allowing such a call, you will pay more to the provider than the user pays you for it,

meaning that you will take a loss on the call.

If this setting is unchecked (the default setting), cheaper providers will not be used in dialing.

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Resellers

Resellers can check this setting for their own users.

When a Reseller checks this setting for his users, the user is allowed to dial out even when his

rate is lower than the rate that the reseller is paying the system admin.

See also

204 No suitable providers found

218 Reseller does not allow loss calls

Device groups

It is possible to group devices. This feature is intended for branches of a company, so that

each device will belong to a single company group.

This functionality is not widely used and we do not recommend using it unless you really

need it.

To access groups for each user in the Users window, click Dev.Groups:

In this window you see all device groups for the selected user. Under each group, you see

devices assigned to the group.

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Once new groups are added, they can be deleted. However, the primary group cannot be

deleted.

Edit Details – lets you a change group's name and address.

Call Flow

Definition

Call Flow is set of rules on how a call should behave when it is trying to reach a device.

Call Flow can be reached by clicking Call Flow link in device details:

A call has several states that can be seen from this image:

Before Call – call has not reached device.

Call – call to device is in progress.

Answered – call was answered.

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No Answer – call was not answered and ended after timeout.

Busy – callee was busy.

Failed – call to device failed.

Actions

We can take actions for several call states: Before Call, No Answer, Busy and Failed.

The possible actions are:

Empty

Forward

Voicemail

Fax detect (only for the Before Call state)

Empty

Empty – means no action should be taken. It is the default in most cases.

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Forward

Call can be forwarded to Local device or External number:

Device Timeout is only available in the No Answer state. It decides for how long a device

should ring before the system decides that it is unreachable (when to activate the No Answer

state).

The system owner (admin) can select any device he likes. If a user edits his devices' Call Flow,

he can only select his own devices.

Using the "forward" action it is easy to implement:

Unconditional forward – when forward is used in the Before Call state.

Forward on no answer/busy/failed – when forward is used in No Answer, Busy or

Failed states.

CallerID options

The following options are available to manage CallerID for a forwarded call:

From device - puts CallerID for the device that is forwarding the call.

Same as comes - unchanged - leaves the CallerID as it comes from the original caller.

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From DID - selects a CallerID from available DIDs.

Custom - enters any CallerID.

Forwarding example

Let's make a chain of forwards: when someone calls to a device (extension) and until the

device timeout there is no answer the call is forwarded to another device, if there is still no

answer then forwared to the next extension.

In Forward to select the second extension. Then go to Devices main page and make the same

call flow to that second extension just in Forward to select the third extension:

the number of rings before forwarding can be changed in /etc/asterisk/indications.conf.

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Voicemail

This action sends the call directly to voicemail. It's simple. Put in Before Call, and if a call never

reaches a device, it will go directly to voicemail.

You can put it in the No Answer/Busy/Failed state to achieve a different functionality.

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Fax detect

This action can be set ONLY in the Before Call state and only if a Fax device is enabled in

Settings:

You can select only a FAX device to route the call if a Fax is detected. MOR will use fax

detection when this action is set.

If a fax is detected, it routes calls to the set fax device. Otherwise, they go to the basic device.

NOTE: Fax over VoIP is not reliable and this setting should be used only for testing.

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Which codecs should I choose for devices

What codecs are supported?

Audio:

G.711 A-law

G.711 u-law

G.723.1

G.726

G.729

GSM

iLBC

LPC10

Speex

ADPCM

16 bit Signed Linear PCM

Video:

H.261

H.263

H.263+

H.264

Audio codecs

Most commonly used are:

G711

G729

G711

PSTN quality codec which is supported in 99% of all devices/providers (but often disabled to

save traffic).

It has best quality but highest bandwidth (64kbps + ~20 kbps overhead ~= 84kbps to one

direction).

It comes in 2 flavours - G711u (ulaw) - used in USA, and G711a (alaw) - used mostly in Europe

and other countries.

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G729

Little worse quality but is often used because it saves traffic (~8kbps + ~20kbps overhead ~=

28kbps to one direction).

It takes a lot of CPU power to transcode so total call count on the server is greatly reduced

using this codec.

It needs to be installed separately and it is not free.

G722

G.722 is a ITU-T standard 7 kHz wideband speech codec operating at 48, 56 and 64

kbit/s. 64 kbit/s (comprises 48, 56 or 64 kbit/s audio and 16, 8 or 0 kbit/s auxiliary data)

HD VoIP in the Asterisk world involves selecting the G722 codec for VoIP calls.

G722 is known as a wideband codec as opposed to g711 which is narrowband.

The nice thing about it is that it does not require any more bandwidth than G711.

Both require 64kbit/s each way for a 2-way conversation.

The difference is that G722 employs compression combined with a higher sample: rate

to obtain a practical acoustic bandwidth of 50 Hz-7000 Hz as opposed to G711 which

has a practical acoustic bandwidth of 200-3000 Hz.

The difference in voice quality is quite noticeable.

Instead of sounding like you are talking into a tin can it now sounds like the other

person is in the same room.

You can hear more subtle nuances in the person’s voice and certain letters/syllables

that are hard to distinguish on regular calls become much easier.

If one endpoint is using G722 and the other is using G711, the endpoint using G722

will hear better sound quality from the other end but the G711 end will not notice

much if any difference.

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Transcoding overhead between G722 and g711 is minimal. Roughly equivalent to half

the transcoding overhead between G711 and GSM. G.722 patents have expired, so it is

freely available.

Video

Read manual for your device to check which codec is supported on your device.

H263/H264 are most popular.

Make sure same codec is chosen between devices you are trying to communicate.

Fax

If your system/provider/devices does not support T38 you should use G711 codec for fax

transmitting/receiving.

Notes

Make sure you have always selected audio codec! Otherwise you will not be able to place

calls!

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Change Audio/Video Codecs priority

Starting from MOR 9 you can specify audio and video codecs priority. To specify the priority

you have to drag and drop codecs upwards and downwards. Remember to tick the codecs

you want to be enabled!

In this example only the codec G.711 A-law is enabled. Remember to tick other codecs you

want to be enabled.

Additional Notes

Codecs priority can be adjusted in Setup -> Settings -> Default Device (this

configuration will be applied to all new devices)

Codecs priority can be adjusted for a single device also: just navigate to a required

device edit and change the priority. You can find all Devices in Users -> Devices (hint:

you can use search from this menu). Also you can navigate to required device by

user - go to users list and select the Device icon, then select the required device for

that user.

In MOR 8 and older versions you can only enable/disable codecs. You can't prioritize

codecs in older versions.

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Authentication

There are two types of authentication in MOR: IP authentication, and using login and

password.

IP authentication means accepting calls from a given IP automatically, without requesting any

additional login data.

To use login and password authentication, the sender needs to register his phone or pbx to

MOR before sending calls, otherwise MOR will reject them.

Note: IP and login/password authentication cannot work together for the same device

or provider.

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IP Authentication

This example shows how to configure device to use IP Authentication:

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Restrictions

You cannot create/edit Provider if you already have Device with same IP and port

combination. However, you can have two Providers with same IP and port

combination.

You cannot have two Devices with same IP and port combination.

NOTE: if you need to send and receive calls from same IP and port, please Configure Provider

which can make calls.

Registration Status

About

This feature was developed to make MOR administrator’s life easier. With this feature an

administrator is able to track SIP device status from GUI and solve problems faster.

Technical details (requirements)

These requirements must be fulfilled in order this functionality would work correctly:

Works only with SIP devices.

Device must try to register or someone should call to that device in order that device

status would be visible.

Device must register to an Asterisk server where it is assigned.

Example:

If a device is assigned to a server A but registers to server B - status will not be shown

(devices must register to the server they are assigned to).

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Status

OK - Device is registered, all is OK

LAGGED - Device is registered but connection to it is poor

UNREACHABLE - Device tried to register before, but currently it is unreachable

UNKNOWN - means the device state status could not be found. This is NOT an alternative

message to the previous UNREACHABLE message.

This status is created when you add a new device and the system does not yet have any data

about its registration status.

Unmonitored - Device should not register to server so its status is unmonitored, this

device made calls, so server is aware of its existence

NO STATUS - Device never tried to register to Server and Server does not know nothing

about this device.

Device list

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Device edit

Very technical details

Most probably you will never need these.

These files must exist:

/usr/local/mor/mor_retrieve_peers

/etc/cron.d/mor_minute_actions

Logs

If you want to get a lot of logs put DEBUG = 1 in mor.conf

Logs for this feature can be found here:

/var/log/mor/ami_debug.log

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Device rules

Device Rules transforms an E.164-compatible number to any number

format the Device requires.

Description

Different Devices expect different number formats from you. You can send them whatever

they want using Device Rules.

Device Rules are only used when Device is reached over DID. Local calls between Devices will

not use Device Rules.

Device Rules can be reached by clicking the icon next to a Device in the Devices window:

IMPORTANT: Device Rules are applied ONLY to those devices which are accessible over

DID!

We see a similar window as the one in Localization. The rules to manage Device Rules are

identical to those for Localization management, which are covered in Number Manipulation

page.

Destination localization rules

IMPORTANT: Localization Rules only matter to the Trunks (Devices marked as Trunks).

Because simple Devices do not care about Destination.

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Let's say our device/trunk expects to get a number with the international prefix 00. We can do

this in following way:

That's it – very simple. Now every number that is sent to this Device will be transformed by

adding our created int. prefix.

For example, if 37063011111 should be sent to this Device, this rule will be activated and the

number 0037063011111 will be sent to Device.

Special cases

Cut everything till Hash

With rule Cut: *# it is possible to cut everything till # (# included) in the dialed destination:

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For example:

Dialed number: 999#37063042439.

After Localization: 37063042439.

This function can be used in fancy MNP (Mobile Number Portability) schemes and perhaps

elsewhere.

NOTE: Add value still can be used to add any value after Cut is used on the Destination.

Destination same as received

If you want to send a number to a Device exactly as it was received, set Cut = '-' (minus sign

as shown in the screenshot).

That way, the same number that comes to MOR also leaves MOR:

CallerID localization rules

Source Rules manages CallerID which is sent to this particular Device.

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In the example bellow we will add international prefix to every CallerID sent to this Device:

Special case

CallerID overwrite

Sometimes Trunks only accept calls with special format CallerIDs.

It is possible to overwrite any CallerID before sending a call to such a Device.

In Device Rules (for this Device) set up the Rule:

In this example, all calls going to this Device will have CallerID = 37063033333.

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Providers

Description

In the old PSTN world, Providers were called Trunks. Providers are your route out of your VoIP

network to the outside world.

Providers can be one of several types: ZAP, SIP, IAX2, and H323. The type determines which

technology is used to connect to the Provider.

At the very start, you need to create a Provider through which you will dial out to PSTN or

other VoIP networks.

Every Provider will charge you for calling a destination through his service (although the rate

might be zero). The rate to every destination is different. Because of this, we need to have a

Tariff (or Price List) for each Provider. This Tariff will tell our billing what price we will pay to a

destination when using this Provider's services. In other words, this will be our Selfcost.

Providers' configurations can be found in SETTINGS -> Billing -> Providers.

To create new provider, click New Provider.

Here, enter the Name for a new provider, select its Technology, Tariff, Asterisk server which

should be used for this provider and click Create.

NOTE:

You can't create a new Provider if there are no Tariffs available for Providers.

A Provider is also able to send calls, not just receive them. In case you want to sell

services to a provider you have to create such a provider.

Provider balance

Provider balance increases when calls are made via that provider. Currently it is not available

to decrease provider's balance. You can read more about Provider Billing Addon.

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Settings

General

Name – a name for the provider, for informational purposes.

Technology – out of SIP/IAX2/ZAP/H323, choose which technology your provider uses

- that is, the way you connect to the provider.

Tariff – the list of rates the provider charges you.

DTMF Mode – the available options are inband, info, RFC2833, and auto. Choose the

one used by your provider.

Location – which Localization rules are set to apply to INCOMING calls COMING FROM

this provider.

Automatic Number Identification - used in a special case explained here: Provider with

ANI

Ringing Timeout – this setting lets you set for how long this provider should be dialed

before giving up. The default value is 60 seconds, and the minimum value is 30

seconds.

Call Timeout - this setting lets you set the duration of the whole call: ringing and the

period after connection.

Device ID – informational data. No longer used starting from MOR 0.8.

Call limit – explained here: Simultaneous call limitation.

Hidden? - hide/unhide provider

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Authentication

For ZAP providers

Channel – which channel (or channel group) to use on a PRI/BRI/PSTN card. Channels

and groups should be configured in zapata.conf.

For SIP/IAX2 providers

Login – username if your provider asks for it.

Password – password used for authentication by your provider.

Authenticate by IP – just check the checkbox (Login/Password can be left empty).

o Make sure you put the correct Hostname/IP address!

Register – should we register the provider? (option is not visible when Dynamic is

selected in Network related section)

Register extension – if the provider asks to use an extension for registration (usually

they don't), enter it here.

Provider CallerID

CallerID – consists of two parts: Name and Number. The "Number" part is transferred by

default on all technologies (GSM, PSTN, SIP etc), but the "Name" part only on some. The

number you see on your mobile phone when someone is calling you is the "Number" part.

Name – the "Name" part of CallerID.

Number – the "Number" part of CallerID.

NOTE: if you leave these fields empty, calls coming from this provider will have CallerID set by

the Provider. It should almost always be this way.

Network related

Hostname – hostname of the provider.

Server IP – the provider's IP. The value can be dynamic, which means that provider can

change its IP. To discover this value, do ping to the provider's hostname.

Port – port used to connect to the provider (default: 5060 for SIP, 4569 for IAX2, 1720

for H323).

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H323 specific network settings

Fast Start – an option specific to H323; it is either on or off.

h245 Tunneling – an option specific to H323; it is either on or of

SIP specific network settings

Media control – can reinvite/transfer. Do you want your server to stay in the media

path between your clients and your provider? Disable if you have a lot of 1-second

answered calls.

NAT – the available options are: yes, no, never, and route. For a detailed explanation of

these settings, refer to here [http://www.voip-info.org/wiki/view/Asterisk+sip+nat].

SIP and IAX2 specific network settings

Qualify – how often to "ask" the provider about availability. More details here

[http://www.voip-info.org/wiki/view/Asterisk+sip+qualify] for SIP providers and here

[http://www.voip-info.org/wiki/view/Asterisk+iax+qualify] for IAX2 providers.

Servers

Allows to select on which Asterisk servers this provider should be used.

Codecs

Choose the codecs your provider uses.

NOTE: When no fields are checked, all codecs are available. ( For example, settings in

sip.conf or iax.conf are effective).

Advanced

Grace time - if call duration is less than Grace Time, it will not be accounted.

Progressinband:

o yes - when "RING" event is requested, always send 180 Ringing (if it hasn't been

sent yet) followed by 183 Session Progress and in-band audio.

o no - send 180 Ringing if 183 has not yet been sent, establishing an audio path.

If the audio path is established already (with 183), then send in-band ringing

(this is the way Asterisk historically behaved because of buggy phones like

Polycom's).

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o never - whenever ringing occurs, send "180 ringing" as long as "200 OK" has

not yet been sent. This is the default behavior of Asterisk.

Video support – does your provider support Video over IP? More info here

[http://www.voip-info.org/wiki/view/Asterisk+video].

Allow duplicate calls - the default setting is "no".

Interpret NO ANSWER as FAILED - default option is no (This will change the provider in

LCR like it is failed, in last calls you will see the real hangupcause - no answer).

Interpret BUSY as FAILED - default option is no (This will change the provider in LCR

like it is failed, in last calls you will see the real hangupcause – busy).

Fake Ring - should system generate Fake Ring or not? Use only if your Device/Provider

has bad protocol implementation and does not send RING signal properly.

Common Use - Provider can be used by Resellers which are able to create their own

Providers. Reseller will be billed using this Provider. If provider is marked as common

use and is used in one of reseller's LCR or is in Terminators list - admin is not allowed

to delete this provider or remove common use option.

SIP Specific

Fromuser – more details:

[http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+fromuser]

Fromdomain – more details:

[http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+fromuser]

Trustrpid – This defines whether or not Remote-Party-ID is trusted. It's defined

in http://tools.ietf.org/id/draft-ietf-sip-privacy-04.txt

Sendrpid – defines whether a Remote-Party-ID SIP header should be sent. It defaults

to "No". This field is often used by wholesale VoIP providers to provide calling party

identity regardless of the privacy settings (the From SIP header).

Insecure – more details:

[http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure]

T.38 Support – should we support T.38 pass-through?

SIP 302 Reinvite Support – turns this SIP feature on or off.

P-Asserted-Identity usage - The P-Asserted-Identity header field can be used to

convey the proven identity of the originator of a request within a trusted network.

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Debug

Save Call Log - when this option is enabled you can access Call info information in last

calls page.

Failover provider

You can also assign a failover provider to a LCR by selecting No in Use only one provider (Turn

off failover). Default value is Yes.

Failover provider is a provider used only in case of emergency. If all the other providers fail

(returns FAILED error code) a failover provider is used.

If Provider returns NO ANSWER or BUSY - it means, that call not FAILED.

If you think that NO ANSWER/BUSY == FAILED, then you can use NO ANSWER/BUSY

interpretation for providers setting.

See also

NO ANSWER/BUSY interpretation for providers

H323 Provider settings

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Terminators

A Terminator is a group of Providers (in MOR's terms).

In reality, a Terminator is a company that sells/buys traffic to/from you. Then Providers (in

MOR's terms) are gateways of this Terminator.

Terminators are used in reporting - in Aggregate and Summary reports.

Provider with several IP

If a VoIP Provider gives you several IP addresses to which to send traffic (or from which to

receive traffic), you need to create a SEPARATE Provider in MOR for each IP address.

After that, you can unite them under Terminators.

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Provider connection testing

It is possible to test Provider connectivity by clicking on in the Providers window:

Good response:

Bad response:

NOTE the NOTE!

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H323 Provider settings

Provider with ANI

First scenario

Automatic Number Identification (ANI) for Provider is used ONLY (at least we don't know any

other situations) when the provider gives you an access number after which (IMPORTANT!)

the dialer enters the destination number WITHOUT PAUSES, so that the WHOLE number is

sent to the provider instantly.

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For example:

Let's say the provider gives you the access number 1234.

When somebody wants to dial a destination (let's say 22234567890), he enters on his

phone: 123422234567890.

22234567890 comes from the provider to your MOR box.

Because the provider is with ANI, we check the user's CLI.

If CLI is found, we know which user is dialing.

Before he dials out, we change the CallerID of this call to the CallerID found in our

database - that's a minor job, but it's part of the algorithm.

After that, the call is sent to 22234567890.

IMPORTANT!!! - Do not check ANI for Provider if this description does not match your

working scenario!

Similar scenario which can't be used with ANI for Provider

If you have an access number as DID, you can enter p (pause) after your DID when calling the

provider and then enter the destination. But that is a totally different scenario. It should be

configured with DID + Auth. by ANI DialPlan, because the destination is send to MOR by

DTMF, not by the provider as in the previous example.

Example:

If your DID is 234567890 and you want to dial dst: 987654321 on (for example) your mobile

phone, you enter:

234567890p987654321

That way the call will reach MOR. When p (pause) is reached, MOR will play "Please enter

destination" and 987654321 will be sent by DTMF to MOR.

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Configure Provider which can make calls

In order to allow a Provider to make calls, we need to:

1. Create a new User for it.

2. Assign the Provider to this User.

Video that shows how to do this can be found here:

http://wiki.kolmisoft.com/index.php/Configure_Provider_which_can_make_calls

Configure Provider with dynamic IP

In Providers settings, enter the appropriate values in the following fields:

Login - the username for your device.

Password - the password for your device.

Make sure IP Authentication is unchecked.

Make sure Register is unchecked. This option is not visible in MOR 10 when Dynamic

selected

In the Hostname field enter: dynamic.

In the IP Address field enter: dynamic.

Check NAT = yes if the device is under NAT; check NO if the device is not under NAT.

Check Qualify = YES.

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For example:

Make sure you enter the correct Login and Password!

Save your changes.

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When your device registers, the current IP address of the device will be in in the IP Address

field. Do not change it!

Simultaneous call limitation

It is possible to limit simultaneous calls per provider/DID/user/device basis.

These settings limit simultaneous calls, not how many total calls the

user/device/provider(/did) can make.

When no more calls can be made, the dialer hears the BUSY signal.

Provider

Call limit - describes how many simultaneous calls the provider can make. 0 (zero) means that

calls are unlimited.

Hangupcause code 223 means that the provider can't make more simultaneous calls.

This limit is for outgoing calls through this provider.

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User

Call limit - describes how many simultaneous calls the user can make. 0 (zero) means that

calls are unlimited. This limit is for outgoing calls made by this user.

Hangupcause code 220 means that the user can't make more simultaneous calls.

Hangupcause code 222 means that a reseller (as a user) can't handle (allow to his users to

make) more simultaneous calls.

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Device

Call limit - describes how many simultaneous calls the device can make. 0 (zero) means that

calls are unlimited.

Hangupcause code 221 means that the device can't make more simultaneous calls.

This limit is for outgoing calls made by this device.

DID

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Call limit - describes how many simultaneous calls DID can receive. 0 (zero) means that calls

are unlimited.

Hangupcause code 219 means that DID can't receive more simultaneous calls.

This limit is for incoming calls made to this DID.

Special case

When DID points to a device and the limit is reached, calls will be routed to the device via

BUSY - Call Flow. This will ring another device or send a call to VoiceMail when the main

device already has a call.

NO ANSWER/BUSY interpretation for providers

By default, the NO ANSWER or BUSY call end disposition is counted as the end of a call.

After this it is pointless to try to dial the same destination again, because the destination is

not answering (NO ANSWER) or is currently busy (BUSY).

If we have several providers in our LCR, the call ends after we receive NO ANSWER or BUSY

from a provider, and our routing ends.

This is okay for an ideal world in which all providers follow standards and send the correct call

end codes, e.g.:

NO ANSWER when calls are not answered after a certain period (timeout).

BUSY when a recipient is busy (talking with somebody else).

ANSWERED when a call has been answered and ends.

FAILED when an error occurs.

However, providers sometimes cheat (or do not have technical skills to make this correctly).

For example, if they can't deliver a call to a destination because of some problem in their

network or something similar, they may send the NO ANSWER or BUSY signal for a FAILED

call. And then our routing breaks. If we have several more providers in our LCR, our LCR will

not try to send call the over the remaining providers and will end the call.

This often happens for GREY (illegal) routes.

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To partly solve this problem, MOR provides a workaround: "NO ANSWER/BUSY interpretation

for providers".

This makes it possible to set a NO ANSWER and/or BUSY interpretation as FAILED.

That means, if a provider reports NO ANSWER and/or BUSY, MOR treats this as FAILED and

the next provider in LCR (if there are further providers) is tried to send the call.

These options can be set in Providers Settings:

It is advised to use these settings ONLY and ONLY on confirmed

cheating/misconfigured providers. That is, these settings should usually be 'NO'.

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Net2phone configuration

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Broadvoice configuration

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Eutelia

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As Eutelia sends calls from different IPs, we need to configure it as a second provider:

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Configuring DIDWW

DID World Wide, International DIDs forwarded to PSTN and VoIP by SIP, IAX, H323, Skype,

Gtalk, MSN Messenger. Origination services, SIP DIDs.

Known DIDWW issues

CallerID does not come in E.164 format - only in the local format.

CallerID does not come in E.164 format, only in local format

QUESTION:

Hello! From Ukraine DID CallerID is not coming in the E.164 format.

It is coming in the local format like 80519539xx, but must be 3805019539xx.

Is it possible to fix this?

ANSWER:

Your DID number is forwarded with CLI exactly as the local carrier sends it.

Please check if it is possible to change the CLI on your end.

Best Regards,

Customer Support Department

DIDWW

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Configuring multiple DIDWW IPs for receiving calls to DID

The problem with this DID Provider is that it sends calls from many IP addresses.

This is done due to load balancing techniques.

Please be sure you have granted access on your end to DIDWW Public IP Addresses:

The newest list can be found at DIDWW IP Adresses:

[http://www.didww.com/Knowledgebase/the_list_of_didww_ip_addresses/]

It is possible to implement multiple DIDWW IP addresses by making these changes:

1. Upgrade your MOR 9 GUI

2. Create a provider in MOR, with a name didww:

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3. Afterwards, enter any IP (or leave blank) in Provider Network settings, and write down

Provider Device ID value, you will need it later.

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4. Now connect to your server over ssh and locate this file

/etc/asterisk/extensions_mor_didww.conf and find these lines:

[from-didww]

exten => _X.,1,Set(CDR(ACCOUNTCODE)=0)

exten => _X.,2,Goto(mor,${EXTEN},1)

Now you need to change (ACCOUNTCODE)=0 value (0) to Provider Device ID value as

follows:

exten => _X.,1,Set(CDR(ACCOUNTCODE)=8)

exten => _X.,2,Goto(mor,${EXTEN},1)

5. Make sure, file /etc/asterisk/sip.conf has this line included:

#include sip_didww.conf

6. Restart Asterisk

Now your system is updated with all the DIDWW IP addresses that are listed in their

recommendations.

Now there is no need to create one Provider for each IP

DID configuration

Then:

Create DID - IN E.164 FORMAT!

Assign to a DIDWW Provider.

Assign to some device.

Configure correct routing (mapping) in www.didww.com.

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Configuration in www.didww.com

We will assign a Thailand, Bangkok number to our server:

First we need to create "mapping", that is, the route to our server:

After this, we need to assign our DID to our just-created "mapping":

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Troubleshooting

Watch Asterisk CLI to see what's happening.

Maybe you have DID in the wrong format - for example, not E.164 compatible?

Vitelity

http://www.vitelity.com/

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Vono

http://www.falevono.com.br/

This Provider requires the following Provider Rules:

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How to add Voicetrading Provider

About

Voicetrading Provider configuration in MOR GUI.

This manaul is an example how to configure Voicetrading provider, but it can be used

as an example of another Providers configuration which authentication type is registration by

username/password.

The username & password can also be used to connect to our SIP-server in order to make

calls.

Getting connected

Technical details which Voicetrading provides to you:

Username: Your VoiceTrading username

Password: Your password

SIP/Proxy registrar: sip.voicetrading.com

We support the following codecs:

G.711 (64 kbps)

G.726 (32 kbps)

G.729 (8 kbps)

G.723 (5.3 & 6.3 kbps)

GSMFR (13 kbps)

iLBC (20ms & 30ms)

If you cannot connect by using Username & Password, we can also authenticate your fixed IP

address(es).

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After we authenticated your IP, you can connect by using the following IP-adresses:

For SIP: 77.72.169.129 or 77.72.169.131

For H323: 77.72.170.132

Configuration in MOR GUI

To add Provider in system go:

Billing -> Providers -> New Provider

Enter name of Provider.

Select Technology (protocol) which will be used.

Choose tariff and mark server which will be used for this Provider and press create.

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Entering Provider details

Fill fields login/Username, password with details your Provider provided to you.

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Different quality routes

Voicetrading and another Providers often uses different prefixes for different quality routes:

VoiceTrading EUR/USD Grey: 00001 49 12345678

VoiceTrading EUR/USD Standard: 00 49 12345678

VoiceTrading EUR/USD Premium: 00000 49 12345678

How to add special prefix for Provider

Example you buy from VoiceTrading EUR/USD Standard 00 49 12345678 traffic.

VoiceTrading EUR/USD Standard route which need special prefix 00

To add special prefix for Provider, Provider rules are used.

To configure it using Provider rules.

Press on Rules icon and enter rule add special prefix in this case 00

When Rule is added cal will work in this scenario: customer will dial 49 12345678 system will

automaticly add special prefix 00 and number will be send to Provider in this format

004912345678.

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Portech GSM as Provider

Connect Portech GSM as Provider to MOR

We will connect like this:

Phone will dial to MOR

MOR will use Portech as Provider to reach GSM network

Provider configuration

Make sure you do not have Device with same username/password as for your

Provider!

In this example Portech has dynamic IP.

So we need to register it to MOR.

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Create Provider (NOT user/device!!!) in MOR with following settings:

NOTE: password field is empty - somehow in our tests Portech refused to register with

password.

Create LCR, assign Provider to it, assign LCR to user, and do all other necessary steps.

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Portech configuration

Configure Portech to register to MOR:

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Port settings

Set different ports for separate SIMs:

Do not forget to route these ports on the firewall if your Portech is under NAT!

Problems

We do not recommend these devices because they are unpredictable and unstable. They may

stop working without notice at any time!

Some examples:

http://lists.digium.com/pipermail/asterisk-users/2009-November/241371.html

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Call Routing

LCR

LCR (Least Cost Routing) is a set of Providers that should be used to send a call when trying

to dial any destination - for example, by the cheapest rate.

The correct name for such a function is a Routing Table, because the function supports other

ways of routing, not just finding the cheapest route.

LCR supports fail-over. That is, if one provider fails, the next one is dialed. This is completely

invisible for the caller and lets you increase the chances of the user reaching his destination. If

one Provider is not capable of delivering a call, the system will try to use the remaining

Providers.

The LCR window can be found in SETTINGS –> Billing –> LCR.

Supported routing modes:

By Price - cheapest first (Least Cost Routing).

By priority.

By percent or Weight-based routing.

The user has one base LCR, so that all his calls are routed by some preset rule.

It is possible to route different destinations for a User using Call routing per destination basis.

Make tariff from LCR

In MOR there is also a feature that gets you a CSV with the best prices selected from the list

of Providers in LCR. Go to SETTINGS -> Billing -> LCR. You will see this window:

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Click on the icon and the download should start automatically.

Copying LCR between resellers

Sometimes you need to make a LCR which takes quite a long time to make. If you have made

a similar one before you can just copy it and change appropriate details. Go to SETTINGS ->

Billing -> LCR -> Copy LCR (as shown below in the picture). You willl see this window:

Choose Original LCR reseller and Cloned LCR reseller and click Clone button.

Please note that LCR will not be allowed to copy if reseller B is not allowed to use Providers

which are used by Reseller A.

See also

LCR Logic

How to add Provider to LCR

Call Routing by priority (Manual LCR)

Call routing per destination basis

LCR/Tariff change based on call prefix

Routing by Percent or Weight-based routing

Why was a call not routed through another provider/trunk?

Make Tariff MIN/MAX from LCR Providers to CSV

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LCR Logic

Here simple LCR (Least Cost Routing) is explained.

Example

We have two providers, A and B.

The dialed destination is 921XXXXXXXXXX.

Provider A has the following rate for the prefix (destination):

92 rate 0.1

No other destination starts with 92.

Provider B has the following rates for prefixes:

92 rate 0.05

921 rate 0.2

So LCR in MOR will group these providers in the following order:

Provider A with rate 0.1 (92)

Provider B with rate 0.2 (921)

MOR first dials through Provider A. If the call fails, Provider B will be used to send the call.

If the call through Provider A is ANSWERED, BUSY or there's NO ANSWER, the call ends, and

Provider B is not used.

Provider B is used ONLY when the call fails through CONGESTION.

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The following graphic illustrates LCR logic:

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How to add Provider to LCR

Video at [http://wiki.kolmisoft.com/index.php/How_to_add_Provider_to_LCR] will show you

how to add providers to LCR.

Call Routing by price

About

LCR by price routes the calls using the cheapest routes.

Usage

1. Go to SETTINGS -> Billing -> LCR

2. Create new LCR by pressing the New LCR

Advanced Options

Do not use next available provider if the price is higher than x%

Use this feature when you do not want to use a provider when the price between the

cheapest and next available provider is higher when x %. The call will simply fail if no other

providers meet the requirement.

Usage

1. Go to LCR list: SETTINGS -> Billing -> LCR

2. Press Edit on LCR by price LCR:

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3. Enter the desired percent for call price difference:

Example 1

Provider A has a price of 0.10€ and provider B 0.11€ and we set x=20%

so if call can´t be establish by provider A then system will try with

provider B (0.11<0.10+20%)

Example 2

Povider A has a price of 0.10€ and provider B 0.30€ and we set x=20%

so if call can´t be establish by provider A then the system wil NOT try with provider B

Hangup Cause Code

When Provider is skipped, HGC 235 is shown.

Call Routing by priority (Manual LCR)

Call Routing by priority means that a call is routed by a predetermined order, not by the

cheapest price for the destination.

This method of routing is used when it is known that a Provider provides better quality for

some destinations, or for some other reason a call should go in some special order.

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To set this feature, you need to set the type of priority in LCR:

You can go to Providers and click on Change Order to change the order of Providers for this

LCR:

Now you can Drag&Drop Providers to change their order:

Video at

[http://wiki.kolmisoft.com/index.php/Change_Provider_order_by_Drag%26Drop_video]

demonstrates how to do that.

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Click Back when you are finished. Assign this LCR to users and route calls manually.

Call routing per destination basis

Some destinations needs fine tuning to better tell which providers should be used to reach

that destination and in which order.

Each user has an assigned LCR. For some destinations, it is possible to set a different LCR.

This logic is demonstrated in the following example.

Example

We take Paraguay as example, because this country has a small number of destinations.

Let's say we have three providers (in LCR: Primary), and we use them all together for all our

destinations. The route is picked by the least cost route, that is, the LCR order is by Price:

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Then we have the situation that calls to Paraguay (except those to Mobiles) do not go

through the provider Eutelia.

To handle this situation we create a new LCR: Secondary, in which the order is Priority. It

excludes Eutalia and has only two providers:

Now we need to configure Routing per destination.

Click on the By destinations icon for Primary LCR (because Primary LCR is used by the user

who dials):

We create the following configuration:

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Explanation

Now when the user dials any Paraguay FIX number, the call will be routed using LCR:

Secondary with order: priority:

We can click on Active destinations and will see for which prefixes such an LCR applies:

As we know that Eutalia can route calls to Mobile networks of Paraguay, so MOB destinations

will be handled by old LCR:

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We can click on Active destinations and will see for which prefixes such an LCR applies:

How to route call to specific Destination through specific

Provider

Let's say you want to route calls to one specific destination through one (or several) specific

Providers.

As an example we will take a destination with the prefix 370630.

1. Create new LCR with only 1 specific Provider in it.

2. Use Call routing per destination basis to assign this LCR to destination 370630.

That's it - very easy.

Now calls to this destination will be routed ONLY through this Provider.

You can add more Providers to your LCR and use it with many Destinations.

You can create many LCR/Providers/Destinations combinations and use them in any way you

need.

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LCR/Tariff change based on call prefix

Also called: Tech Prefix functionality.

The main purpose of this function is for calls to be routed through a provider or providers

based on call prefix.

You have to create a provider, tariff, LCR (where the specified provider has to be), and a user

with his device. And most important, the localization rule or rules. It should be applied per

device basis (Device settings -> Location), making sure the specified device has the correct

rule applied. It must be noted that specified rules have higher priority than global rules.

1) Create a new tariff, provider, LCR, user and device.

2) Assign the newly made provider to the new LCR.

3) Create a new localization rule, named for example "test1".

4) Assign the device to location "test1".

5) Now edit the "test1" rule. It should be done las with normal localization rules, but

when you are creating a localization rule based on tariff or LCR, assign the specified

Tariff or LCR to correct place.

You can find more info on how to make rules in Number Manipulation.

Example

Lets say we have a provider that provides three quality routes: bad-cheap/normal/good-

expensive.

The bad-cheap route is selected when dialing an E.164 number with prefix 00.

The normal route is dialed when dialing the normal E.164 number.

The good-expensive route is selected when dialing the E.164 number with the prefix

000.

To accomplish this, we need to:

Create three providers with the same connection settings:

o Bad_Provider

o Normal_Provider

o Good_Provider

Create three Tariffs for providers:

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o Provider_Tariff_Bad

o Provider_Tariff_Normal

o Provider_Tariff_Good

Enter/Import appropriate rates to these Tariffs.

Assign these Tariffs to providers.

Create three LCRs and assign appropriate providers:

o Bad_LCR - assign Bad_Provider here

o Normal_LCR - assign Normal_Provider here

o Good_LCR - assign Good_Provider here

Create three Tariffs for users:

o User_Tariff_Bad

o User_Tariff_Normal

o User_Tariff_Good

Enter/Import appropriate rates to these Tariffs.

Create Location BadNormalGood with the following Rules:

Assign User_Normal_Tariff and Normal_LCR to User.

Assign all the user's devices to Location BadNormalGood.

That's it.

Now when the user dials E.164 numbers, he will be using the normal route; when 00, the bad

route; and when 000, the good route.

Routing by Percent

This type of routing selects a route by percentage value of Providers.

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It is similar to Priority-based routing, except that here Providers are selected randomly with

greater or lesser probability.

Example 1

For example, we have three providers with the following preferences:

A - 50%

B - 30%

C - 20%

This means that:

Provider A will be chosen FIRST with 50% probability.

Provider B will be chosen FIRST with 30% probability.

Provider C will be chosen FIRST with 20% probability.

MOR orders these providers using the following algorithm:

As a first step, it randomly selects the first provider. There is a 50% chance that Provider A will

be selected, 30% for B, and 20% for C.

Lets say the randomly selected Provider is B.

At the second step, MOR selects from the remaining providers:

A - 50%

C - 20%

There is a much greater chance that A will be randomly selected than C. So let's say A is

selected.

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The final order of our providers will be:

B - A - C

Every time, the order is RANDOMLY done in the same way, using percent values for

probability.

So we can get various results, such as A-B-C, A-C-B, B-A-C, C-A-B etc. BUT, following

mathematical reasoning, when we make a huge amount of tries (going till infinity), sets will

follow the rule in the first step: the Provider selected first will be A in 50% of the cases, B in

30%, and C in 20%.

Example 2

To better illustrate the method, let's take a simpler situation. We have two providers with

these percentages:

A - 99%

B - 1%

Provider A will be selected first 99% of the time, so that when we have many ordered lists,

they will look like:

A-B, A-B, A-B, A-B, A-B, ........., B-A, A-B,........A-B, ..........

The probability that the provider order will be B-A is just 1%.

Fail-over

This LCR ordering still keeps Fail-Over intact.

That is, after we have ordered the provider list, if the first provider in the list fails, the second

will be dialed and so on.

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Why call was not routed through other provider/trunk?

If you see that a call was not routed through the second (third/fourth/...) Provider/Trunk, take

the following steps:

1. Check Call Tracing – it is possible that other Providers can't handle the call (no rates, not

configured properly, not assigned to LCR, etc).

2. Check STATISTICS –> Calls –> Last Calls – check Hangup Cause for your failed call. If the

Hangup Cause is NO ANSWER or BUSY, then MOR does not route the call to the next

provider. More details in NO ANSWER/BUSY interpretation for providers.

Intelligent internal routing

It is possible to connect unlimited amount of MOR servers into the Cloud.

And MOR is capable of routing calls over all it's servers internally.

In provided example we have only 3 servers in the Cloud but it is enough for demonstration.

MOR can route calls between:

Devices (over DIDs)

Trunks (as PBX'es connected to the MOR)

Providers (which helps to connect to the PSTN)

In provided example all devices (phones), e.g. 1, 1-2, 1-3, 2, 2-1, 2-3, 3, 3-2, 3-3 can talk

between each other.

It is important to remember, that:

Device need to be assigned to DIDs in order to be reachable from Devices on different

servers

o As example in order for Phone1 to call to Phone2, Phone2 should be reachable

over DID, not over local Extension!

Each Device can be assigned to only one server

o And this Device should Register to or be accessible from correct server

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Each Provider can be assigned to many servers

o Shortest path to use (reach) this Provider will be used by Internal Routing

See also

Multi Server support

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DIDs

DID Management

Description

What is a DID?

In short – it is an external number to have incoming calls to your VoIP network reach a

particular device. (This is a very simplified explanation that will serve for the moment. For a

more extensive explanation, please check here [http://www.voip-info.org/wiki/view/DID] ).

VERY IMPORTANT RULE: DIDs in MOR MUST be in E.164 format.

You can reach the DIDs window in SETTINGS –> Billing –> DIDs:

Since MOR 12 you can choose different tone zones. Tone zone is a different ring tone

depending on destination (country).

Here you can see all DIDs and their info.

User/Dial Plan and Device show who will answer when dialing this DID.

Clicking on User or Device brings you to their DID settings. If you click on the icon, you will

go to the device's Call Flow window.

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Status

Status is very important. For every just-created DID, the status is

Free.

A DID can have several statuses:

Free – this status means that the DID is free to be reserved

for any user/device. You can also Terminate it.

Reserved – the DID is reserved but NOT actually used for a

user. "Reserved" is useful at the negotiation stage before

signing a contract. When a user decides to sign up, we

Activate the DID. We can also make the DID Free.

Active – the DID is assigned to a Device and is functional. We

can Close it.

Closed – the DID is no longer functional, so that all calls

going to this DID will get Hangup. One important feature of

this state: when a DID is Closed, it will be closed for a set time (the default is 90 days)

in case the user changes his mind and decides to use this DID again. This is useful

when a user is not paying for services, as we can suspend the DID by using Closed.

MOR will make it Free after the Closed period runs out. You can manually make it Free

or Terminate this DID.

Terminated – the DID is no longer used in MOR. It remains only for historical

purposes. You can make it Free if you want.

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Settings

If you want to edit a DID's settings, click on the icon:

The above window is for Free DID. You can:

1. Change Provider and language for this DID. "Language" is for voice prompts in the

Asterisk Dial Plan.

2. Reserve a DID for a user.

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3. Assign to Dial Plan.

4. Terminate a DID.

When a DID is assigned to a Device (Reserve for User first), you have a few different choices:

Here you can:

1. Change provider/language.

2. Assign DID to a different device (for the same user).

3. Assign DID to Trunk.

4. Stop a DID subscription and reserve the DID for 90 days (by default) to this user in

case he changes his mind and wants to get the DID back.

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When DID is assigned to Dial Plan, the choices are almost identical.

DIDs grace time

If call duration is less than Grace Time, it will not be accounted, for example: Grace Time = 3s,

call duration <= 3s, then price for the call = 0

See also

Provider Grace Time

Device Grace Time

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Timeout Digit

The maximum amount of time permitted between digits when the user is typing in a number.

When this timeout expires after the user has started to type in a number, the number will be

considered complete, and will be interpreted (sent to the system). Works both with entering

destination and ANI/PIN.

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Timeout Response

The maximum amount of time permitted for user to START entering a number. If the user

does not begin typing a number in this amount of time, system will notify that nothing was

entered. Once the user begins to type a number, system will wait for Digit Timeout to be

reached, and Response Timeout has no effect no more.

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CallerID Name Prefix

By setting this option you can add a CallerID Name prefix to the incoming CallerID.

Tariff for Calling Cards

Tariff for Calling Card lets you set a tariff for DID which will be applied for a calling card. It will

not be important what tariff is assigned for that card's group.

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Work days and Free days

It is available to set different DID rates for work days and free days. If you want to do this go

to SETTINGS -> Billing -> DIDs. Click on rates in the window you will see and then Split to

Work Days and Free Days. To combine back click Combine to Work Days and Free Days:

Add new DID(s)

In the main DIDs window, if you want to add new DID(s), click on New DID:

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In this window, you can add one DID or series of DIDs. Also, you should select the provider

which sold you this DID.

Delete DID

At first, mark it as "not used" (Free). Then terminate it. When you do, you will see next to

the DID in the main DID window. By clicking on it, you can delete the DID.

Edit a DID's number

It is not possible to edit a DID number. You can only delete an old one and create a new one.

DID bulk management

In the main DIDs window, click Bulk Management and you will be able to perform various

actions with a series of DIDs:

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Assign DID to Trunk

What is trunk?

"Trunk" refers to another server/PBX that can be connected to the MOR system. For more

information please visit Trunks page .

How to assign DIDs to trunk?

1. Go to SETTINGS -> Billing -> DIDs:

2. Select Bulk management:

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3. Enter DID interval you want to assign to trunk and select the marked option:

4. Select a user you want DIDs to be reserved for:

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5. Select a trunk you want your DIDs to be assigned to:

6. Completed - your DIDs are assigned

See also

Trunks

DID management

Work flow for DID

Technical explanation:

1. Let's say a call comes through a certain Provider. The connection type is not important.

So it could be PSTN/SIP/IAX2, or whatever.

2. Call -> Provider -> PSTN/SIP/IAX2 -> your Asterisk.

3. The call hits Dialplan [mor] exten => _X.,1, MOR:

4. Call -> Provider -> PSTN/SIP/IAX2 -> your Asterisk -> MOR.

5. MOR checks its database (DB) for the incoming DID. If it's found, MOR sees where to

route the call. In the MOR GUI, you can select to route the DID to a special Dial Plan

(CallBack, Calling Cards, Authorization by PIN/ANI) or to some local device.

6. The local device in its Call Flow can forward the call to:

1. VoiceMail.

2. Another Local Device.

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3. An External Number.

7. So summed up, MOR can forward a DID to:

1. Dial Plan (CallBack, Calling Cards, Authorization by PIN, IVR).

2. A Local Device.

3. VoiceMail.

4. An External Number.

8. Call -> Provider -> PSTN/SIP/IAX2 -> DID -> Your Asterisk -> MOR -> (Dial Plan

or Local Device, Voicemail, External Number) -> ...

DID Billing

Description

DIDs can be billed. DID billing is quite complex, as they have three types of rates. That means

three prices are counted for one call. DID rates can be reached in the main DIDs window by

clicking the Rates icon for the selected DID:

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Incoming rate

For incoming calls, when the Incoming rate is greater than zero, the rate is paid to the system

owner by the user who is dialing to that DID. It represents a profit for the system owner.

If the rate is less than zero, the price is paid TO the dialing user for the DID usage.

The dialing user is the user in MOR, and this rate applies only then when MOR recognizes the

dialing user (by PIN/ANI).

Owner rate

For incoming calls, when the Owner rate is greater than zero, this rate is paid to the MOR

system owner by the DID owner. This represents a profit for the system owner.

If the rate is less than zero, this price is paid TO the DID owner.

This rate is charged even if the local user dials another local user using DID with Owner Rate

set.

DID usage is not allowed if this rate is > 0 and the DID owner does not have a large enough

balance to cover call costs.

DID owner rate is not applied to the DID's owner - that is, the owner of the DID will be not

charged when dialing his own DID.

Who is a DID's owner?

When a DID is assigned to User/Device, the owner is the User:

If a DID is assigned to a Dial Plan, then its Owner can be selected in the DIDs settings and any

user can be considered as the 'owner' of the DID and can be billed at this rate when anybody

dials this DID.

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Select "Device" to see calls to this DID near this Device's calls:

Provider rate

For incoming calls, when the Provider rate is greater than zero, this rate is paid to the system

owner by the Provider. It represents a profit for the system owner.

If the rate is less than zero, this price is paid TO the Provider for the DID usage.

See also

DID Billing Time explained

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DID Billing Time explained

Which time in DID is billed?

Let's imagine we have a call consisting of the following parts:

Here:

A is when a person dials DID number.

B is when a call arrives to MOR

C is when the dialed destination is sent to the provider.

D is when the calling party answers the call

E is when one of the parties hangs up and the call is ended.

So:

DID billable time is B - E.

Normal call billable time is D - E.

C - E is Call Duration.

D - E is Call Billsec.

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DIDs Usage

STATISTICS -> Various -> DIDs -> Usage

This page shows current DIDs status and Actions with DIDs in a selected period:

Personal DIDs

PERSONAL MENU –> Various –> DIDs

This page shows DIDs which are associated with a User. No changes to configuration can be

made in his window.

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Example - Configuring DID to ring some Device

Primary situation

We have:

DID - an external number (02180995165) provided by a Provider.

The number is from the UK.

The connection to the provider is via IAX2, IP: 23.45.67.89:4569.

Devices SIP/201 and SIP/202.

We want:

To dial the DID 02180995165 using our mobile phone and ring Device SIP/201.

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To dial the DID 02180995165 using the local Device SIP/202 in order to ring Device

SIP/201 (local SIP-SIP call not going to PSTN).

Configuration

Plan

1. Configure Localization for Provider.

2. Create Provider.

3. Register Provider.

4. Create Devices SIP/201 and SIP/202.

5. Create/Configure DID.

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Configure Localization for Provider

As the DID is not in E.164 format, we must configure MOR to handle this.

Since the DID 02180995165 does not come from the provider in E.164 format, we must create

a Location for this provider and add a rule to it. We create Location: UK and add a rule to

convert the non-E1.64 DID to E.164 DID: 02180995165 -> 4402180995165.

Create Provider

Create Provider in MOR GUI.

Do not forget to set Location: UK for this Provider.

Register Provider

IF your Provider asks you to register it, do so.

Create Devices SIP/201 and SIP/202

Create Devices as usual (there is no need to edit any .conf files!)

DO NOT forget to set Location: UK for these Devices.

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Create/Configure DID

Create DID in E164 format

Reserve this DID to your User

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Assign DID to Device SIP/201

That's it.

DID with VoxBone using his URI

Question:

I have DIDs with Voxbone. They send the URI using [email protected].

It's okay for receiving calls, but I need to know how I can use this system to redirect to a

Calling Card Dial Plan.

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Answer:

When you configure your Calling Card Dial Plan (DP), you should assign it to a DID.

Let's say your Calling Card DP DID is 12345. Configure your Voxbone DID to call to:

[email protected] where sip.server is your server's IP or hostname.

NOTE: 12345 is an EXAMPLE! DO NOT CREATE 12345 as your DID - create a real one!

Make sure you have Voxbone configured properly in your system.

Forward DID to External Number

Create a User/Device.

In the Device's Call Flow, in the Before Call section, select Forward.

Enter the External Number to which you want to forward the call (make sure the

number is in E.164 format).

Assign your DID to this User/Device.

Now when a call comes to the DID, it will be forwarded to the External Number.

Video example how to forward DID to external number can be found here:

[http://wiki.kolmisoft.com/index.php/Forward_DID_to_External_Number]

See also

Purchase DID at didx.net and forward DID to an external number

How to charge DID on a monthly basis

Assign DID to the User's Device.

Create a Service with a name, for example: "DID price for 1 month".

Allocate a price for it.

Create a Subscription for the User of this service.

This subscription will be visible in the User's Invoice.

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Video example how to charge DID on a monthly basis can be found here:

[http://wiki.kolmisoft.com/index.php/How_to_charge_DID_on_a_monthly_basis]

DID handling by Localization

When a call comes in to your system from the Provider, it should be localized to E.164 format

for DID search. In other words, as DIDs are in E.164 format, whatever comes from the provider

should be localized to E.164 to find the correct DID.

To do this, each Provider also has a Location. This means that Location rules can be created

and applied for the Provider. You can do this in the Provider settings window:

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For example, you have DID 37052058393 and the provider sends you 0037052058393. You

need to strip this international prefix with the following rule:

Don't forget to apply the correct Location to the Provider!

Special case: In some countries when E1 is connected to the server, only the four last digits of

the number reach MOR. Localization should be used here to add the remaining part of the

number.

See also

Number Manipulation

o Localization

o Provider Rules

o Number Manipulation Examples

DID Blocking

If a customer is not paying or you want to block calls to his DIDs for any other reason, go to

the DID window and Close DIDs which belong to this customer.

"Blocking" in the User EDIT menu will not block his DIDs. It will only block dialing, so that the

user will not be able to dial out.

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See also

DID Management

Work flow for DID

DID Billing

DIDs Usage

Example - Configuring DID to ring some Device

DID with VoxBone using his URI

Forward DID to External Number

How to charge DID on a monthly basis

DID handling by Localization

Dial Plans

Dial Plan is the name for a specific functionality in MOR. Currently it has the following types:

Authorization by PIN

Authorization by ANI

Calling Card Dial Plan

Callback Dial Plan

IVR Dial Plan

Authorization by ANI + CallCard on one DID

You can find all dial plans in SETTINGS –> Billing –> DIDs –> Dial Plans:

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Authorization by PIN

Description

This function is for customers who can dial an access number and enter their device PIN

number in order to get a dial tone and enter a destination number. It is used in some

countries to get cheaper calls abroad.

The algorithm for the Authorization by PIN dial plan is seen in the following image:

The main steps to configure this functionality are:

1. Create an appropriate Dial Plan.

2. Assign an DID to this Dial Plan.

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First, in the main Dial Plan window, click on New Dial Plan and enter values to create an

Authorization by PIN dial plan:

IMPORTANT! Check the Use ANI? field – otherwise it will create a different type of Dial Plan.

Ask Number/PIN (times) – how many times the user can try to enter the PIN.

Ask destination (times) – how many times the user can try to enter the destination.

Click on to create this dial plan. The first step is complete – we have created our dial plan:

Now we need to assign a DID to this Dial Plan. To do this, in the DIDs edit window we assign

the selected DID to the Dial Plan:

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In the main DIDs window we see that this DID is now pointing to our dial plan:

That's it. Very simple. Now, dialing 37063042439, we reach our Dial Plan. Here we hear a

message that asks us to enter our device PIN. If the PIN is correct, we can dial the desired

destination.

Save original CallerID into CallerID Name field

Sometimes the CallerID of the original Caller is changed by the CallerID set in MOR. To see

the real CallerID in CDR, it can be saved in the CallerID Name field.

Enable this setting if you want to save it:

See also

Dial Plans

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Authorization by ANI

Authorization by Automatic Number Identification (ANI) is a variant of the Auth. by PIN

method that makes the system act in a different way. If you check the Use ANI? field in the

Dial Plan creation window, the user's CallerID will be checked automatically when he dials in,

and he will only be asked to enter his PIN if his CallerID is not found.

CallerIDs are entered for every device in Device Settings using the icon.

ANI - Automatic Number Identification

ANI is used to authenticate who is calling, by CallerID.

When a call comes into MOR, the CallerID is checked with all available CLI's in the database. If

a match is found, the call is authorized and assigned to the device which CLI has matched

against the CallerID of the call:

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All other details are the same as for Authorization by PIN dial plan, and the dial plan is

created in the same way. Just check Use ANI?:

In main Dial Plan window, we have:

Note the difference between Auth. by PIN and Auth. by ANI dial plans.

Assigning a DID is similar to Auth. by PIN method.

ATTENTION! Be careful because it is very easy to fake a CallerID.

Default device

This option is available when Use ANI option is checked.

Use this feature with great understanding what it is used for. It allows all calls pass over

this Dial Plan and assigns such calls to a selected Device.

This feature can be used if you want to offer any services for unauthenticated clients. Owner

of device is charged for these calls.

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End IVR

When making an authorization by PIN dial plan you must set and end IVR to it. You can

change the IVR by clicking Edit.

Troubleshooting

If the system asks for a PIN to be entered

This occurs if the CallerID comes in a different format from that which is entered in the Device

CID. Use Last Calls to check it and Number Manipulation to fix it.

See also

Dial Plans

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Calling Card Dial Plan

Overview

Once we have generated a Calling Card Group and Cards, we can create a Dial Plan for this

Group.

A Dial Plan can be treated as a set of rules for how to handle a particular Calling Card Group.

You can set different rules for the same Calling Card Group.

The Dial Plan gives you more control over Calling Card Groups.

Dial Plan creation

Let's go to SETTINGS –> Billing –> DIDs –> Dial Plans:

In this image, we see an existing Dial Plan for a Calling Card Group.

Let's review each field:

Name – just for informational purposes.

Number length – tells Interactive Voice Response (IVR) how many digits to wait for in

NUMBER following a request to enter a number.

PIN length – how many digits to wait for in a PIN.

Ask only PIN – as the PIN is unique, it is sometimes wise to ask only for the PIN,

because users can make mistakes and get annoyed about having to enter too many

digits.

Tell time – should IVR tell the time after the Destination is entered?

Tell Balance before Call – should IVR tell the balance before the user enters the

NUMBER/PIN?

Tell Balance after Call – should IVR tell the balance after the user enters the

NUMBER/PIN?

Ask Number/PIN (times) – how many times the user can try to enter Number/PIN.

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Ask destination (times) – how many times the user can try to enter Destination.

Assigned DID(s) – just for informational purposes, to see which DID(s) are assigned to

this Dial Plan.

If Card balance is lower than - self explanatory.

Balance IVR - a Balance IVR will play.

When creating a new Dial Plan, you will need to enter the correct Number and PIN length

values of an existing Calling Card Group. Otherwise you will not be able to create the Dial

Plan.

Assign DID to a Calling Card Group Dial Plan

When we have created a Dial Plan for a Calling Card Group, we can assign one or more DIDs

to this Dial Plan.

A DID (in other words, an Access Number) is vital for using Calling Cards. The user will use

this number to reach IVR.

You can assign unlimited DIDs to the same Dial Plan, but a particular DID can only be

assigned to one Dial Plan.

The Calling Card Dial Plan is useful when you are selling Calling Cards in different regions,

such as different countries. In country A, the user will dial one number (which is cheaper for

him or even free), and in country B the user will dial another number. They will reach the

same IVR and will use the same service, but they will save on telephone fees.

In SETTINGS –> Billing –> DIDs:

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We click on the icon for the first DID:

We select our Dial Plan and click the confirmation icon to assign our Dial Plan to this DID:

Now the user can dial number 12026911622 (in the example) and he will be able to reach IVR

and use his Calling Card.

Callback Dial Plan

Example

An example can be seen in action in Callback setup example.

Options

Ask for Card data after callback

This option only applies to Calling Card Callback. You can ignore it if your Callback Dial Plan is

not for Calling Cards.

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Mark it to ask for Calling Card Number/PIN only after system calls back to the caller who

initiated callback.

By default, the system asks for the caller to enter a Number/PIN at the point when he tries to

initiate the callback.

CallerID for callback

This feature provides CallerID management for callback. The following options are available:

Enter custom CallerID for callback.

Leave field empty to send empty CallerID.

Enter * (star) which means that CallerID will be the original CallerID of the caller who

initiated Callback (the same functionality as prior to MOR9).

This function is useful when Providers check Destination and CallerID and block the call when

they match. For example, without this function it is not possible to use Callback in the Telus

Mobility network in USA/Canada.

Original CallerID for callback

Starting from MOR 10 you can enable setting to use an original CallerID. To do this:

1. Create a CallBack Dial Plan

2. Go to CallBack Dial Plan edit

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3. Enter * sign in a marked field:

IVR Dial Plan

The IVR (Interactive Voice Recognition) Dial Plan can be seen in action in IVR for Calling

Cards.

Please note that the above example only illustrates how to create and IVR Dial Plan and it

does not mean that IVR Dial Plan is only applicable for Calling Cards. Using IVR you can

program automatic answering machine to redirect the caller to an appropriate departament,

for example. For better understanding of IVR logic see IVR system.

STATISTICS -> Various –> DIDs

STATISTICS -> Various –> DIDs is showing various parameters of DIDs using statistics.

DIDs - DID number which information is showing.

Date - Here is showed date from which DID is active.

Provider - provider which are used for calls.

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Calls - number of calls which was made.

Incoming price - price of incoming call.

Owner price - Price which user pays.

Provider price - Provider rate.

Profit - amount expressed as a value that is left over from sales once all expenses have been

deducted.

Accounting

Accounting in MOR is implemented to keep track of all payment activity. It is also possible to

create various services, make subscriptions for them, and generate invoices for the clients.

Services

A Service is a product that can be sold to the client. There are various types of services. For

example:

Monthly fee.

Line rent.

Phone rent.

Internet plan.

Example:

You can create an unlimited number of Services. All of them can be found in SETTINGS –>

Accounting –> Services:

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Service types

Periodic fee per month - user is charged every month for this service.

One time fee - user is charged only once for this service.

Flat-Rates - user is charged every month for this service and can dial a certain number

of minutes to certain destinations.

Selfcost

It allows you to include all your expenses to track the real profit correctly.

Example:

You have a subscription to charge your clients for internet connection by 10 EUR, but it cost 6

EUR for you. Client pays 10 EUR to you, but real profit is 4 EUR.

Selfcost functionality allows you to see a real profit in statistics.

Subscriptions

Users can subscribe for some services. When they do, they get a service Subscription. Service

Subscriptions can be found for each user in the Users main window by clicking the icon for

the selected user:

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Subscriptions can have a Memo (if you sell a DID, for example, you can make a note of its

number here). Subscriptions also has the important fields Activation start and Activation

end. This data interval will be billed in the invoice, but dates before and after this interval will

not be billed.

You can add/edit/delete subscriptions for the user.

It is not recommended to delete subscriptions, or the user will not be billed for that

subscription. If the user stops using a service, set the Activation end value to the date when

he does so.

NOTE: A service is billed from Activation Start till Activation End, and from the start till the

end of the invoice period. The time period that fits within both the previous periods is the

actual period that will be billed.

Every month a subscription about all subscriptions made in the system during the month is

sent to admin. Also resellers' subscriptions are also sent to admin.

Example of email with subscriptions

Charging for subscriptions.

Date: 2012-5

1 User: Test User #1(101):

Service: Test_periodic_service - 10.0000

Service: Test_periodic_service - 10.0000

2 User: User Resellers(user_reseller):

Service: res_1 - 110.0000

Subscription calculation

Subscriptions are calculated by the following rules:

When a Subscription is assigned to a Prepaid user, the price for the first month is

deducted from the user's balance instantly (and an appropriate Payment with type =

subscription is created).

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*If the Prepaid user does not have enough balance to cover this subscription, the

subscription fails.

When a Subscription is assigned to a Postpaid user, no changes are made to his

balance instantly.

Every month day 1, 00:10:00, an action is executed which performs the following calculations

(blocked users are skipped).

Prepaid users

For Prepaid users, the balance subtracts the Subscription price for this new month.

A New Payment is created, with type = 'Subscription', to mark the user's balance deduction

(only for Prepaid users).

If a Prepaid user's balance becomes negative, this user is blocked and receives a notification

email (as does his owner).

Allow prepaid users balance to drop below zero

SETTINGS -> Setup -> Settings -> Various -> Do not allow prepaid user balance drop

below zero

If this option is not enabled - then MOR allows user balance to drop below zero. If this

option is enabled - user is blocked when there is not enough money for a subscription - both

user and admin are informed about this by email.

Postpaid users

For Postpaid users, the balance subtracts the Subscription price for the previous month.

If the Postpaid user's balance + credit < 0, he is blocked and receives a notification email (as

does his owner).

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Subscriptions scheme

Account type Prepaid or Postpaid can be changed in user details

Credit is an amount of money which user is allowed to owe you.

Example 1: user balance is -4.00 USD, credit = 5.00 USD. Balance + credit = 1 USD, so user

will still be able to call.

Example 2: user balance is -4.00 USD, credit = 4.00 USD. Balance + credit = 0 USD, so user

will not be able call.

Credit can be changed in user details

Balance is an amount of money user already paid you, but have not been spent yet on

services you provide (can make calls using this money). Current user balance can be

checked in user details

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Allow prepaid users balance to drop below zero

Example

A user is postpaid.

On the first day of his first month

User balance is: -$75.

-$75 is for calls. Also, the user has to pay $475 for subscriptions.

The system calculates the subscription price and deducts $475 from his balance.

The user balance becomes -$550.

The user gets an invoice for $550.

He makes a payment of $500.

The user's balance becomes -$50.

On the first day of the next month

In the next month, the user gets an invoice for calls made over that month + subscription

prices in that month.

-$50 will be shown on the invoice, but will not be included in the new invoice.

Invoices

Configuration

All settings for Invoices can be set in the Invoices section of the Configuration window.

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Invoices can be viewed and created in SETTINGS –> Accounting –> Invoices. In the main

window, you can see all the invoices:

The icons and show whether this invoice is Paid or Unpaid.

Click on the icon to get details for the invoice:

Here you can view the details of all invoices and click on the or icons to generate invoice

files in PDF or CSV formats accordingly.

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Invoice generation

In the main Invoices window, click Generate Invoices to generate additional invoices:

Select for which users to generate invoices. Prepaid and Postpaid users can have separate

details. Also, you can check to generate an invoice for only one particular user.

Invoice generation when total call price is zero

Sometimes it is needed to generate invoice when total call price is 0 anyway. By default this

possibility is disabled if you want to enable it go to SETTINGS -> Setup -> Settings ->

Invoices tab, select Calculate 0 price Calls into Invoices and click Save changes.

Note: Issue date option is available starting from MOR 10

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Detailed invoice settings

To access detailed invoice settings go to: SETTINGS -> Setup -> Settings. Click on tabs:

Invoices -> Postpaid or Prepaid -> Details. You will see this window:

Fill in the fields with wanted information.

In MOR 12 there are new additional options: Add Average rate in detailed invoice (shows total

price/total minutes) and Show Time in Minutes (shows time in mm:ss instead of hh:mm:ss, for

example, not 02:13:42 but 133:42).

Invoice recalculation

An invoice cannot be modified after being sent. But if you want to allow recalculations after

sending invoices, go to: Settings -> Setup -> Settings. Then click on the tabs: Invoices ->

Functionality.

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Select Invoice allow recalculate after send:

To recalculate invoices in the main Invoices window click Recalculate invoices:

Select for which users to recalculate invoices.

Send invoices by email

Click to open more options to enter the period in which you want to send invoices. Click

Send to send invoices to their Users in this period:

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PDF invoices

The layout of PDF invoices is not customizable.

Logos are not supported in PDF invoices.

You can print your invoices on paper with a pre-printed logo if you want to send paper

invoices.

Invoice comment

Sometimes it is useful to add a comment to the invoice:

Recalculate a single invoice

If an invoice is not marked as paid - it can be recalculated for the same user, same period. To

achieve this - press the marked icon:

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Invoice list to csv

You can export invoice list to CSV:

Invoice language

Default language in MOR is English. However, you can send invoices in different languages

(click on the flag icon on the top of the page).

See also

Accounting

Invoice configuration

Pay Invoice with balance deduction

Invoice by CallerID

MOR API invoices

Cannot generate Invoice

Invoice configuration

In User's settings it is possible to configure some invoice settings for this user:

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Generate Invoice

This option marks whether an Invoice for this user should be generated.

PDF/CSV Options

Here, checks decide what kind of invoices (as attachments) should be sent to users by email.

See also

Configuration from GUI -> Invoices

Pay Invoice with balance deduction

When an invoice for a user is generated, and the user pays it, we need to check it as

completed. After this is done, the user's balance is increased by the value of invoice.

Let's see our user's balance (SETTINGS - Users):

Then in invoices (SETTINGS - Accounting - Invoices), we see an invoice for this user:

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Click Details to see the invoice details. Check Create payment for completed invoice and click

the red cross icon above to mark the invoice as completed and paid:

You will see the invoice as paid:

In user details, we see that the user's balance has decreased:

... and in Payments (SETTINGS - Accounting - Payments), we see that payment for invoice is

created:

Other notes

Note: if you want to revert all changes, it is enough to click the green check mark in Invoice

details. The user balance will be decreased, payment deleted, and the invoice marked as not

processed.

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Invoice by CallerID This type of invoice groups all calls by CallerID.

Only outgoing calls are included in this invoice.

It looks like this:

Client number: 37052058393

Number Date Duration Rate Price Destination

37063042439 2007-12-01 16:21:03 00:00:20 0.7597 0.2532 Lithuania

...

37068723423 2007-12-14 12:50:17 00:00:22 0.7597 0.2785 Lithuania

Lithuania 150.4886 LTL (Without TAX)

Total: 150.4886 LTL (Without TAX)

Client number: 37063042438

Number Date Duration Rate Price Destination

37068723423 2007-12-01 14:45:14 00:03:20 0.7597 2.5322 Lithuania

...

37065017476 2007-12-13 21:31:14 00:04:11 0.7597 3.1779 Lithuania

Lithuania 26.9805 LTL (Without TAX)

Total: 26.9805 LTL (Without TAX)

Monthly fee - test memo2 10.0000 LTL (Without TAX)

TOTAL TO PAY: 187.4692 LTL

Invoice by CallerID CSV

Starting from MOR 9 you can export invoices by CallerID into CSV:

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MOR API invoices

MOR API

Retrieves a list of invoices in the selected time period.

Request

If a user has 'admin' rights, he will get ALL invoices from the selected period. If user has 'user'

rights, he will get only HIS invoices from the selected period.

HTML POST:

http://<SERVER

IP>/billing/api/invoices?u=username&p=password&from=1188604800&till=1191196799&la

ng=en

Here:

from/till - time period for invoices in Unix Epoch Time format. Invoice's period start

should be >= from and period end <= till.

o http://www.epochconverter.com/ can help to convert time for testing.

lang - choose the language in which you want to get invoice details (mainly 'Calls').

Example result

<Invoices from="2007-09-01" till="2007-10-01">

<Invoice agreementnumber="0000000000" user_id="0" clientid="240887380">

<Product>

<Name>Monthly fee - test</Name>

<Quantity></Quantity>

<Price>10.0</Price>

<Date_added>2012.02.02</Date_added>

<Issue_date>2012.02.02</Issue_date>

</Product>

</Invoice>

<Invoice agreementnumber="0000000000" user_id="0" clientid="240887380">

<Product>

<Name>Calls</Name>

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<Quantity>65</Quantity>

<Price>112.3324</Price>

<Issue_date>2010.01.01</Issue_date>

</Product>

</Invoice>

</Invoices>

Errors

<Error>user not found</Error>

<Error>no invoices found</Error>

Credit notes

About

We can say that Credit notes functionality is opposite to invoices. If "paid invoice" means

that users balance decreased then "paid credit note" means that users balance increased.

Sometimes due to various inaccuracies user pays more than he should. That is why this

functionality useful - to correct those inaccuracies by paying the money back to the user.

Create Credit notes

To create a Credit note go to SETTINGS -> Accounting -> Credit notes:

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Click Create credit note. You will see this window:

Issue day - by default it is the day credit note was created. But you can choose different date.

User - select a user from dropdown for which you are making this credit card.

Number - same as with invoices, you can enter any symbol or symbols here.

Price - enter price without tax you want to transfer to user.

Comment - your comment.

Click Create.

A user whom you made and paid a credit note will see this in his Payments:

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Possible errors

"Incorrect hash" - self-explanatory

"Bad login" - self-explanatory

"Credit note was not found":

o editable credit note was not found

o credit note does not exist

o credit note belongs to the user you cannot see

"Credit note was not updated" - if you see this error possibly something in MOR

database went wrong

"Credit note was not created" - this error shows up if Issue day or Price fields are

empty.

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Payments

Definition

A Payment is created when a user pays the system owner for some Service:

Possible payment types are:

Card - when a card is bought.

Invoice - when an invoice is paid.

Paypal - when a payment is made using Paypal.

Webmoney - when a payment is made using WebMoney.

Linkedin - when a payment is made using Linkedin.

Cyberplat - when a payment is made using Cyberplat.

Voucher - when a Voucher is used.

Subscription - for Prepaid users when charged for a subscription.

Manual - a manually added payment.

Credit note - when a credit note is paid.

Initial balance - it's a user's primary balance.

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Manual payments

Sometimes customers send money through a bank or other means, or they might use some

online gateway not implemented in MOR. To handle such transactions, MOR has Manual

Payments.

When a user sends an amount to you as the owner, you need to enter this amount into MOR

using the Manual Payment window, which can be found in SETTINGS –> Accounting –>

Payments –> Add manual payment.

The first window lets you choose the user and enter the amount (with TAX) in the selected

currency:

Next, click Proceed to see the calculated values for this Payment:

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Click Confirm to enter this payment, which can be seen in the Payments window:

After payment is entered, the user's balance will increase and he will see the payment in his

Payment sections.

The Administrator can delete this payment by the user. After it is deleted, the user's account

will decrease accordingly.

Use negative payment to remove money from a user's balance.

Initial balance

Initial balance is the balance that was added to user during its creation. Initial balance is valid

for registered users (via GUI or API), users created by admin and users created by reseller.

Initial balance lets you see user's primary balance without CLI.

See also

Accounting

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Vouchers

What is vouchers?

Vouchers are used to fill the balance of a user. You can sell vouchers printed on paper, or sell

them over the Internet. When a user logs in and enters a voucher number, his balance will

increase by the value issued for this voucher.

Where I can find vouchers?

You can find vouchers here: SETTINGS –> Accounting –> Vouchers

All voucher configuration options are in the Vouchers section of the Configuration window.

You can view created vouchers in SETTINGS –> Accounting –> Vouchers:

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This window show the status of the voucher. Active? shows whether we can use the voucher.

All other information is self-explanatory.

In order to add new vouchers, click Add vouchers:

Here you can enter the necessary information to generate an array of vouchers.

The tag field is only used to mark a series of vouchers. It is just for informational purposes or

for easier search.

Click to make various changes to many Vouchers at once:

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After that remember to enable vouchers:

1. Go to SETTINGS –> Setup –> Settings –> Payments:

2. Select tab vouchers and ensure that option Vouchers enabled is enabled.

When a user logs in, he sees vouchers icon on the main page:

He can click it and enter the voucher number. He can try up to X times, where X is set in

configuration menu. If he fails to enter the correct voucher number X times (the maximum

number he is allowed), he will not be allowed to enter the voucher number again for a set

period of time. The values for how many times he can try and for how long to disable

attempts to enter are configurable in the Configuration menu.

When the user enters a correct voucher number, his balance will increase, he will get a new

payment record, and this voucher will be marked as used.

Vouchers are not accepted

Make sure Active Till is tomorrow or a more distant point in the future.

Vouchers are not visible for postpaid users

1. Go to SETTINGS –> Setup –> Settings

2. Go to "Visual" tab and check if Hide payment options for postpaid users option is set. If

so - unset it and vouchers will be visible.

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See also

Accounting

PBX Functions - Use Voucher

Flat-Rates

Description

Flat-Rates means a service plan under which a Client can dial a number of minutes to certain

Destinations for a fixed amount. After the user has used all his flat-rate minutes, all additional

minutes are charged at the normal rate.

Setup

1. Create a New Service with type: Flat-Rate.

2. Set a Price for this Service and set how many minutes the User can chat for at this

price.

3. Create the Service by clicking the Create button.

4. In the Services window near the just-created service, click the icon.

5. Add the Destinations to which this Service should be applied. Use Include/Exclude to

create an exact list of the prefixes you want to use with this Service.

6. Add Subscription to a user to use this Service.

Acknowledgments

Flat-Rates are calculated by seconds.

Every month, the minutes available under Flat-Rates plans are reset to the initial value,

which allows the client to use the same amount of minutes each new month. The

clients' unused minutes for the previous month are lost.

A Service is applicable for a whole month - that is, the service cannot start/end in the

middle in the month.

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IMPORTANT: If a user is PREPAID, limit his calls to 1, because otherwise a loss is

possible. (Explanation: calls will not be cut when Flat-Minutes will run out and

small loss is possible till all Flat-Rate based calls will finish. After that no more

calls will be possible).

Only Admin can create and manage Flat-Rates and assign to Resellers/Users.

o Resellers CAN NOT create/manage Flat-Rates nor assign them to their Users.

o Admin CAN NOT put Flat-Rates on Reseller's Users

Reporting

Active Calls

Description

In Quick Stats in the main window, it is possible to see Active Calls. These are the calls that are

in the system at the moment.

It is possible to click and open a new window with detailed Active Calls. You can get to this

window in STATISTICS –> Calls –> Active Calls.

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In this window all active calls are presented with more details, such as Start Date, Duration,

Src, Dst.

In order to see the correct duration on multi-server systems, the time should be synchronized

on all of the servers.

It is possible to reset statistic by clicking the "Reset active calls information" button. As the

text next to the information icon says, it will not affect your calls in any way.

NOTE: local calls are not visible on MOR GUI, only calls to/from outside.

Hangup Active Call

To hang up an active call, click the delete icon next to it.

The system will try to hang the call up.

If the call is still here, it means that Asterisk can't hang up the call. It has stuck. Stuck calls are

cleared every two hours. There's no need to do anything further.

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Active Calls for Users/Resellers

In order to enable this functionality for Users/Resellers, check the relevant setting (it is

disabled by default):

Users/Resellers will now be able to see an Active Calls selection in their Personal menus:

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Active Calls for Users/Resellers shows codec used for call

System requirements:

MOR version 11 or higher.

Asterisk 1.4.42.

Monitorings addon.

This functionality shows which codec is used and PDD for active calls in real time.

When call is in ringing stage when system shows information of Leg A.

When call is answered it shows information of codec used for LegA and LegB and call time.

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Show DID in Active Calls

To make a DID column appear in the Active Calls window go to SETTINGS -> Setup ->

Settings, click on the Functionality tab and look for Active Calls tab in the appeared window.

Select Show DIDs in Active Calls.

The result in Active Calls window:

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Last Calls

STATISTICS –> Calls –> Last Calls shows the most recent calls in the system. It is used in

Troubleshooting.

Please note that when occurs a call where DID is involved it consists of two legs (call from DID

to MOR and call from MOR to Destination). They will be shown seperately in Last Calls list.

Search

Starting from MOR 9 a search bar was added for Last Calls statistics.

Search is available both for users and administrators.

CSV

Using the "Export to CSV" icon you can export your currently displayed calls into CSV

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PDF

Using the "Export to PDF" icon you can export your currently displayed calls into PDF

Call information representation

When call is initiated by the caller, at least one call leg is created - from user agent

client(UAC) to user agent server(UAS) - usually called LEG A. Based on routing rules, call may

be canceled(in case there is no suficcient funds, no appropriate rule etc.) or one more call leg

may be created - from UAS to destination another UAC (LEG B). Many call legs may be

created depending on routing rules and whether call to destination was succesfull. For

instance:

1. User A dials user B

call leg from user A to PBX is created

PBX can route call to user B

call leg from PBX to user B is created

when user B answer these two legs are bridged

there we have two bridged legs and one call

2. User A dials user B.

call leg from user A to PBX is created

PBX cannot route call to destination and the call is canceled

there we have one failed call and one call leg (LEG A - OK, LEG B - FAILED)

3. User A dials user B

call leg from user A to PBX is created

PBX can route call to user B through provider C

C fails to route the call

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but PBX can route the call through another provider D

D succeeds in routing the call

user B answers

there we have three call legs, two of them were bridged and we would say that there

was only one call which was connected.

If user A is interested only in calls made, information, about how many attempts were made

to call user B, has no interest for him. Majority of users will be interested only in call duration,

who terminated the call etc.

Hence this is how call information is represented in call statistics, trying not to overwhelm

users with too much technical details, only information about successful call is presented,

ignoring details about made call legs (FAILED or NOT, etc). Provided you want to check the

information on FAILED and connected call LEGS, you can use the Call Info page.

See also

Troubleshooting

Reporting

Users Calls

This report shows all the calls a user has made in a selected period. It is possible to export this

list to PDF or CSV format.

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To find this report go to STATISTICS –> Calls and click on icon.

Users Finances

The Finances window lets you view a user's entire financial situation in one window. It can be

reached in STATISTICS –> Finances:

Loss Making Calls

Sometimes you may import or set rates that are lower for a provider than for the user. In such

cases, when the call is made, you are taking a loss.

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To find such calls, you should use this function. Just select the desired period and all loss

making calls will be visible.

You can find this report in STATISTICS –> Calls –> Loss making calls:

This window shows you loss making calls in the selected period. If you click on the user or

provider rate, you will go directly to the rate's edit window and be able to edit it.

Providers Statistics

You can reach this report in STATISTICS –> Various –> Providers. This report shows all

providers' statistics at once:

In the selected period, you can check various statistics about all the providers in your system.

If you click on a number in the Answered/No Answer/Busy/Failed columns, you will open a

window where these calls are listed.

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While making a prefix search, if you want to see all prefixes starting with a certain

combination, you should enter % at the end. For example, if you want to see all prefixes

starting with 37 you will have to enter 37%. If you enter only 37 you will only see this prefix.

Providers Calls

1. Go to SETTINGS –> Billing –> Providers:

2. In the main Providers window, click on the to open the Providers' Calls page:

Here you can see all calls for the selected provider in the selected time period. You can

change the type of the call.

You can also export the selected calls to PDF or CSV format.

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First Activity

This window shows the User's First Call, if the call was made in the selected period.

NOTE: It does not necessarily show the first call the user made! It only shows the first call of

the user in the selected period. The first call of the user may have been made outside the

selected period. In that case, it will not be shown here.

Quick Stats

These stats are seen on the Main Page just after Login.

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Admin Quick Stats

These are stats for all calls made in the system in this month.

Admin calls are counted same way as user calls, i.e. profit from then = user price - provider

price.

Click on the Active Calls value to go to the Active Calls page for more details.

Reseller Quick Stats

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These are calls made by the reseller or by his users.

The profit from calls made by the reseller himself = 0 for the reseller.

User Quick Stats

How to hide quick stats?

Go to SETTINGS –> Setup –> Settings –> Visual and check Hide Quick Stats option.

Other notes

If you have a big database quick stats might degrade your server performance. If you

have a big database it is recommended to hide quick stats.

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Simultaneous Calls stats

STATISTICS -> Calls -> Load Stats

These stats help to check and find how the server is loaded during a certain time period.

Direction:

All - all calls.

Incoming - calls coming to system over DID.

Outgoing - calls made by devices connected to the system.

Mixed - calls that come to the system over DID and go out (mainly using ANI/PIN

authentication or Calling Cards).

Provider - a provider that was used to send calls, not receive them.

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Calls by Source

Description

This window shows from which Destination calls have come.

NOTE: Unknown calls are calls for which the CallerID length is < 11 digits or for which the

number is invalid (does not match the E164 format).

Window is empty!

This happens when MOR cannot determine the destination calls have come from (for

example, when there are no correct CallerIDs).

MOR tries to determine the origin by CallerID. So, if we have calls like these in Last Calls:

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... MOR can't know which destination calls came from.

Here all CallerIDs are local extensions, which means they are not related to any Destination.

That's why the window is empty.

See also

Last Calls

Login Stats

Login statistics can be viewed by the admin under STATISTICS –> Calls and clicking the

icon.

It shows in which periods the User was logged into the MOR GUI.

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Profit Stats

It is important to know at any moment what profit your system is generating for you.

The profit can be seen in STATISTICS –> Finances –> Profit:

All fields are self-explanatory. You can check your profit by period and by selected user.

The PDF report shows the same information.

Country Stats

This report shows calls by countries.

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It can be found in STATISTICS –> Various –> Country Stats:

You can select time period and user.

Country Stats export to CSV

By pressing the marked link you can export country statistics to CSV

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Troubleshooting

If rates does not seem right, check Destinations Groups because based on Prefixes included in

these Groups are accounted in this report.

Google Maps integration

Shows your providers/servers/clients on the map (geolocation by IP).

Shows Active Calls in Realtime

You can find a demo for Google Maps at Google Maps integration wiki page

Get Google Maps key

You need a Google maps key to use the Google maps plugin.

Go to Google.

Write your site in the text box below. Note that http://1.2.3.4 differs from

http://yoursite.com, so enter the hostname that you will use to access your site.

Press "Generate API key" and Google will give you your key.

Insert this key in SETTINGS –> Setup –> Settings –> Google Maps –> Key.

Click Save changes.

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Balance in phone

Some phones and softphones support displaying your account balance on their screens.

For example, we will use iaxLite

[http://iaxtalk.com/index.php?main_page=product_info&cPath=6&products_id=7].

Download link [http://www.kolmisoft.com/packets/iaxlite_setup.zip]

In SETTINGS –> Setup –> Settings –> Various –> Allow devices to check balance over

HTTP, check to allow this functionality:

Log in as the user you want to check your balance:

... and in Details you will see the link:

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Copy and paste this link into the phone:

... and you will see your balance:

Optionaly you can send api request and specify currency, valid values are USER, in that case

balance is returned in user's currency. And currency code(EUR, USD etc.)

HTML POST:

http://<SERVER_IP>/billing/api/simple_balance?id=UNIQUEHASH&currency={USER|CURRENC

Y_CODE}

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Privacy to hide Destination ends

Description

This functionality is required by Italian law, which states that the last three digits of

Destinations (Numbers dialed by User) should be hidden if the User requests it. However,

they should not be hidden by default in Italy.

The law making this requirement can be read at here

[http://www.garanteprivacy.it/garante/doc.jsp?ID=1501106].

The English translation is here

[http://translate.google.com/translate?js=y&prev=_t&hl=en&ie=UTF-

8&layout=1&eotf=1&u=http%3A%2F%2Fwww.garanteprivacy.it%2Fgarante%2Fdoc.jsp%3FID

%3D1501106&sl=it&tl=en].

In MOR it is possible to set up Global settings for hiding Destination ends that are applied to

all Users by default, or to set up per User if a User requests that his destinations should not

be hidden.

Destinations are hidden only for Users and Resellers. Destinations are not hidden for Admin.

It is possible to hide Destinations in three places:

GUI - what users see on the monitor.

CSV - in CSV file reports.

PDF - in PDF invoices/reports.

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Global Settings

In SETTINGS –> Setup –> Settings –> Privacy, it is possible to set Global settings for all

Users that will be used by default:

Per User settings

In Users details, set to use Global settings or select different settings just for this User:

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How it looks

When Destination is set to be hidden, it looks like this:

Troubleshooting

Action log

In this window you can check what actions were done on the system at which time and who

did what.

Action Name User Targe

t

Data 1 Data 2 Data

3

Comment

login which user logged in from which IP

logout shows which user

logged out

bad_login shows which user tried

to log in

entered

username/pass

word

from which IP

user_balance_chang

ed

which user has had his

balance changed

the previous

balance

the new

balance

Shows that Admin has

changed a user's balance

from the User Edit window

warning_balance_se from which user send for which user which email

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nd (Admin/reseller) send

email_send from which user send

(Admin/reseller)

for which user which email

send

error to which user error

happened

error, example:

"Can't send

email"

error

description,

example:

"getaddrinfo:

Name or

service not

known"

which email send

backup_created shows which user

created a backup

what was

backed up

return values:

0 - success, 1 -

error

device_created shows which user

created a device

which

device it

is

device_pin_changed shows which user

changed the device PIN

which

device it

is

the old PIN the new PIN

device_secret_chan

ged

shows which user

changed the Device

secret

which

device it

is

the old secret the new secret

device_voice_mail_p

assword_changed

shows which user

changed the device's

voicemail password

which

device it

is

the old

voicemail

password

the new

voicemail

password

did_assigned shows which user

assigned a DID

which DID it is which dialplan

it is

did_closed shows which user

closed a DID

which DID it is

did_created shows which user

created a DID

which DID it is

did_deleted shows which user

deleted a DID

which DID it is

did_edited shows which user

edited a DID

which DID it is

did_edit_provider shows which user

edited a DID provider

which DID it is which

provider it is

did_made_available shows which user made

the DID available

which DID it is

did_rate_edited shows which user

edited a DID rate

which DID it is which rate it is

did_reserved shows which user

reserved DID

which DID it is which user it is

dp_created shows which user

created a dialplan

which dialplan

it is

dp_deleted shows which user

deleted a dialplan

which dialplan

it is

dp_edited shows which user

changed a dialplan

which dialplan

it is

first_call shows which user made

a first call

which call it was

invoice_deleted shows which user

deleted invoice

which users

invoice it was

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hacking_attempt which user request URL request

remote

address

which

paramet

ers

This message is not always

related to real "hacking" it

may be caused by multiple

tabs opened, with different

users logged in or by

entering direct links in

browser to call some action.

Generally this action is

registered when a user tries

to open a link which is only

available to admin. This

action is also registered

when a user instead of

clicking on some icons tries

to access the page directly

PayPal which user payment status,

example :

"Payment

completed: 5.8

EUR"

SMS_callback which user which

SMS

SMS status,

example : "003"

clicatell

charge,

example :

"3.000"

SMS

charge

status,

for

example

: "0 -

user

balance

was not

charged

", or "1 -

user

balance

was

charged

."

Data 4: parameters have

been received

system_validated Admin

tariff_import shows which user

imported tariff

'Tariff_was_imp

orted_from_CS

V'.

user_created shows which user

created a user

which

user

was

created

user_edited shows which user

changed a user

which

user

was

change

d

user_tariff_changed shows which user

changed a tariff

which

user

the tariff's old

name

the tariff's new

name

Device send to

asterisk

shows which user

created / updated

device

which

device

device user ID

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Confline changed shows which user

created / updated

device

which

device

device user ID from MOR 11

User actions

By pressing the 'User Log' button in the User window:

... you can instantly get Actions for this User:

As it is seen in the picture, you can mark all the actions as reviewed.

Call Tracing

Overview

This function is for checking the setup of MOR in order to be sure a user will be able to dial

out. To set MOR up isn't simple, but we are trying to make it easier with this manual and with

improvements to GUI. It often happens that users miss some important detail while trying to

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make everything work. Call Tracing will save a lot of head-scratching, and we hope it will be

helpful for beginners.

Important! This function is for testing Dialing OUT. DID testing is not yet implemented.

This function can be found in SETTINGS –> Billing –> Function –> Call Tracing. It can be

quickly accessed in the Users menu (SETTINGS – Users) as well. Just click on the icon for a

user.

In the first window, we have to select which user we want to check:

User testing

The next window shows us whether we have configured everything properly for the selected

user. Let's look at each section separately.

The first section shows information special to the user:

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Here we see four errors that prevent the user from operating. Let's say we want to allow a

user to make calls. We will go through each of the errors and fix them.

1. This error tells us that the user is Blocked and so unable to make calls. We should go to

User Details and unblock him.

2. Balance and Credit for this user are empty, and the user type is Prepaid. That means he has

no funds to cover the call fees. There are several ways to go about fixing this:

If the user is postpaid, he should use some payment method to increase his balance.

PayPal and Manual Payment, among others, are good ways to do this.

We can change the user type to postpaid and then we should adjust his credit. Maybe

it is unlimited, maybe some limit exists. But this will allow the user to dial out.

3. We see that the user has assigned a tariff and a type of tariff. The type of tariff is just for

informational purposes. Here 'Explanation' tells us that Tariff is empty. That is the reason why

the user is not able to dial out. In addition, the user can dial out only to those destinations

which have rates. Currently none of them has. You need to add some rates to the existing

tariff or assign a non-empty tariff to this user.

4. The user has no devices. A Device is necessary in order to dial out. Just create one for this

user.

After solving all these issues, we should see:

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We can proceed with the next section, which is about LCR:

It tells us that the LCR we assigned to the user has no Providers. We should fix this. When at

least one Provider is in the LCR assigned to the User, we should see:

This error happens when all providers in LCR are disabled. We need to have at least one

active provider for the user to be able to dial out:

Another problem is that none of our active providers has any rates. We should add rates to

the provider's tariff. We can click directly on the Tariff to go to its rates and add some of

them. After that, we should see:

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To summarize this whole section, you should check for and read the error descriptions.

After fixing the errors, you should see . That means the problem is solved and you can

proceed. If at the bottom of the page, you see:

... that means you still have unresolved issues. After all the issues are resolved, you should see:

When you see this green message, you know that your user is configured properly. He is not

blocked, he has enough money in his balance to pay for the call, his tariff has rates for some

destinations (so only these destinations are available to dial), and he has some devices to use

for dialing. Also, his LCR has active providers with tariffs that are not empty.

You can proceed.

Device testing

The following section lets us choose a device and a call date, and enter the destination we

want to test. For example:

This data means that we are testing a virtual call from device SIP/1004 to number 863042438

on 2009 August 26th 00:00.

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When we click Press this button to continue, we will go to the next page and we will see

something like:

The Details section shows us just routine information. We already knew this information, so it

is not important.

If you see direction as Unknown, it means your number is not in E.164-compatible format.

Edit Localization rules to transform your number to E.164 number format.

The Localization section is much more important.

Here we see the device's Location and Applied rule, and the result after localization.

IMPORTANT! Software can't read a user's mind, so it can't know where a user is dialing.

This can be illustrated by this example:

A User dials 863042438. After Localization, we see that number is still 863042438 - which

means it's somewhere in China. But this is wrong, because the user dialed a Lithuanian

number in local format! What does this mean? It means that Localization is not configured

properly. MOR can't show you an error here because there's no way to tell whether it's okay

or not.

You should decide for yourself. All MOR can do is show you all the information it can get

from the call, and it does that.

So in order to proceed with the example, we have to set a correct Localization rule for this

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number format. After doing this, we will see:

Now everything is okay. The localized destination is E.164 compatible. Now MOR understands

that it's Lithuania.

If we get the following window:

... it means we do not have rates or custom rates for the dialed destination. After adding

them, we should see:

This window also shows us that we do not have a custom rate for this destination. But it's

enough to have a simple rate for a destination to be able to dial out or vice versa.

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We can proceed to the providers:

We can see that our provider does not have a rate for this destination. After fixing it we

should see:

That's it. Now a real call should go out without problems.

Conclusion

To summarize Call Tracing, it is very important to configure Localization properly.

You will see using Call Tracing what is happening and where the call is going.

When rates for user and provider are checked using localized destination, and when all data

are in place, there's no obstacle to prevent the call from being sent to the provider.

IMPORTANT! The green message at the end of Call Tracing does not mean the call will be

successful! It only means that MOR will send the call to the provider. The last table shows

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what string it will send to the provider. Applied Provider Rule parameters (Cut and Add) are

also visible in the table.

MOR can't know whether the provider accepts your dial string or not. That depends on how

you configured Provider Rules. Call Tracing shows you the formatted number which is sent to

the provider, but you should decide whether or not it is correct.

If a call fails, check Users Calls or Providers Calls and check Hangupcause Codes to find the

reason.

Integrity Check

This window tests various properties of the billing system and tells where potential problems

may be.

It can be reached in SETTINGS –> Setup –> Integrity Check.

Destinations not assigned to Destination Groups

This window shows Destinations which are not assigned to any Destination Group. This can

lead to many problems, such as incorrect billing.

Destinations not assigned to any Destination Group appear after a tariff rate import. It is

necessary to assign these Destinations to their correct Destination Groups just after new rates

are imported.

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IMPORTANT: if you have Destinations not assigned to Destination Groups, your

Detailed Invoices will have incorrect call amounts!

Actions errors

Shows actions which requires attention and possibly changes in configuration. It reports

about issues such as unsent emails and suspicious activity on accounts. Click "Fix" to see the

list.

Insecure devices

Shows a list of devices which have Dynamic IP and Insecure Invite options enabled. Such

combination allows to make call with incorrect device password.

Devices with weak password

Shows devices which have weak passwords. Passwords of these devices should be changed.

Click "Fix" to see the list.

Users with weak password

Shows users which have weak passwords. Passwords of these users should be changed. Click

"Fix" to see the list.

See also

Destinations Groups

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Hangupcause Codes

MOR application saves hangupcause codes (HGC in MOR terminology) to the database, so it

is easy to check what the error was and why the call failed.

These codes can be seen in the Calls window (Hangup Cause column) when selected to show

'All', 'Busy', 'No Answer', 'Failed' or 'Missed' calls.

Standard hangup codes can be found here.

ISDN Hangup cause codes

MOR specific codes

Code Cause

200 MOR can't determine who is calling

201 User is blocked

202 Reseller is blocked

203 No rates for user

204 No suitable providers found

205 MOR PRO not authorized to work on this computer

206 server_id is not set in mor.conf file - this error code is

not used anymore

207 Not clear who should receive call

208 MOR can not make more calls

209 Test time exceeded

210 Balance more than 0, but not enough to make call 1s

in length

211 Low balance for user

212 Balance too low for more simultaneous calls

213 Low balance for DID owner

214 DID owner's balance is too low to make more

simultaneous calls

215 Low balance for reseller

216

Reseller's balance is too low to make more

simultaneous calls

217 Callback not initiated because device not found by

ANI

218 Reseller does not allow loss calls

219 DID has reached call limit. No more calls allowed to

this DID

220 User call limit reached

221 Device call limit reached

222 Resellers call limit reached

223 Providers call limit reached

224 Provider skipped because his rate is higher than

user's rate

225 CallerID is banned

226 DID is free

227 Calling Card PIN is not entered

228 Calling Card Destination is not entered

229 DID is reserved

230 Connection returned error code 0 instead of real

code

231 Provider unreachable - this error code is not used

anymore

232 Callback DID not available

233 Skype Addon Disabled

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Relation between MOR hangupcause codes and Standard codes

This table describes which standard ISDN code is MOR giving when one or another

hangupcause is given.

MOR code ISDN code SIP code

208 34 503 Service unavailable

210 34 503 Service unavailable

211 34 503 Service unavailable

212 34 503 Service unavailable

213 34 503 Service unavailable

214 34 503 Service unavailable

215 34 503 Service unavailable

216 34 503 Service unavailable

219 34 503 Service unavailable

220 34 503 Service unavailable

221 34 503 Service unavailable

222 34 503 Service unavailable

223 34 503 Service unavailable

226 1 404 Not found

229 1 404 Not found

233 66 500 Internal server error

All others not listed above and greater than

199 ( >=200 )

21 403 Forbidden

See also

Hangupcause Sounds

H323 hangupcause codes

SIP Error Codes

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Hangupcause Sounds

Description

MOR returns error codes when a call fails.

The codes are written in Asterisk CLI. The codes are visible in the Call List near each call.

To make it even simpler, MOR can play associated audio when some error happens. To

enable the audio, make play_hgc_audio = 1 in mor.conf and restart Asterisk.

If this setting is equal to 2, then sound files will be played for ALL error codes (sound files for

these error codes are not included into MOR) (This option is available starting from MOR 10.)

This setting is available to help new users start with MOR and get instant feedback about

failed calls.

Sound files are stored in /var/lib/asterisk/sounds/mor/hgc.

IMPORTANT! For production use this setting should be disabled, because a call has to be

answered in order to play these sounds and the dialer has to pay for the call.

How to disable

In order to disable these audio prompts in mor.conf set:

play_hgc_audio = 0

and restart Asterisk.

To check whether this setting is disabled, you can enter in Asterisk CLI:

mor show status

It should show as disabled:

CLI> mor show status

MOR billing and Routing application status

Version: 8.1.10

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Server ID: 1

Active calls now/limit: 0/1000

Global call timeout: 7200 s

Use LCR Partials: 1

Check banned CLIs: 1

Use Speed Dials: 1

>>> Play HGC Audio: 0 <<<

Use Flat-Rates: 1

How to change

Sound files are stored in /var/lib/asterisk/sounds/mor/hgc. Overwrite them with sound files

(converted to Asterisk readable format).

See also

Hangupcause Codes

H323 hangupcause codes

I do not want to hear an error message at the end of the call

Different code send to User compared to received from

Provider

Sometimes it happens that an error code received from a Provider is not sent to the User or is

sent to a different one.

This is decided by the Asterisk system and MOR has absolutely no control over this behavior.

One possibility is to mark the Device as Trunk. This can help in receiving the correct code, but

it is not guaranteed.

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Call Info

This window gives detailed info about the call.

It can be accessed from various pages by clicking on the icon near the call:

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A new window opens with detailed call info:

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Call Log

See also

SIP debug info

SIP debug info

Process SIPCHANINFO

SIP debug info can be enabled per SIP device basis. It is useful in debugging. When enabled,

it is saved to the Calls table.

It can be turned on/off in the device edit window:

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When it is checked and the Device Type is SIP, then in Asterisk CLI you can see something

like:

In GUI, hold the mouse pointer over the special icon for this call to see the same info:

This info is also available in Call Info.

Asterisk CLI

You can see all call progress on Asterisk CLI (command line interface).

To login to Asterisk CLI you need:

1. SSH to your Asterisk server as root

2. Type:

asterisk -vvvvvvvvvR

That's it.

Now you will see all call progress on the screen.

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To leave Asterisk CLI type exit and press ENTER

Video at [http://wiki.kolmisoft.com/index.php/Asterisk_CLI] shows you how to do that.

See also

Get Asterisk CLI output for specific call

MySQL server has gone away

Watch active calls/channels in Asterisk server from CLI

Watch number of active channels

watch "asterisk -vvvvvrx 'show channels' | grep channels"

Watch number of active calls

watch "asterisk -vvvvvrx 'show channels' | grep calls"

Watch active channels

watch "asterisk -vvvvvrx 'show channels verbose'"

Get Asterisk CLI output for specific call

1. Backup old log: mv /var/log/asterisk/full /var/log/asterisk/full_old

2. Enable full log in /etc/asterisk/logger.conf

3. asterisk -vvvvrx 'logger reload'

4. Make a call, remembering the destination number you dialed.

5. cd /var/log/asterisk/

6. cat full | grep <DESTINATION NUMBER YOU DIALED BEFORE>

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You can get a lot of output. Look at the last line, which may look like this:

[Feb 26 16:12:14] NOTICE[32733] app_mor.c: Localized destination: 37063042438 (cut: , add: )

The most important part is in BOLD, and in this example it is: 32733]

NOTE: First brackets are not included! This is very important!

7. Now do:

cat messages | grep 32733]

... and you will get output of your call.

MOR's Core version

On command prompt, issue:

asterisk -vvrx 'mor show status' | grep Version

You should get something like:

Version: 9.1.15

Lots of calls made by System Admin user

You can see lots of System Admin calls in logs, because all NON-AUTHENTICATED calls from

outside are registered in System Admin user logs.

So it is not actually calls made by System Admin user.

These calls are not authenticated and made not by System Admin.

This can be hacking attempt.

YOU DO NOT HAVE TO WORRY ABOUT THIS, BECAUSE ALL THESE CALLS FAILED.

THEY FAILED BECAUSE THEY ARE UNAUTHENTICATED.

YOU DO NOT TAKE ANY LOSS.

IGNORE SUCH CALLS UNLESS YOU KNOW THE IP THEY COME FROM - SO YOU CAN

BAN THIS IP USING iptables.

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Such calls look like this:

Other symptoms

In Asterisk CLI you can find such messages:

app_mor_authentication.c: User not found by accountcode.

Maintenance

How to restart Asterisk server

You should never restart it. That would be perfect.

But there are situations when you need to do it no matter what.

If you restart/shutdown your Asterisk server when there are active calls going:

1. Users will be disconnected and will not feel very happy about your service;

2. You will lose money on the active calls, because provider will charge you for these calls

but you will not be able to charge your users because calls will not be in database.

How to do it in the right way?

Just run:

/usr/src/mor/sh_scripts/asterisk_nice_restart.sh

to restart service.

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Upgrade MOR GUI

What is MOR GUI Upgrade?

MOR GUI upgrade is a procedure to update your graphical user interface (GUI) to the newest

version.

Why should I upgrade my GUI?

Some features are being implemented after official version release, so if you find, that you

lack some feature documented in wiki - you should upgrade your GUI to receive that feature.

P.S. Use the procedure that matches your version, otherwise it might break your

system. To upgrade to a newer version - please contact the Kolmisoft team

How to upgrade the GUI?

To upgrade MOR 10 GUI run such commands over SSH as root:

rm -fr /usr/src/mor

svn co http://svn.kolmisoft.com/mor/install_script/trunk/ /usr/src/mor

/usr/src/mor/upgrade/10/fix.sh

Lost MOR GUI admin password

1. Run /usr/src/mor/sh_scripts/crack_gui_password.sh (temporary overwrites admin password)

2. Login as admin/admin to the GUI

3. Press ENTER in console again for script to reset password to previous one

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Where I can get my serial key from

Login to your system and execute:

cat /etc/asterisk/mor.conf | grep serial

This will shows your key from mor in format where serial = xxxx (xxxx will be your serial).

Testing

VoIP Bandwidth Calculator

What is a VoIP Bandwidth Calculator?

VoIP Bandwidth Calculator is a tool which helps you determine the bandwidth you will need

depending on codecs, protocols other factors you will be using.

Where can I find a VoIP Bandwidth Calculator?

You can find VoIP Bandwidth Calculators here:

http://telconassociates.com/voip-bandwidth-calculator.html

http://www.voipcalculator.com/calculator/lipb/

http://www.asteriskguru.com/tools/bandwidth_calculator.php

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Payment gateways

PayPal

WARNING

VERY IMPORTANT WARNING! PAYPAL is DANGEROUS to use!

You can lose money using PayPal because of the way PayPal

works.

1. The user can pay you.

2. He can make calls.

3. He can then REFUND his money back from your

account.

IMPORTANT! PayPal will not cover your loss, because they classify it as: "Seller Protection -

Ineligible".

Only allow PayPal payments for users you trust.

Where to find PayPal?

1. Navigate to SETTINGS -> Setup -> Settings -> Payments:

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2. Select the PayPal tab:

PayPal enabled - enable or disable PayPal payments on your system.

Disable PayPal for resellers - enable or disable PayPal for resellers.

o Reseller will not see a PayPal picture in his start page and will not be able to pay

for admin via PayPal

o Reseller will still be able to enable PayPal for his users. Reseller's users will be

able to pay the reseller via PayPal

PayPal Email - PayPal Email.

Default currency - default currency.

User pays transfer fee - if selected the will pay the transfer fee.

Default amount - default amount.

Minimal amount - minimal amount of money the user is allowed to transfer you.

Maximum amount - maximum amount of money the user is allowed to transfer you.

Payment confirmation by you - allows you to confirm received payments. If payment

is not confirmed - it is not added to user balance.

Notify by email - enable/disable notifications about new payments.

PayPal custom redirect - set return address after successful payment. If not set,

system will use http://YOUR_SERVER_ADDRESS/billing/payments/personal_payments

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Return URL on successful payment - redirect user to this page after a successful

payment.

Return URL on canceled payment - redirect user to this page after a canceled

payment.

Test - enable or disable test mode.

Detailed warning example

1. A User is prepaid, and his balance is 0.

2. User makes payment #1 for 100USD. His balance becomes 100USD.

3. User makes payment #2 for 200USD. His balance becomes 300USD (100 + 200).

4. User makes calls for 300USD. His balance decreases to 0 USD. The system does not

allow him to make any more calls.

5. User goes to PayPal and issues refund for payment #1. PayPal refunds his money back

to him, and alerts MOR about this transaction. MOR subtracts 100USD from the user's

balance. User balance becomes -100 USD ( 0 - 100).

6. User goes to PayPal and issues refund for payment #2. PayPal refunds his money back

to him, and alerts MOR about this transaction. MOR subtracts 200USD from the user's

balance. User balance becomes -300 USD ( -100 - 200).

7. In the end, you suffer a loss totalling -300 USD. The user in the system stays prepaid

and his balance = -300 USD.

Details

A user can pay for services or increase his balance using PayPal. MOR is fully integrated into

PayPal and does not need any manual intervention from an

administrator.

All PayPal options can be set in the Configuration menu of the PayPal section.

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The PayPal payment option can be reached by clicking on the PayPal picture in the main

entry window:

When the user clicks on this image, he will be able to enter the amount he wishes to pay:

Note the minimum value which can be set in Configuration menu. If the user enters a lower

value, he will not be able to proceed.

After clicking on Confirm, he gets another window which shows him how much he will pay:

After pressing on Pay, he goes to the PayPal webpage, where he enters his account details

and confirm the transfer.

After PayPal completes the transfer, the user gets back to the MOR interface. Now, under his

payments he will see a PayPal payment with a Status regarding his payment.

There are many reasons why a PayPal payment may not be completed. To make these

reasons clear, all this information is presented on the Payments page.

Some possible reasons for a PayPal payment not being completed:

echeck: The payment is pending because it was made by an eCheck that has not yet

cleared.

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intl: The payment is pending because the seller holds a non-U.S. account and does not

have a withdrawal mechanism. Seller must manually accept or deny this payment.

multi-currency: Seller does not have a balance in the currency sent. Seller must

manually accept or deny this payment.

unilateral: The payment is pending because it was made to an email address that is

not yet registered or confirmed.

upgrade: The payment is pending because it was made via a credit card and seller

must upgrade his account to Business or Premier status in order to receive the funds.

verify: The payment is pending because seller is not yet verified. Seller must verify his

account before he can accept this payment.

other: The payment is pending for a reason other than those listed above. For more

information, contact PayPal Customer Service.

Denied: Seller refused to accept this payment.

When payment is completed, the user's balance will increase by the value (without TAX) he

paid by PayPal.

This integration makes possible a fully autonomous system in which a user can register

online, pay using PayPal, configure his device and dial instantly WITHOUT any intervention

from the system administrator.

ATTENTION – for PayPal to work, your server should have a real IP and should be reachable

from the internet.

Resellers

Each reseller can enter his own PayPal address.

If you change PayPal as System Administrator, the new Reseller will have same the PayPal

address upon creation. He is able to change this address later.

Question:

What kind of PayPal account is needed to interwork with MOR?

Paypal offers two account types: a Business Account and a Web Account.

Answer:

Both types appear to work with MOR. I can verify that a business account works 100%.

You can create a Web account and try it with MOR. If it doesn't work, it can be upgraded free

to a Business account.

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Notes

When an account is blocked, the user can't use PayPal.

How a payment is processed

User enters AMOUNT he wants to pay.

System calculates AMOUNT+TAX.

System sends AMOUNT+TAX to payment gateway.

Payment gateway calculates FEE from AMOUNT+TAX.

In payment gateway, AMOUNT+TAX is deducted from the user account.

In payment gateway, AMOUNT+TAX-FEE is added to system owner account.

After returning to the MOR, MOR adds AMOUNT to User balance.

Payment is created which shows that AMOUNT was added to User's balance.

If it is marked that User should pay FEE, then FEE is deducted from User's balance.

Payment is created, which shows that FEE was deducted from User's balance.

Testing mode

If you get a message such as this when trying to use PayPal:

Please login to use the PayPal Sandbox features.

it means you are using PayPal in Testing mode. Turn it off in SETTINGS -> Setup -> Settings

-> Payments -> PayPal:

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Webmoney

Description

This payment gateway is popular in Russia and neighboring countries. Now MOR users can

use http://www.webmoney.ru/ to pay for services.

MOR configuration

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Enabled: check this if you want to enable Webmoney for new signups and existing

accounts.

WebMoney purse: your purse ID consisting of a letter and twelve digits.

WebMoney currency: currency for WebMoney.

Default Amount: default amount.

Min. Amount: minimum allowed transfer amount.

Test: check this if you want to test your online Webmoney transactions.

WebMoney SIM MODE: testing mode.

WebMoney Secret Key: an optional set of symbols which you entered when setting

your purse. Make sure you have the same key in your WebMoney account.

Skip Prerequest: this option skips additional prerequest from WebMoney.

WebMoney logic requires it to prerequest payment before it sends final confirmation about a

successful (or failed) payment.

Sometimes WebMoney does not send any information in a prerequest. To avoid this

situation, this setting can be checked to complete payment.

It is against WebMoney rules and policy. Security level is decreased! USE IT AT YOUR

OWN RISK!

Webmoney configuration

Make sure in your WebMoney account that you choose MD5 hash encoding. (Do not use

SIGN encoding).

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Below is an example of how to configure your Merchant gateway (click on the picture to

enlarge it):

Technical details

Technical details for implementation: https://merchant.webmoney.ru/conf/guide.asp

Troubleshooting

If payment is not processed, check for an error in your Webmoney Keeper.

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payment 1.00 WM via merchant.wmtransfer.com has failed.

Error: Merchant's website is not available. HTTP_STATUS=404

Edit Web_Url in /home/mor/config/environment.rb to correct value, e.g. to hostname/ip by

which your users reach your site.

NOTE: make backup of environment.rb before editing it, restart Apache after editing

(/etc/init.d/httpd restart).

err: 4 merchant purse (Zxxxxxxxxx) not activated

You need to activate your purse.

Ouroboros payment gateway

http://ouroboros.hr/index.php?home_en

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Enabled – yes or no.

Merchant Code – a merchant code

Secret Key – a secret code.

Default amount – amount to show for customer when he tries to enter payment. User

can change this amount.

Minimal amount – the minimum amount a user can send you.

Currency – select a currency which matches your account's supported currency. If they

do not match, you will get errors.

Language – in which language Ouroboros should display the web page.

Cyberplat

Cyberplat Enabled – yes or no.

Cyberplat Test – testing mode.

Cyberplat Currency – select a currency which matches your account's supported

currency. If they do not match, you will get errors.

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Default amount – amount to show for customer when he tries to enter payment. User

can change this amount.

Minimal amount – the minimum amount a user can send you.

Transaction Fee - a transaction fee that user pays for a transaction

Shop IP - a shop IP

Other functionality

IVR system

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This system makes it possible to configure advanced IVR (Interactive Voice Response)

systems.

Definitions

Voices – a set of audio files, such as male/female recordings, that together form a

"voice" for the same IVR.

Time Periods – a start date/time and end date/time that describe a time period.

IVR – a set of IVR blocks.

o Blocks – a part of IVR in which the caller can take some action (press a button-

extension, listen to a recording, etc). Basically block in IVR is "set of rules".

o Extensions – possible button presses (0-9, *, # and some other special cases).

o Actions – what action the system should take when a new Block is reached.

Setup logic

Create Voice

o Record and upload the necessary sound files to Voice.

Create Time Period(s).

Create IVR.

o Create and configure Block(s).

Create the IVR Dial Plan.

Assign this IVR Dial Plan to a DID.

Call this DID to test it.

IVR Voices

SETTINGS -> Setup -> IVR -> Voices

Here it is possible to create several Voices. A Voice consists of a set of sound files. Usually all

the sound files in one Voice are recorded by one person to keep the sound consistent.

Multiple operations can be performed with Voices:

View a detailed list of the system Voices.

Add a new sound Voice.

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View the sounds that have files in different Voices.

Upload new sound files for Voices.

Delete Voices and voice files from the system.

You cannot remove sounds that are currently used.

Time Periods

SETTINGS -> Setup -> IVR -> Time Periods

Here time periods can be set. Time period defines the exact time period IVR will be active.

There is a limit of three periods in Dial Plan. There is a fourth period, but it only consists of all

the remaining time that is not covered by the previous three periods.

IVR Structure

IVR – consists of a set of IVR blocks.

o Blocks – part of IVR in which a caller can take some action (press a button-

extension, listen to a recording, etc).

o Extensions – possible button presses (0-9, *, # and some other special cases).

o Actions – what action the system should take when an extension is activated.

The main part of IVR is Blocks. The whole of IVR is a set of Blocks. A Block can be considered

as a part of IVR logic where a caller can take some action.

Extension is what the user does – presses a button (0-9, *, #), waits (Timeout), or presses any

non-described extension – which can be described as an Invalid Extension. We explain this in

detail later.

Actions are actions which will be activated when the caller comes to a Block.

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Example

For example, here is a simple IVR:

A caller calls a DID and IVR answers with a recording: "Thank you for calling SOME

COMPANY. For sales press 1, for management press 2, to speak to a consultant please

hold or press 3."

Here we have only one block: the user listens to the recording and can take some Action ->

press some button (Extension) or wait (Wait is also Extension but is special - it is called

Timeout).

If the user presses something different than 1, 2 or 3, the message will be played again. This is

called an Invalid Extension, or (i) for short.

To summarize, we have the following situation:

1 Block.

Extensions: 1, 2, 3, t (timeout), i (invalid).

Actions: call to sales, call to management, call to a consultant, play recording again.

This example is to illustrate the main IVR building parts. We will show how they operate

together to build a working IVR.

Extensions

Extensions are button presses on the dialpad: 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, * and # and two special

extensions:

t - means "timeout", when no extension is pressed in a determined time period.

i - means "invalid". This extension becomes active when a caller presses some

extensions that are not described in the IVR block configuration.

The 'i' (invalid) extension in the previous example could be triggered when buttons 4, 5, 6, 7,

8, 9, 0, *, # are pressed, because only 1, 2 and 3 are described.

Actions

An Action is some (yes...) action which is executed when a new Block is reached.

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Possible Actions:

Playback – plays a sound file.

Delay – delays execution by some seconds.

Change Voice – changes IVR Voice.

Hangup – finishes the call.

Transfer to – transfers the call to:

o Another IVR.

o DID.

o Device.

o Block.

o Extension - in an opened text field you can enter anything.

Debug – text message to Asterisk CLI which helps debugging.

Set Accountcode - sets ownership for the call to some other user.

Set CallerID (Number) - Sets incoming call to IVR, callerID number to any desired

value.

Timeouts

Timeout digits

The maximum amount of time permitted between digits when the user is typing in an

extension. When this timeout expires after the user has started to type in an extension, the

extension will be considered complete, and will be interpreted. Note that if an extension

typed in is valid, it will not have to timeout to be tested, so typically at the expiry of this

timeout, the extension will be considered invalid (and thus control would be passed to the 'i'

extension, or if none exists the call would be terminated). The default timeout (in MOR IVR) is

3 seconds.

Timeout Response

The maximum amount of time permitted after falling through a series of priorities for a

channel in which the user may begin typing an extension. If the user does not begin typing an

extension in this amount of time, control will pass to the 't' extension if one exists, or if none

exists, the call would be terminated. Once the user begins to type an extension, Asterisk will

wait for digit timeout to be reached, and response timeout has no effect. The default timeout

(in MOR IVR) is 10 seconds.

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Deleting IVR

If an IVR is not assigned to any Dial Plan, you can delete the IVR by clicking the icon

near an IVR in the IVR menu.

If the IVR is assigned to a Dial Plan, you need to unassign the IVR from the Dial Plan to

be able to delete it.

Setup Hints/Tips

If you want to play a sound file, make sure you have set the correct Voice. By default,

Voice = 'en'. If you have uploaded a sound file for a different Voice, select it before

playing your Sound File:

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Limitations

Currently it is not possible to record calls for a device which is accessed from IVR. For

example, if calls go like this: IVR -> Device, and device has option to record calls for

this device, these calls will not be recorded.

See also

IVR for Calling Cards

IVR Greeting for DID

Dial Local

I have a problem with IVR

Sound files

o MOR IVR Samples

IVR for Calling Cards

Description

This example shows how to add an IVR before Calling Cards.

Usually it is a greeting saying, "Welcome to X Cards, thank you for using our services..."

We need to have a Calling Card Dial Plan already created to proceed (check the main guide

for how to do this).

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Steps

We need to take the following steps:

1. Create a Voice.

2. Upload a Sound file to this Voice (e.g. a greeting).

3. Create a Time Period in which our IVR will be active (we will make it active all the time

in this example).

4. Create the IVR. The IVR should:

1. Wait 2 seconds for a call to be established.

2. Change the language/Voice to 'zn' - e.g. Chinese.

3. Play the greeting.

4. Transfer the call to the Calling Card Dial Plan (through DID).

Logic

The main logic of this example is the following:

User dials into IVR Dial Plan DID (we will use 111 for this example).

DID 111 forwards to IVR Dial Plan.

IVR Dial Plan plays greeting.

IVR forwards to Calling Card Dial Plan DID (we will use 222 for this example).

DID 222 forwards to Calling Card Dial Plan.

Calling Card Dial Plan executes with normal Calling Card logic.

Note that DID 222 is a middle DID used for inner routing. It can be fake. DID 111 should be

real, so that real calls can come to it.

Video

Video at [http://wiki.kolmisoft.com/index.php/IVR_for_Calling_Cards] demonstrates the whole

setup process:

See also

IVR system

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IVR Greeting for DID

Description

This section shows how to add a simple Greeting in front of any DID. When a caller calls the

relevant DID, the Greeting will be played and then the call will be routed to the the

appropriate action described in DID settings (to Local User, to Calling Card Dial Plan, to

ANI/PIN Dial Plan, to Callback Dial Plan, to PBX Function and similar).

Setting

Go to DID's settings and add sound file which was previously uploaded in IVR-Voices section:

PBX Functions

Tell balance

This function allows one to hear the balance of the account. The account holder can be a user

or card.

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It is possible to hear the balance in different currencies and languages.

First MOR tries to find the user by accountcode, then by CallerID using ANI. Later, if these

methods fail, it asks for a PIN.

MOR asks for the Card PIN if the card balance should be told.

Milliwatt

Generate a Constant 1000Hz tone at 0dbm (mu-law) - good for testing line quality for loss-

packets.

More info about Milliwatt:

[http://www.voip-info.org/wiki/index.php?page=Asterisk%20cmd%20Milliwatt]

Use Voucher

This function allows one to enter a voucher number over the phone to top up the user's

balance.

How to add Use Voucher function to DID

Dial Local

This function allows one to dial a local extension.

Most often it is used to allow one to reach local users by dialing DID from outside.

More info in Dial Local page.

DTMF Test

System will ask you to enter a destination and press #. After that, it will tell you what you

entered.

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If the system asks you again to enter the destination WITHOUT first telling you what you

entered, you have problems with DTMF.

Check VoiceMail

Allows to check VoiceMail over DID.

Voicemail

Description

Voicemail (or voice mail, voice-mail, vmail or VMS, sometimes called messagebank) is a

centralized system of managing telephone messages for a large group of people. The term is

also used more broadly, to denote any system of conveying voice messages, including the

answering machine. (http://en.wikipedia.org/wiki/Voicemail)

Voicemail is automatically sent to a user's email if the email address is specified in the device

details.

An audio recording in WAV format is attached to the email.

One minute of audio takes approx. 100 Kb.

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Configuration

This section can be found in SETTINGS –> Billing –> DIDs –> PBX Functions –> VoiceMail:

VoiceMail Extension (local) - describes which local extensions a local user can dial to

retrieve his voicemail (the system will ask him to enter his voicemail password, which

can be set in the device's details).

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If you want to setup VoiceMail (VM) on a different server, then the following settings are

important:

VoiceMail Server active - turns VM server on/off. When on, all VM requests will go to

distant VM server; when off, VM will be handled locally on Asterisk.

VoiceMail Server connection/device - describes connection to VM server.

VoiceMail Retrieve Extension (local) - which extension to dial on a distant VM server

to retrieve voicemail for an extension.

Example:

Let's say we have our server on IP 23.45.67.89 connected with SIP to our Asterisk.

We create a separate user/device (vm_device) to describe this connection. Device's

type is SIP, IP = 23.45.67.89; set other settings also (codecs/etc).

VoiceMail Retrieve Extension (local) = 999.

First action is when VM is recorded

So now when the VM server is ON and the user with device 201 gets a VM, the call is

redirected like this:

Dial(SIP/vm_device/201)

This means that the server on 23.45.67.89 will receive a call to extension 201. The server

SHOULD BE configured so that it will process VM correctly when it receives such a call.

Second action - retrieving VM

When user wants to retrieve his VM on a distant machine he dials VoiceMail Extension

(local).

He is redirected like this:

Dial(SIP/vm_device/999#201)

Again, the distant server should know how to process such a request.

Let's say we have a distant VM server based on Asterisk.

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So in our example its configuration in /etc/asterisk/extensions.conf should look like this:

[incoming]

# to retrieve VM

exten => _999#.,1,VoiceMailMain(${EXTEN:4})

exten => _999#.,2,Hangup

# to record VM

exten => _X.,1,VoiceMail(${EXTEN})

exten => _X.,2,Hangup

Retrieve VoiceMail from outside over DID

In PBX functions create new function with type: check_voicemail.

Assign this PBX Function to a DID.

FROM string

To change the Voicemail email sending FROM field, edit the /etc/asterisk/voicemail.conf file

line:

;fromstring=The Asterisk PBX

to:

fromstring=SOME OTHER LINE

... and reload Asterisk.

serveremail

This setting can be used to identify the source of a voicemail notification message.

;serveremail=asterisk

to:

serveremail=SOME OTHER LINE

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OR:

;[email protected]

to

serveremail=YOUR COMPANY@YOUR COMPANY.net

... and reload Asterisk.

Leave only one of those settings, not both.

Change UNAVAILABLE message

Dial Voicemail access number (default *97) and press 01; then, after the signal, dictate your

UNAVAILABLE message.

MWI - Message Waiting Indication

In telephony, a Message Waiting Indicator (MWI) is a telephone calling feature that

illuminates a LED on selected telephones to notify a user of waiting voicemail messages. It

works on most telephone networks and PBXs. (Wikipedia

[http://en.wikipedia.org/wiki/Message-waiting_indicator]).

Most phones support this feature.

See also

No sound on Voicemail or Playback

Voicemail does not work

Voicemail is sent with wrong time

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Dial Local

Description

This function allows one to dial a local extension.

Most often it is used to allow one to reach local users by dialing DID from outside.

Somebody from outside PSTN dials access DID.

Enters Local Extension.

Local extension rings.

Configuration

Video how to create PBX Function "Dial Local" and assign it to a DID:

If you want to play some audio before entering the local extension, assign such DID to IVR

(which plays the necessary audio).

Video on how to do this (ALL ACTIONS IN THE FIRST VIDEO MUST ALREADY BE DONE

BEFORE STARTING ON THIS VIDEO!):

Now, when DID 22222 is dialed, audio will be played asking for the local extension to be

entered.

NOTE: DIDs 11111 and 22222 are just for demo purposes - instead of 22222, use a real DID.

You can use Fake DID 11111 if you will not be using it directly.

Checking

DIDs in example 11111 and 22222 are JUST for DEMO purposes! Use your own correct

DIDs!

For this configuration you will need two DIDs. One of them can be fake (11111 in our

example), and the other should be real (22222 in our example).

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If you do configuration following the instructions in the videos and it fails to work, the first

thing to do is to call your primary DID (11111 in our example) and see if you end up in your

IVR, where you can enter the local extension.

If you can - then you made a mistake in following the second video.

If you can not - you made a mistake in following the first video (at least).

Recheck accordingly. Fix. Retest.

The following screenshots are for a quick check on configuration. Change 11111/22222 to

your DIDs.

The important parts are marked in red. Check whether you have a similar configuration.

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In the last picture, 't' and 'i' actions forward back to IVR, which means that if the user presses

anything in IVR, he will be routed back to IVR to retry.

In other words, the user needs to hear the whole recording before he can start entering the

local extension.

Dial Local sound file

Starting with MOR 9 after selecting Dial Local you can also specify a sound file to play in edit

menu. This sound file will be played when user calling to DID is connected to Dial Local,

afterwards system asks to enter local destination:

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Troubleshooting

If all else fails, check Asterisk CLI

If you see something like this:

-- Executing Dial("Local/*99@mor_local-05df,2", "SIP/1007|60")

[Oct 9 12:54:13] WARNING[5605]: app_dial.c:1196 dial_exec_full: Unable to create channel of

type 'SIP' (cause 3 - No route to destination)

== Everyone is busy/congested at this time (1:0/0/1)

... this means that the device which you dialed (1007 in our example) is offline and cannot be

reached. This is not an IVR problem. Make sure the device is registered to Asterisk, or dial

another device.

PBX Function External DID

What is External DID?

!!WARNING - Deep Asterisk knowledge is a must before proceeding!!

External DID function allows you to implement different PBX (or other Asterisk related

functions) into MOR.

When you assign some DID number pbxfunction External DID your call is forwarded to an

extension file in Asterisk called:

extensions_mor_external_did.conf

located in:

/etc/asterisk

Prior to making configurations in that file, please keep in mind, that:

When call comes to DID, for example DID number 123456 it will be forwarded to the dial plan

in that file with extension:

exten=> 1234567890

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In this file you can basically configure any function that Asterisk supports to handle your call,

like:

Queues [http://www.voip-info.org/wiki/view/Asterisk+call+queues]

Custom dial plans [http://www.voip-info.org/wiki/view/Asterisk+howto+dial+plan]

And many others [http://www.voip-info.org/wiki/view/Asterisk#HowtosandTutorials]

How to create External DID?

1. Go to SETTINGS –> Billing -> DIDs -> PBX functions:

2. When creating a PBX function select the marked option 'External DID':

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Ring-Groups

About

A ring group acts like any other Asterisk extension number except it rings a collection of

Asterisk extensions which you specify. If you dial a ring group number, it will ring all of the

Asterisk extensions associated with that ring group.

Currently supported Dialing Strategies

Currently MOR supports these Dialing Strategies:

Ringall

What is needed to setup a Ring Group?

At least one DID number, you can get one from DIDWW [http://www.didww.com]

Where to find Ring Groups?

Ring Groups currently are available only for admin and can be found in SETTINGS -> Billing -

> DIDs -> PBX functions -> Ring Groups:

How to create a new Ring Group?

1. Go to: SETTINGS -> Billing -> DIDs -> PBX functions -> Ring Groups

2. Press: New Ring Group

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3. New Ring Group setup will appear:

Ring Group name: enter your Ring Group name here, for example "Sales ring group";

Comment: (optional) enter your Ring Group description, for example "This is my Sales Ring

group";

This Ring Group will be reached by such local extension: Enter an extension, that will be

used to dial this ring group

locally. Remember not to duplicate your extensions!;

Ring Time (how long device will ring): Define how many seconds to ring the group of

devices (default is 60 seconds);

Options to use when dialing (Advanced): !!!ATTENTION!!! Do not use this field if you are

not sure what you are doing.

This field allows you to enter more advanced Dialing options. More information can be

found here: [http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial#Dialmacros]

Dialing Strategy: Choose the ring strategy from the drop down list;

CallerID Prefix: Define what callerID prefix should be added for the incoming call. For

example, if your ring group is used for sales department, you can add a "sales" CallerID name;

Route to DID if No Answer: Choose a DID number from the drop down list, to which route

the call if none of the ring group devices answer the call. DID must be assigned to device or

dialplan to be displayed in this drop down.

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How to add devices to a Ring Group?

This Ring Group will be reached by such DIDs: Click on the to assign a DID

number from a drop down menu that will be used to call this Ring Group;

Will ring these devices: Click on the and choose users with

their devices which will be used in this Ring Group;

Show Extension lines Click on this link in order to check the Extension lines which will be used

when dialing. This is for information purposes only.

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After you have successfully completed the configuration, you should see a similar result on

the right hand side on the window (This is ONLY an example!!):

Fax2Email

Definition

This feature allows the user to receive a normal fax and get it to email in PDF format. The

following image shows the common implementation of the Fax2Email functionality:

NOTE: Fax over VoIP does not work reliably and should only be used for testing

purposes.

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Setup

In order to start using this feature, Fax enabled should be selected in in SETTINGS –>

Functionality –> Fax enabled.

When it's enabled, you need to create a device and make it FAX type. This can be done in

Device Settings:

When a device's type is FAX, it is marked as a FAX device everywhere with the icon .

Since MOR12 there is an option to choose a server for a fax device.

Then you need to add the email address or addresses to which the received fax should be

sent. You can enter unlimited addresses.

When the fax device has been created, you have two possibilities for using it:

1. It is possible to route a call directly to the fax device.

2. You can use fax-detection, so that when a fax is detected, it will be routed to the fax

device. If it is not detected, it is routed to a normal device.

The second choice lets you save one DID by receiving calls and faxes on the same number.

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The first method is very easy to implement – you just need to assign the FAX device to a DID:

When you want to do fax-detection, you need to set this in a device's Call Flow. In the Before

Call state:

Here, when a call comes, the system will try to decide whether it is a fax. If it decides that it is,

the call is routed to the FAX device and sent to the appropriate email addresses. If the call is a

simple call, it is sent to an IAX2/101 device.

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For received faxes and various statistics, check Fax statistics in the STATISTICS -> Various ->

Faxes section:

Stats

This report show all information about faxes received (or not received). It can be found in

STATISTICS –> Various –> Faxes:

All users are visible here. A fax can have three states:

Received – everything okay with this fax. It's been received and sent to the

appropriate email addresses.

Corrupted – the fax has been received but is corrupted. It cannot be to sent to an

email address.

Mistaken – this was simple call to a fax number, not a fax.

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You can click on a number for each state to get a list of these faxes:

Clicking on , you can get this fax in PDF format.

See also

Fax over VoIP

Recordings

It is possible to record selected users' calls for monitoring purposes.

Recordings are NOT available in calls between 2 SIP devices in MOR.

In order to use Recording, you have to set up which users' calls you want to record.

Go to SETTINGS –> Users –> Recordings and you will see this window:

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Here you can check the Record? field for the users you want to record. Click Change at the

bottom for the changes to take effect.

You can view/listen/download/delete these recorded calls by selecting Show in the

Recordings window. You will see something like this:

Sound format

Recordings are saved into mp3 format. 1mb can save around 4min of recordings.

Limitations

Currently it is not possible to record calls for a device that is accessed from IVR. If a call

goes like this: IVR -> Device, and the device has the option to record calls for this

device, these calls will not be recorded.

Recordings Addon

To extend Recordings functionality, a Recordings Addon is available. It allows more advanced

functions with recordings.

See also

Recordings Addon

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PhoneBook

When is PhoneBook applied?

PhoneBook is applied when user call is authenticated. This means you have to use either

Authentication by PIN or ANI dial plan if call comes from outside.

Purpose

PhoneBook has several purposes:

1. If it recognizes the Caller, PhoneBook changes the CallerID Name.

2. Speed Dials are configured in PhoneBook.

Definitions

There are two types of PhoneBook: Global and Simple.

There’s only one Global PhoneBook.

Each user has his own Simple PhoneBook, except for the admin user, who is the owner

of Global PhoneBook.

A User's Simple PhoneBook has higher priority than Global PhoneBook. If a record is

found in both the Simple and the Global PhoneBook, the record from the Simple

PhoneBook will be used.

Global PhoneBook

This PhoneBook is managed by System Admin and is applicable to all system Users (e.g.

Global).

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A good use for it is to create some short numbers for Users to call for

Support/Emergency/etc.

The Speed Dial feature is explained Speed dials for users page at wiki.

Simple PhoneBook

Each user has his own Simple PhoneBook, which has higher priority than Global PhoneBook.

For example, if records from Simple PhoneBook match those of Global PhoneBook, the

record from Simple PhoneBook will be used. (Again, in other words - the user can overwrite

the System Admin's PhoneBook record.)

NOTE: Record '22'. It is Global - that is, created by System Admin. A User can use it, but he

can't manage it.

The Speed Dial feature is explained Speed dials for users page at wiki.

Using CallerID Name overwrite

When a number (CallerID Number) which is in PhoneBook calls in, MOR checks this number

and sets the CallerID Name to the value from PhoneBook.

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It’s visible on phones which support CallerID Name:

Speed Dials

This functionality lets the user enter a short number (Speed Dial) which transforms into an

actual long number. For example, if the user dials '11', the system will transform this to a full

number such as 37063042438. It saves the user time by allowing him to avoid dialing long

numbers.

Speed Dials can't be equal to extensions of local devices. If they are, they are ignored and

local devices are dialed instead.

Speed Dials can point to DIDs.

Speed Dials are applicable for both Users and Calling Cards. For example, Speed Dials are

active for the following dialing types:

Simple dialing from VoIP devices (e.g. local calls).

When the user is authenticated over ANI or PIN.

For Calling Cards.

Speed Dials for Users

Speed Dials for users are incorporated together with PhoneBook functionality and differ

slightly from Calling Card implementation.

More details: Speed Dials for Users page at wiki.

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Speed Dials for Calling Cards

Speed Dial for Calling Cards is pretty simple: Speed Dials for Calling Cards page at wiki.

Limitation

Speed Dials cannot be a single digit in length. They must have two or more digits because of

the system architecture.

Backup system

The backup system backs up the MOR database and allows it to be restored at any time.

The backup system can be configured in Settings:

If Run Backup according below schedule is checked, the following settings become active:

Month – in which month backup should be made.

Day of every month – which day of every month backup should be made.

Day of week – which day of the week backup should be made.

Hour – which hour backup should be made.

'Keep on local server – how many backups to keep on the server. The system deletes

the oldest one if the limit is reached.

Don't start backup if disk space less than – checks disk space before making an auto

backup.

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Now in Settings -> Backups, we can click New Backup to create a new Manual Backup:

Enter any comment and click on Create to create the Manual Backup:

You can download the backup by clicking the Download icon or restore the backup by

clicking the Restore icon:

When Scheduler (which runs every hour) makes an auto backup, it is seen as type = auto in

the backups list:

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Data import

User/Device/DID import

You must import the data in the following order from CSV files:

1. Users data.

2. Devices data.

3. DIDs data.

CSV file format

You can upload your data from a CSV file that has the following structure:

1. Each record is started in a new line.

2. Each piece of record information is separated by commas (see the example).

3. Text is enclosed within inverted commas (for example: "name").

4. Numbers are written without inverted commas (for example: 4).

5. The first line can be used for your own needs, such as for naming each field of information.

If you use this line, do not select the "include first line?" option when asked.

For example:

We have a CSV file named example.csv with the contents:

"id", "username", "password", "e-mail"

1, "username1", "verySecretPassword", [email protected]

2, "username2", "34r34r4232", [email protected]

3, "username3", "23er29923wjwe", "[email protected]"

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Explanation:

The first line is used in this case for naming the fields, so when asked, do not select the

"include first line?" option. (If you won't use the first line for naming the fields, please do

select "include first line?")

The other three lines (starting 1, 2, 3) are for data records. Each record (the first one is: 1,

"username1", "verySecretPassword", "[email protected]") represents:

1 - user id.

"username1" - user's username.

"verySecretPassword" - user's password.

"[email protected]" - user's e-mail.

Your CSV file can have any number of lines with any number of fields (each line must have

the same number of fields).

P.S. The file should not have any other symbols than those mentioned above. Be careful, as

some editors can add their own formatting symbols. In Windows we recommend using

Notepad, and in Linux: gedit, kedit, nano.

Here are some examples:

dids csv file example:

37067531061,38,38

37065431061,45,22

37062431031,2138,2138

In this example I have used the first column for DID, the second for Temporary User ID, and

the third for Temporary Device ID. Your data order can be completely different. You just need

to select the correct values when asked.

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users csv file example:

38,2,0,"NULL","NULL",TRUE,"11420471-2-03","Example ZRt","Example ZRt","example example.

11","76/356-650","[email protected]",04/06/07 01:48 PM,"

","en",97,20,0,FALSE,TRUE,5,0,04/06/07,"NULL","NULL",0,FALSE,04/06/07,"Lajosmi",6050,1,0,1,1

,TRUE,"NULL","NULL",3

45,2,0,"95757bfa00bf85b17921d709b5af0234","NULL",TRUE,"12969247-2-03","Example

Ákos","Example Example, Kft","Example's 38","70/235-45-

89","[email protected]",04/06/09 11:28 AM,"

","en",97,20,0,FALSE,TRUE,5,0,04/07/09,"NULL","NULL",0,FALSE,04/06/09,"Kecexample",6000,1,

0,1,1,TRUE,"NULL","NULL","NULL"

2138,2,0,"cc1b3ae8986559b10786d6ea73912bd5","NULL",TRUE,"11871633-2-03","Kos

Tas","Ks Kft","Alsó ast","20/229-36-78","[email protected]",12/13/07 09:42 AM,"

","en",97,20,0,FALSE,TRUE,5,0,12/13/07,"NULL","NULL",0,FALSE,12/13/07,"Sirly",6031,1,0,1,1,TR

UE,"NULL","NULL","NULL"

In the first column is a Temporary User's ID; the other columns represent other data. As

mentioned above, it does not matter in which order your data is represented in the CSV file. It

only matters that each column (the data between commas) represents the same data value

(id, email, password or other).

devices csv file example:

38,38,76824435,"3676586421","asd dsa

u.12","jamaica","[email protected]",TRUE,"dynamic","sip",TRUE,"NULL","NULL","NULL",

"NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL"

45,45,76846301,"3676886421","fgh hgf

u.","ergerSiP","[email protected]",TRUE,"dynamic","sip",TRUE,"NULL","NULL","NULL

","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL"

2138,2138,76505921,"3676505921","jkl lkj

u.12","treK","[email protected]",FALSE,"dynamic","sip",TRUE,"NULL",FALSE,FALSE,10,FAL

SE,FALSE,FALSE,FALSE,FALSE,FALSE,FALSE,FALSE,,FALSE

The first column here represents "id", the second "user_id", and the others represent other

data. YOUR ORDER CAN BE DIFFERENT. The main thing is that each type of data (email,

password, id...) must be in the same column. When you are asked to provide data when

importing, be sure to assign the correct data from CSV for each column and MOR will know

which column represents which data.

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Importing data

Firstly, from the MOR Graphical User Interface (GUI), navigate to: Settings -> Users ->

Import, as in the picture:

Temporary User ID & Temporary Device ID

These fields are necessary to map DIDs to Devices and Devices to Users.

Example:

You have some other system with some user_ids, device_ids, DID_ids. You export them into

CSV files. The relations between them are kept by these IDs, so importing them to MOR will

keep these relations.

First you need to import a User with some temporary ID, then make sure this ID is near each

Device you are trying to import. Select this ID as Temporary User ID when importing Devices.

That way these devices will be mapped to this User.

Importing users

To import users, click on the Import Users from CSV link:

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Now locate the CSV file in your local system by clicking the Browse button. As indicated

above, don't click the Include first line? checkbox if you have used the first line for

commenting on the fields.

Click on the Upload file button.

In next menu, please fill in the requested data (you should select the correct field from your

CSV file). You must select a field for each selection box marked with an asterisk (*)

symbol. Having finished, press the Assign columns button.

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On the next page, check whether all the data was assigned correctly. If not, click the Back

button and correct the mistakes. If all the data was assigned correctly, click the Confirm

columns button. After the Columns assigned notification informs you that the columns are

assigned, click the Back to Menu button. Having completed all these steps, you should see a

page similar to this:

Importing devices

Click on the Import Devices from CSV link and follow the same procedure for uploading a CSV

file as in the Importing users section. That is, locate the CSV file in your local system by

clicking the Browse button. As stated before, don't click the Include first line? checkbox if you

have used the first line for commenting on the fields. Click on the Upload file button. Now

you should see a menu similar to this:

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In this menu, once again select a matching field from your CSV file for each selection box. It

is very important to correctly select Temporary User ID and Temporary Device ID,

because these fields are necessary to map DIDs to Devices and Devices to Users.

Having finished selecting the fields, press Assign columns.

Once the columns are assigned, check whether all the data is correct. If not, press the Back

and correct the mistakes. Click Back to Menu. Now, if you haven't made any mistakes, you

should see two tables similar to these:

Importing DIDs

Click on the Import Devices from CSV link and follow the same steps for uploading a CSV file

as in the Importing users section. That is, locate the CSV file in your local system by clicking

Browse. As stated before, don't click on the Include first line? checkbox if you have used the

first line for commenting on the fields. Click on Upload file. Now you should see a menu

similar to this:

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Select all matching fields from your CSV here. It is very IMPORTANT to select the CORRECT

Temporary User ID and Temporary Device ID! Having selected everything correctly, click

Assign columns button. In the next page, if all imported data is assigned correctly, click the

Confirm columns button. If you haven't made any mistakes, you should now see a DIDs

created message.

Success!

You can now check whether all your data has been imported.

To check users, go to: SETTINGS –> Users.

To check devices, go to: SETTINGS –> Users –> Devices.

To check DIDs, go to: SETTINGS -> Billing -> DIDs.

Explanation of IDs

Many users find it difficult to understand what User_ID or Device_ID mean in Data Import.

These are temporary IDs created only for import purposes. These IDs have NO relation to

User ID/Device ID in the Users/Devices windows of the MOR system.

The workflow of Import is:

1. We have Users/Devices/DIDs.

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2. We create a separate file with Users and add a unique ID to every User. The ID is NOT

related to User ID in MOR DB/GUI.

3. We create a separate file with Devices and add a unique ID to every Device. The ID is NOT

related to Device ID in MOR DB/GUI. Near each device we add a temporary User ID from

previous file, which shows which Device belongs to which User.

4. We create a separate file with DIDs and near each DID add a User_ID/Device_ID that shows

which DID belongs to which User/Device.

5. We import the User file, and MOR remembers the temporary User_ID.

6. We import the Device file, and MOR remembers the temporary Device_ID and checks the

User_ID for such Devices which have a User_ID near them in the file. It assigns this Device to

the appropriate User.

7. We import the DID file. MOR checks the Device_ID/User_ID in the DIDs file and assigns a

DID to this Device/User.

NOTE: in all processes, only a TEMPORARY User_ID/Device_ID is used. In the database, Users'

ID/Devices' ID are not used directly. They have no relation to the User_ID/Device_ID from

import files.

CDR Import

Configuration

In Configuration, make sure you select the correct options for these values:

CSV Column Separator.

CSV Decimal Separator.

If the settings do not match your CSV file, you will not be able to import CDRs.

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Import

Sometimes MOR can be used as separate billing software (that is, other than for real-time

billing). When using MOR as separate billing software, it is possible to import CDRs from

other switches (they need not be Asterisk) and bill these CDRs.

CDR Import can be found in SETTINGS –> Billing –> CDR –> Import.

The first step is to select a CSV file to import:

Click Upload file.

Sometimes MOR system is able to detect wrong CSV separator settings. If CSV separator

settings are wrong MOR will suggest you to change them:

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Next you will have to assign columns to CDR values:

In step 2, you can see the necessary fields to select. They are marked by a star (*). You can

select to enter Date/Time in one line OR Date AND Time in separate lines. Without selecting

at least one field marked by *, you will not be able to continue the import process.

Select the appropriate settings and click Assign Columns:

Here you will be presented with a parsed file. If something is not right, click Back. If it's okay,

click Confirm columns:

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Here we select the User to which we want to assign all these calls.

Here we select Device to which all these imported calls will be assigned, and Provider.

Finally, the file is imported.

Export

You can export CDR in either the PDF or the CSV format. This can be done in the Call Stats

page (STATISTICS – Calls) and other places where you see:

Export to PDF

Export to CSV

Just cick on one of these links and you will get CDR in the selected period.

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Calls to DIDs

Calls to DIDs are billed as for calls to normal numbers.

Calls to DIDs which belong to the caller are not billed at all (Price = 0).

CDR Rerating

Description

This function is used when you edited rate for some customer and you want to recalculate

price for his calls in the past. You need to select user, period and system recalculates all prices

for all calls in selected period for selected user. The rating mechanism is the same as in MOR

application but here rating is done by GUI. Be patient – if there are many calls – it can take

some time to rerate all of them.

You can reach Rerating in: SETTINGS –> Billing –> CDR –> Rerating:

Select date interval and user which calls you want to rerate and press Proceed:

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Here you see call stats for selected date interval. Also you can save old data in PDF or CSV

formats. When you are sure everything is correct – press Rerate:

Here you will see the Difference after Rerating.

Rerate all users

Select 'All Users' from User selection to rerate all users at once in selected period.

Testing mode

You can check Testing mode and select any Tariff for user.

This way Rerating will not save changes to calls/users.

Testing mode is useful for testing "What if User had this Tariff" scenarios.

Tariff from Localization

If call's prefix is in Localization Rule which has some Tariff, this Tariff is used to count call's

price instead of User's Tariff.

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Other notes

Re-rate does not take into account flat rates and user will be charged for such calls. Do

not do rerating for such user.

Re-rate does not work when call's DID, LCR or Owner Device is changed by

Localization rules.

Re-rate does not work for reseller users calls and reseller when you try to rerate the

reseller account only. You can re-rate one user at a time, for example if the reseller

made calls through his/hers account which is created by admin.

Online registration

This function lets a user register online. It can be accessed in MOR's entry page by clicking on:

If you are new user you can signup here:

Registration should be enabled in Settings in order to work. Various settings are also set in

Settings -> Registration.

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In the following window, the user should enter his details. Fields marked by * are mandatory:

Note: if phone number is entered, it will be automatically added as CLI for user device.

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After clicking on Sign up, the user is registered, his account is created and he will see

connection info on his screen:

If email sending is enabled (in Settings) and the email SMTP server details are valid, the user

will receive an mail containing all the settings. The administrator will also get an email with

information about the user who has registered with the system. These two settings should

also be enabled.

If you have set every setting to send email to Admin when a new user registers, but no email

arrives - check whether you have Email set for Admin user.

See also

Registration Configuratin

ReCAPTCHA

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Online registration customization

We often get requests for information about how to customize the Online registration form.

Unfortunately, it is currently not possible to customize registration.

See also

Online registration

Registration Configuration

Quick Forwards

About

Some DIDs can be assigned to Quick Forwards functionality. When this is done, each user can

enter a different number to be forwarded when he calls a DID.

The main advantage of this function is to use a limited number of DIDs to allow different

users to call these DIDs and be forwarded to different numbers (which are assigned by

users).

Example

We have several DIDs assigned to Quick Forwards functionality:

1111111111.

2222222222.

We have two users with these CallerIDs:

User A with CallerID 8888888888.

User B with CallerID 9999999999.

User A sets his forwards:

From DID 1111111111 forward to 3333333333.

From DID 2222222222 forward to 4444444444.

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User B sets his forwards:

From DID 1111111111 forward to 5555555555.

From DID 2222222222 forward to 6666666666.

When User A dials DID 1111111111, his CallerID (8888888888) is checked, the user is

recognized, and the call is automatically forwarded to number 3333333333.

When User A dials DID 2222222222, his CallerID (8888888888) is checked, the user is

recognized, and the call is automatically forwarded to number 4444444444.

When User B dials DID 1111111111, his CallerID (9999999999) is checked, the user is

recognized, and the call is automatically forwarded to number 5555555555.

When User B dials DID 2222222222, his CallerID (9999999999) is checked, the user is

recognized, and the call is automatically forwarded to number 6666666666

IMPORTANT: Correct CallerIDs should be set for users in order for them to be recognized

them. Only the system administrator can do this.

If the system doesn't recognize the caller, it asks for a PIN number of the device to be

entered.

If a user is dialing a DID that has not been assigned a Forward Number, the user is

asked to enter such a number.

The user can manipulate his Quick Forwards in his menu. He needs to log in to GUI to change

numbers.

Quick forwards based on diversion number

By default quick forwards are based on CallerID but you can change it to be based on user's

diversion number. But diversion field can be used instead of CallerID. It comes in a SIP packet

and points at the device which forwarded a call before it got into a system. for example, if you

put a forward to system's DID on your cellphone and X calls you by default system will catch

Xs CallerID, which is not in the system, so it will ask for PIN. Meanwhile if quick forwards are

based on diversion system will see that the call belongs to you and it will act respectively.

Moreover, Default Device allows not to bill the user who calls to that DID but bill an other

user instead. To see why it is useful here's another example: let's assume that calling to DID

costs the same as calling to a premium rate line. When you call to such a line you pay at once.

MORs owner gets payment from DIDs provider (which itself gets it from the user). Quick

forwards based on diversion allows creating such services where user does not have to pay

for anything, i.e. forwards for him are "free" because he pays for a DID.

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To make such a dial plan go to SETTINGS -> Billing -> DIDs -> Dial Plans. Click New dial

plan, you have to create a Quick Forward dial plan:

Select Use Diversion instead of CallerID and click .

Tell Balance/Tell Time

By default calls via Quick forward DID will not play Balance and Time. If you want it to go to

SETTINGS -> Billing -> DIDs -> Quickforwards -> Quickforward Settings. You will see this

window:

Select Tell Balance: and/or Tell Time: and click Change.

Tell Time/Tell Balance options also can be found in device's edit window:

Here you can also select Global. Value (Yes or No) in the brackets depends on what is selected

in quickforward settings. In this example there is Global (No) to both options because in

quickforward settings both Tell Time and Tell Balance are set to No.

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Why is the Quick Forwards window empty?

The window may be empty because one or more DIDs should be assigned to Quick Forwards

before calls can be forwarded.

See also

DID Management

Quickforwards rules

About

Quickforwards rules are used to restrict users which DIDs they can use for quickforwards.

Quickforwards rules can be created by:

Admin - can create rules for any DID range assigned to quickforwards DialPlan

Reseller - can create rules only for DIDs which are assigned for him with

a Quickforwards rule by admin

Reseller PRO - can create rules only for DIDs which are assigned for him with

a Quickforwards rule by admin

Quickforwards DIDs can be used by:

All system users if they have a rule assigned for them which matches any quickforwards DIDs.

Admin also must have a rule assigned for him to use a quickforwards functionality.

Read more about quickforwards Quick forwards wiki page.

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How to create a quickforwards rule?

1. Go to SETTINGS -> Billing -> DIDs -> Quickforwards Rules:

2. Press the marked button

3. Fill in the required fields.

How to assign a quickforwards rule to a user?

Go to user details and assign a rule you created:

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Transfers with MOR

Transfers by default are not available in MOR because Asterisk cannot guarantee correct

CDRs with Transfers.

In other words - it is possible to dial for free using Transfers with Asterisk (MOR).

If you really want transfers and do not care about possible loss, here is the way to enable

them:

In file /etc/asterisk/sip.conf

change line

allowtransfer = no

to

allowtransfer = yes

And reload Asterisk. Use it at your own risk!

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GUI on Mobile Devices

Description

If you access MOR GUI over your mobile phone or similar device with a reduced screen, you

will see something like this:

The mini option is used currently only if you want to send an SMS from the MOR interface as

shown here: SMS Addon iPhone.

If you want to get the full MOR interface, select the full option.

If you do not have an SMS Addon installed, the mini option is no of use to you, so you

should always use full option.

See also

SMS Addon iPhone

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PBX connection to MOR

Common connection diagram

Any IP/PSTN-capable PBX can be connected to MOR:

To simplify the diagram, this is the common way for a PBX to be connected to MOR:

Here the main principles will be explained. For a detailed example, see the Trixbox section.

In this diagram, one PBX is connected to the MOR Billing system. Two phones (Phone 1 and

Phone 2) are connected to the PBX. Now both phones can dial out to the PSTN through PBX

and MOR.

Dialing from PBX

Our task is to bill each phone separately.

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In MOR we need to create one User/Device for PBX.

We need to create a Device for this PBX to describe how it is connected to MOR.

You should mark this device as Trunk with ANI.

To bill our devices separately we need:

To create one virtual device for each real device connected to PBX.

In the device's CLI menu we need to add CallerIDs of these devices as described in

PBX. Using the CallerID of the device, MOR recognizes who is calling and assigns the

call to the correct device.

Dialing to PBX

DIDs should be forwarded to the PBX for a real device to get the call.

In order to assign a DID to a virtual device, the following steps are required:

1. Mark the virtual device as Trunk.

2. Assign a DID to the virtual device.

3. Go to the device's Call Flow.

4. In the Before Call state, forward the call to the PBX device.

That way, when a call goes to the virtual device through a DID, it should go through PBX

Trunk.

If you do not forward the call to the PBX device, the call will fail.

This lets the user log in to GUI and see all incoming calls to his device.

ATTENTION! In order for the virtual device to receive calls from the DID, the PBX

should route calls to the correct device. MOR does not know the real

location/connection type of the virtual device, it just sends the call to the PBX, and the

PBX should send the call to the correct device. MOR CAN'T influence the PBX's

configuration, so the PBX should be configured separately to handle incoming calls

from the DIDs.

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MOR and Trixbox

MOR and Trixbox (FreePBX) can't work nicely on the same server, because they use same

Asterisk .conf files, which makes a lot of mess.

In order to integrate Trixbox with MOR you need to connect in the following way:

Connection diagram

As an example, we will configure one phone dialing. The phone is connected to Trixbox,

which is connected to MOR:

Here, for simplicity, all the devices are in the same LAN:

MOR server has IP: 10.10.1.10.

Trixbox: 10.10.1.20.

Phone: 10.10.1.30.

MOR and Trixbox are connected with the following settings:

username: mor_trixbox.

password: mor_trixbox_psw.

The phone to Trixbox is connected with:

username: 201.

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password: 201psw.

Dialing from the phone to MOR

Here the device makes a call to example destination 9999999999.

When the call reaches Trixbox, the correct CallerID number for this phone is applied. In

our example it is 1111111111. Then the call is forwarded to MOR.

When MOR gets the call, it sees that the call was sent by Trixbox.

MOR sees that Trixbox is Trunk with ANI. This means that MOR needs to check the

CallerID of the call to find the real caller.

MOR checks for the CallerID number (1111111111) and sees that this CallerID is

assigned to a phone device.

MOR retrieves all info (LCR/Tariff/Rates/Providers/etc) that is necessary to send the call

further.

When the call is finished, it is logged in MOR, not for Trixbox, but for the phone's

user/device. That is what we wanted from our configuration.

Trixbox configuration for phone

Let's connect a phone to Trixbox. In FreePBX, create an extension for phone 201:

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The most important details are:

Here we set the CallerID number that should be applied to the call from this device. The

password and dial string are also here.

Trixbox configuration for MOR

In Trixbox we need to define MOR as a Trunk:

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Define a route in Trixbox to use this Trunk (this is just an example):

Trixbox Trunk Troubleshooting

When passing multiple CallerIDs from multiple extensions in TrixBox through a single ANI-

enabled trunk to MOR, be sure to define the outbound TrixBox PEER CallerID as empty, using

callerid="" <>.

If you don't include this, your MOR trunk will publish whichever CallerID was last passed to it.

This will not allow you to correctly track different users' CallerIDs in MOR to provide them

with separate call cost tracking.

Also, be sure NOT to define the inbound USER CallerID on the same TrixBox trunk as the

same empty entry. If you do, the CallerID will be stripped from all incoming calls.

MOR configuration for Trixbox

We need to create a connection between MOR and Trixbox. In this example, we will configure

the MOR-Trixbox connection by using SIP.

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We need to create a user for Trixbox: pbx_user. Put any details for it - it does not matter

which:

Now we need to configure the device for Trixbox:

The main settings are:

Mark this device as Trunk with ANI support.

Username (by example): mor_trixbox.

Password: mor_trixbox_psw.

Host: 10.10.1.20.

Make sure Dynamic? is unchecked and the port is 5060 (for SIP).

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MOR configuration for phone

To correctly bill phone calls, you need to create a separate user and device for it on MOR.

First, create a user for the phone (phone_user):

Make sure you configure this user correctly by entering the password/LCR/Tariff and other

necessary details. This user will be used for real dialing out.

Next, create a virtual device for the phone:

The settings for this device are not at all important, but you should:

1. Make sure you have marked this device as Trunk. This is important for receiving calls.

2. Make sure you have a good password for security.

3. Make sure that this device's host = dynamic (for security reasons, you cannot enter the

Trixbox IP here, because 201 is not Trixbox!)

CallerID for phone in MOR

In order for MOR to recognize a device/phone, we need to assign a CallerID to our phone's

device:

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NOTE: currently it is not possible to enter a CallerID that is the same as the name of the

device. That way, the call will fail. This happens because of the way Asterisk handles

authentication.

Dialing to phone using DID

Here a call is coming to DID 2222222222.

When it reaches MOR, MOR knows that 2222222222 should be forwarded to device

201.

Device 201 forwards the call to Trixbox device Dial (SIP/22222222222@mor_trixbox).

Trixbox receives call with destination 2222222222.

Trixbox know that this call should be forwarded to phone SIP/201.

Phone rings.

After the call, in the MOR log the call is in device 201's logs (CDR) - as it should be.

MOR configuration

You need to create DID 2222222222 in MOR and assign the phone's user/device to this DID:

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In this device's Call Flow, you need to forward the call to Trixbox:

Trixbox configuration

Now when the call arrives in Trixbox, it should know how to route the call to phone 201.

We can configure this in two ways:

Using Direct DID.

Using the Inbound Route.

Direct DID configuration

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Inbound Route configuration

Final notes

In the example, we used DID 2222222222 and CallerID 1111111111. In real installations, these

values are the same in most cases. This means that if the user is dialing out and his CallerID is

(for example) 3333333333, he can receive calls by DID 3333333333. You can configure this in

the same way using this manual. Different values for DID and CallerID are chosen here to

make the example easier to understand.

Configuring more phones

In order to configure more phones:

1. Connect a phone to Trixbox.

2. Configure user/device in MOR for the phone.

3. Make sure the phone's CallerID in Trixbox is the same as CallerID (CID) for the phone's

device in MOR.

4. Create a DID for the phone on MOR, assign it to the phone's device, and forward calls

from the device to Trixbox in MOR.

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5. Make sure Trixbox knows how to route calls from DID to the phone.

Limitations

Using this configuration to connect Trixbox (actually, FreePBX) to MOR has some limitations:

1. The MOR device's CallerID can't be the same as the device's name.

2. Phones connected to Trixbox can't have similar CallerIDs. (Actually they can, but MOR

will not be able to distinguish which device is making the call, so these devices with

identical CallerIDs will be billed together using one account (user/device) in MOR).

Errors

chan_sip.c:8373 check_auth: username mismatch, have <201>, digest has

<mor_trixbox>

This error may occur when configuring the connection between MOR and Trixbox.

--note--

This is not a connection problem. This happens when you have an incorrect (or no) outbound

CallerID set within Trixbox. Thus "201" (the display name) is being passed through to MOR.

--VCCS

MOR API

About

MOR API lets other software interact with MOR without knowing the internal structure of the

database (DB) and other logic.

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MOR accepts requests by HTML POST and returns results in XML.

MOR users that are allowed to use MOR API

Admin

Reseller

Accountant

Functions

API Related

API Configuration - configuration options for API.

Hash construction - how to construct hash to authenticate data over API.

Login

login - logs user in to MOR.

logout - logs user out from MOR

.

User

user_details - gets user details.

user_register - create new user from params.

user_update_api - update user from params.

Callback

callback - initiates callback.

c2c_callback - initiates Click2Call specific callback.

Accounting

invoices - retrieves a list of invoices in a selected time period.

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Calls

user_calls - retrieves a list of calls in a selected time period for a user/device.

Balance

balance - retrieves a user balance.

simple_balance - gets user's balance

change_user_balance - change user's balance

Rates

Rate - check user rate by prefix

get_tariff - get all tariff's rates

import_tariff_retail - import xml retail tariff

wholesale_tariff - update or create wholesale tariff

Device

device_create - create device

delete_device - delete device

Phonebooks

phonebooks_api - phonebooks list.

phonebook_edit_api - edit phonebook.

Monitorings

ma_activate - turn on monitorings addon

Payments list

payments_list - payments list.

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Credit notes

credit_notes - credit notes list.

credit_note_update - credit note update.

credit_note_create - credit note create.

credit_note_delete - credit note delete.

Financial statements

financial_statements - financial statements list.

Payments

create_payment - create payment.

Calling cards

cc_by_cli - creats and update calling cards, including adding and transferring funds, by

being identified according the Caller_ID.

buy_card_from_callingroup - solds cards from calling card group.

show_calling_card_group - Show calling group info.

MOR version

get_version - get MOR version number.

Send SMS

send_sms - send SMS.

HTML POST

All requests are in a similar format:

HTML POST: http://<SERVER_IP>/billing/api/<function>?u=username&p=password

username/password - the user name and password of the user who is asking for the

data.

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If the user has 'admin' rights, he will get more results and can perform more actions

compared to a user who does not have such rights.

API Configuration

Go to SETTINGS –> Setup –> Settings –> API tab

Allow API - should we enable API?

Allow GET API - should we allow GET method to be used for API? Good for testing -

but do not use in production! It is unsafe.

API Secret Key - key to generate Hash

XML API Extension - send XML output with <HTML> tag - this is used for broken .Net

support, keep it unchecked if everything works for you

Allow API Login Redirect - if login is successful the customer will be redirected to

MOR billing Main site.

Uniquehash - admin's unique hash. Reseller can see it by going to SETTINGS -

> Registration.

Please note that MOR does not allow you to enable Allow API if API Secret Key is not

entered. In such case you will get a message Invalid API secret key. Enter API Secret

Key in order to fix this.

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Cron actions

What is cron actions

Cron actions is MOR functionality which allows you to do some tasks at a specified time.

Currently supported actions are these:

Change tariff (for users)

Change provider tariff (for providers)

Who can use this functionality

Currently only system admin is allowed yo use this functionality.

Important notes on billing

MOR takes the call price before the call and uses it to calculate the final price after the call is

finished.

Example: User starts a call at 9:58 PM and is talking for 5 minutes, you have an action which

changes tariff at 10:00 PM. When call will be finished MOR will bill user using older tariff. If

user starts a call after tariff change - he will be charged using the new tariff assigned for him.

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How to create a cron action?

Settings are self explaining here is an example with an action Change tariff (applied for users)

1. Go to SETTINGS -> Setup -> Settings -> Cron actions:

2. Set the settings:

Name - enter here something that will help you to remember what this settings is about

Action - what to do. Currently only change_tariff is supported.

User - for which user/users action has to be applied

Tariff - which tariff to set

Periodic type - how often to perform the task

Repeat forever - makes action never stop

From - when to start doing the action

Till - when to stop doing the action

Description - you can use this field for your notes

3. Press Create to create the action.

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4. Finished, you will be taken to a list of all defined actions:

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MOR Addons

Calling Cards Addon

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From Wikipedia, the free encyclopedia

A telephone card, calling card or phone card for short, is a small card, usually resembling a

credit card, used to pay for telephone services. Such cards can either employ prepaid credit

system or credit card style system of credit. The exact system for payment, and the way in

which the card is used to place a telephone call, depend on the overall telecommunication

system. Currently, the most common types of telephone cards involve pre-paid credit in

which the card is purchased with a specific balance, from which the cost of calls made is

deducted. Pre-paid phone cards are disposable. When the balance is exhausted you simply

buy a new card. Cards purchased can often be refilled. The other main type of card involves a

card with a special PIN printed on it that allows one to charge calls to a land-line telephone

account.

Short description

The Calling Card module makes it possible to manage cards printed on plastic/paper or sold

over the Internet for international calling.

The calling card platform is dedicated to calling cards, i.e. the plastic/paper cards with a

scratch surface that can be sold in shops or in streets. These cards (or their number/PIN) can

also be sold over the Internet.

When a user buys a card, he gets the number/PIN of the card. Then he dials an access

number and IVR asks him to enter this number/PIN.

The user hears the credit and time remaining and can enter a destination. If the destination is

okay, he is connected and can talk. After the call is finished, the call's price is deducted from

the card's balance.

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Differences between Users and Cards in MOR (for A2billing users)

There are two main ways for a person to use the MOR system to dial out:

1. Using his User account.

2. Using a Calling Card.

The main difference is that the User account is for permanent users. Calling Cards are used by

occasional dialers who use the service from time to time and often anonymously.

Rule about Users and Cards:

Users and cards are not related in any way. This means:

A User (as an entity in MOR) can't have Cards.

A Card (as an entity in MOR) can't belong to a User.

NOTE: MOR is not A2Billing. So the Calling Cards Addon is ONLY for Calling Cards which are

sold over the Internet/in the streets/shops/etc.

System Setup

1. First, a Card Group should be created with the correct details.

2. Cards for the Card Group should be generated.

3. A Calling Card Dial Plan should be created.

4. A DID should be assigned to this Dial Plan.

All these steps are described in detail below.

Work-Flow

1. User buys a Calling Card (over the Internet/printed, in a store/street/etc).

2. User finds the NUMBER and PIN of the Card. It can be on the plastic/paper Card or

given to the user in an online Calling Card Store.

3. User dials an access number, which is given to the user in the same way as the

number/PIN. User can dial this number from anywhere in the world.

4. System asks the user to enter the number and PIN (or only the PIN). When user dials

the access number, IVR will ask him to enter card number and/or PIN.

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5. Are the number and/or PIN good? In this step the system checks the database to see

whether a Card with the provided number/PIN exists. If the Card is not found, the user

is asked to enter new a number/PIN. It is possible to change how many times the

system will permit reentry of the number/PIN. The default is three tries.

6. The system tells the remaining balance on the card. This step is optional and can be

disabled.

7. The system asks for the Destination to be entered. IVR will ask the user to enter the

destination he wishes to dial.

8. The system tells the remaining time. Based on the entered destination and the

remaining balance, the system will tell the user the remaining time he can talk with this

destination. This step is optional. It is influenced by Ghost Minute Percent.

9. The system dials the Destination. Using LCR, the system chooses the cheapest provider

and dials the entered Destination.

10. After the call, its cost is deducted from the Card's balance.

More detailed logic is explained Calling card logic page at wiki.

PINless dialing

It is possible to enable PINless dialing to the Calling Card Dial Plan:

This lets the user enter the Card's number/PIN only once, after which the system remembers

the CallerID. Next time, when the system receives a call from same CallerID, it will not ask the

user to enter the Card's number/PIN.

If the Card's balance is empty, the user will be asked to enter a new Card number/pin.

PINless functionality is only available for a valid CallerID.

When a CallerID is not empty and is numeric and/or has a '+' sign, then it is valid. If CallerID is

invalid (has some non-numeric symbol in it, except '+'), PINless functionality will be disabled.

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Can more than one Caller ID use Pinless Calling Card?

Yes, it is possible. Suppose we have two Caller IDs:

CallerID #1

CallerID #2

Then:

1. CallerID #1 dials DID number, MOR asks to enter PIN number.

2. CallerID #1 enters PIN number 1234. MOR assigns this CallerID #1 to Calling Card with

PIN 1234.

3. If CallerID #1 calls once more, it is not asked to enter a PIN because MOR has assigned

it to Calling Card with PIN 1234.

4. Now CallerID #2 dials DID number, MOR asks to enter PIN number.

5. CallerID #2 enters PIN number 1234. Now MOR deletes CallerID #1 and assigns

CallerID #2 to Calling Card with PIN 1234.

6. If CallerID #2 calls once more, it is not asked to enter a PIN because MOR has assigned

it to Calling Card with PIN 1234.

7. If CallerID #1 calls once more, it is asked to enter PIN because Calling Card with PIN

1234 was assigned to CallerID #2.

8. Etc.

End IVR Type

Description

This option allows to select which End IVR to use for Calling Card Dial-Plan.

End IVR #1

This is default IVR.

When call ends or caller presses * to terminate call, he comes into this IVR.

Here he hears message: "Press 1 to dial new destination, press 2 to dial same destination,

press 3 to check your balance, press 4 to use new card"

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When user presses some button [1,2,3,4] he will be able to do such actions:

1 - Can enter new destination

2 - Can dial same old destination (redial)

3 - Can hear his balance

4 - Can enter new PIN to use new Card

End IVR #2

This type of IVR allows to enter new Destination or redial old one by pressing 1#.

User can enter destination and # to dial to it, o he can enter 1# and then system will redial

old destination.

End IVR #3

This IVR is just a hangup. E.g. call will end at once using this IVR and no further actions will be

possible.

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Calling Card Groups

A Card Group is a set (array) of Calling Cards.

Properties

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Card Image – just for informational purposes and eye pleasure to view on the GUI.

Can be (and should be) the same as the real printed card for consistency. But that's not

the rule.

Description – a short memo for admin to view some more info for the cards. It is

useful when there are a lot of Card Groups and the Card Group name is not enough.

Price with TAX – the price which all Cards in this Group will have before first use. It

does not include VAT. It's in the default currency.

Setup Fee – the fee which will be charged when a call is connected the FIRST time

when the card is used.

Ghost minute percent – a value to trick the user into thinking he has more minutes

when he really has less.

Example:

A user has enough balance in his card to call for 50 minutes. Ghost minute percent = 120. So

when the user enters his number/PIN, the system tells: "You have 60 minutes remaining"

(50/100 * 120). Note: this is not an increase in percent; it is the value by which the actual

minutes should be multiplied to tell them to the user.

Ghost balance percent - same as Ghost Minute Percent, but this value only applies to

the Balance which is told to the user (This option is available starting from MOR9).

Daily Charge – the amount which will be deducted EACH day from the card starting

from the date when the card is used for the first time. This means, if the user makes his

first call on Monday and his second call on Friday, his card will be deducted for 5 days

using this value (Mo, Tu, We, Th, Fr).

Currency - default currency for all cards

Tell cents - enable or disable telling cents when announcing card balance (This option

is available starting from MOR 10).

Tell balance in currency - choose in which currency to tell balance when announcing

it. (This option is available starting from MOR10).

TAX percent – the actual value which will be used when presenting the Card's price to

the User, i.e. when telling the user his remaining balance on the Card or showing the

same information on the screen.

Least Cost Routing (LCR) – which LCR will be used for all Cards in this Card Group.

Tariff – the price list which will be used for this Card Group.

Location – this is useful when Cards will be sold in a region with specific dialing rules.

Rules for location can vary for different Calling Card Groups.

Number Length – the length of the Card's number. You can select any value here. It is

recommended to select different values for different Card Groups to minimize the risk

of anyone guessing another card's number and using it unauthorized. Card numbers

are generated in sequence order.

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PIN Length – the number of the Card's PIN. The same recommendation applies as for

the Card's Number. PINs are generated randomly.

Valid from - from which date Cards are valid.

Valid till - till which date Cards are valid.

Valid after first use - how many days a Card is valid after first use. Default 0 means

that card is valid forever.

Allow loss calls - enable this feature on calling card group if you want to let card users

to make calls when there might be loss.

Deny to double-use a Card with Pinless dialing, when Card is assigned to some

CallerID

Disable Voucher with same Number as Card if Card is used? - this option allows

you to print cards which can be used both for vouchers and for Calling Cards. If one is

used - the other one is deactivated.

Management

Card Groups, like many other entities in MOR, can be created/edited/deleted/viewed.

If a Calling Card module is enabled, you can find it (as Administrator) in: ADDONS –> Calling

Cards.

Here you can review your active Calling Card Groups:

(..) - shows the Cards window, where you can manage the Group's Cards.

- all the details of the Calling Card Group.

- exports Cards into CSV format.

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- edits the Group's details.

- deletes the Group.

Calling Card management

Overview

In this window, you can manage all Cards which belong to the Calling card group. All fields

are self-explanatory.

Card Creation

Cards for the Calling Card Group can be generated or imported from CSV file.

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Generating Cards

Starting from MOR 10 MOR allows you to generate only 20% cards of total possible unique

PIN number count. For example: you have defined that your card group will have 2 digits

length PIN, that means that there is 10*10=100 possible unique PIN combinations. So MOR

allows you to create: 100*0.2 = 20 cards. To generate more cards you have to increase PIN

length.

Add cards – opens the window where you can add additional Cards to the Card Group:

In this window, you should enter the Card's start and end numbers. They will form an interval

which will be created. If any Card of the same number exists, the old Card will remain and a

new one will not be created.

The card's PIN number is created automatically – it's random and unique. It is wise to make it

reasonably long to avoid cheating by users who may guess a short PIN number.

Export Cards to CSV file

When you want to print Cards on plastic/paper, you need to provide the necessary

information for the Card Printing service.

This can be done using a CSV file. Clicking on the icon lets you download all Card

information in one file.

You can process this file later in MS Excel or other compatible software. Use the correct

format for your Card manufacturer and send it to him.

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The CSV file fields are:

Number

PIN

Balance

Sold

First_use

Daily_charge_paid_till

You can delete unnecessary fields and make it look as you need.

Card batch management

To manage cards in groups, you need to use Batch Management:

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Minimal balance

Minimal balance is the lowest card's balance when it is still available to make calls from that

card. For example if you set 10 as minimal balance if card's balance is lower than 10 no calls

will be made from that card. You can find this functionality in main calling card's edit window:

Calling cards distributor

A distributor is a user who can manage admins cards. Please note that distributor is not an

owner.

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To assign distributor to a card go to ADDONS -> Calling Cards. Open any groups cards list

and click Edit. You will see this window:

Select the user from a dropdown list and click Edit.

Assigned distributor will see his cards here: PERSONAL MENU -> Various -> Calling Cards.

Distributor can only activate or deactivate calling cards by clicking (deactivate) or

(activate). Above is the calling cards window that distributor sees.

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Disabled cards

Disabled calling cards are those which you can't call from. The reasons for disability are:

Card is not sold

Cards balance equals to zero

Card is not valid anymore.

Calling Card Dial Plan

Overview

Once we have generated a Calling Card Group and Cards, we can create a Dial Plan for this

Group.

A Dial Plan can be treated as a set of rules for how to handle a particular Calling Card Group.

You can set different rules for the same Calling Card Group.

The Dial Plan gives you more control over Calling Card Groups.

Dial Plan creation

Let's go to SETTINGS –> Billing –> DIDs –> Dial Plans:

In this image, we see an existing Dial Plan for a Calling Card Group.

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Let's review each field:

Name – just for informational purposes.

Number length – tells Interactive Voice Response (IVR) how many digits to wait for in

NUMBER following a request to enter a number.

PIN length – how many digits to wait for in a PIN.

Ask only PIN – as the PIN is unique, it is sometimes wise to ask only for the PIN,

because users can make mistakes and get annoyed about having to enter too many

digits.

Tell time – should IVR tell the time after the Destination is entered?

Tell balance – should IVR tell the balance after the user enters the NUMBER/PIN?

Ask Number/PIN (times) – how many times the user can try to enter Number/PIN.

Ask destination (times) – how many times the user can try to enter Destination.

End IVR – End IVR type.

When creating a new Dial Plan, you will need to enter the correct Number and PIN length

values of an existing Calling Card Group. Otherwise you will not be able to create the Dial

Plan.

Assign DID to a Calling Card Group Dial Plan

When we have created a Dial Plan for a Calling Card Group, we can assign one or more DIDs

to this Dial Plan.

A DID (in other words, an Access Number) is vital for using Calling Cards. The user will use

this number to reach IVR.

You can assign unlimited DIDs to the same Dial Plan, but a particular DID can only be

assigned to one Dial Plan.

The Calling Card Dial Plan is useful when you are selling Calling Cards in different regions,

such as different countries. In country A, the user will dial one number (which is cheaper for

him or even free), and in country B the user will dial another number. They will reach the

same IVR and will use the same service, but they will save on telephone fees.

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In SETTINGS –> Billing –> DIDs:

We click on the icon for the first DID:

We select our Dial Plan and click the confirmation icon to assign our Dial Plan to this DID:

Now the user can dial number 12026911622 (in the example) and he will be able to reach IVR

and use his Calling Card.

Online Calling Card Shop

MOR has a simple Calling Card shop for demo purposes where you can sell Calling Cards.

It can be reached by the same IP as MOR, just using 'ccshop' instead of 'callc'.

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For example, http://<domain/ip>/billing will lead to MOR, while

http://<domain/ip>/billing/ccshop will open the Calling Card Online Shop:

All Card Groups with cards which can be sold are shown here with their images.

If we want to buy one card, we enter the amount and click the Add to cart button for that

Card:

The shopping cart shows what we have selected to purchase. We can continue shopping,

empty the cart, or proceed to the checkout.

Let's go directly to the checkout, because all the other functions are very simple:

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Clicking on the PayPal button, you will be directed to the PayPal site, where you will be able

to pay for the Card.

Calling Card aggregate

What is Calling Card aggregate?

Aggregate reports from all or selected card groups

Where can I find Calling Card aggregate?

Go to ADDONS -> Calling Cards -> Cardgroup Aggregate

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Speed Dials for Calling Cards

When a Calling Card owner logs in to the Calling Card portal (/billing/ccshop), he is able to

manage his Speed Dials:

Here the Calling Card owner created a new Speed Dial: '11'. When he dials '11', the system

will dial to '37063042438'.

Callback and Calling Cards

Negative balance for Calling Cards

A calling card balance will go negative in only one case:

1. A user initiates a callback, and the system dials him back.

2. The user answers and enters a calling card number/PIN.

3. The system recognizes it and checks its card balance.

4. The Card balance is 0.

5. As the calling card is recognized, the charge for Call Leg A is deducted from the Calling

Card balance. The Calling Card balance becomes negative.

This means that the system owner incurs a loss. However, there is no way of preventing a user

from initiating such a call and not paying for it.

Card import from CSV file

It is possible to import calling cards from a CSV file.

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The format is simple:

CARDNUMBER,CARDPIN

When importing, make sure your CARDNUMBER length and CARDPIN length match the

lengths in the Calling Card Group.

Example. You are trying to import this file:

1000000001,6258

1000000002,5725

1000000003,9737

1000000004,9694

....

Your Calling Card number length is 10 and your PIN length is 4. So you can only import cards

into the Card Group which have the following values:

Otherwise you will get an error.

You can find a video which demonstrates Card import from a CSV file here

[http://wiki.kolmisoft.com/index.php/Card_import_from_CSV_file]

One million Calling Cards

One customer was interested in the possibility of having one million Calling Cards, so we

created this possibility.

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The cards were generated on VMWare virtual machine. The host was an Intel Duo Core2

2.4ghz/3Gb RAM Laptop.

It took ~2h.

phpMyAdmin:

MOR GUI:

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Questions about Calling Cards

Q. When MOR asks for CC Number and PIN number, it asks one to enter NUMBER +

PIN + #. Is it possible for it to ask for NUMBER+# and then PIN+# (not only the PIN)?

A. No, it is not possible. It is only possible to ask for the PIN without the number: PIN + #.

Q. Can I modify the first page of the CCSHOP to include some other information or

links to the rate list for each card, for example?

A. Ruby On Rails knowledge is necessary to do that. Also, custom changes will be overwritten

by the next system update.

Q. Can Resellers have their own Calling Cards?

A. Yes if Reseller and Calling Card addons are installed and a reseller permission group is

assigned to a reseller as described in Reseller permissions wiki page.

Q. Where do we see Payments for Calling Cards when we recharge them over web?

A. In SETTINGS - Accounting - Payments.

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Q. Can I create calling cards in different currency?

A. You cannot create cards in different currency. And there is no need to do it. Create Cards in

USD, and in Dial Plan select Currency which will be used for such Cards. Users will hear

EUR/GBP and will not know that in DB card's balance is saved in USD.

How to change default button for call end

Description

By default, the button to end the current _CONNECTED_ call when using Calling Cards and

ANI/PIN dialing is '*'.

Sometimes it is necessary to change it to something different.

As an example, we will change it to '##'.

Configuration

In file /etc/asterisk/features.conf find lines:

[featuremap]

;blindxfer => #1 ; Blind transfer (default is #)

;disconnect => *0 ; Disconnect (default is *)

Change them to:

[featuremap]

blindxfer => #1 ; Blind transfer (default is #)

disconnect => ## ; Disconnect (default is *)

Restart Asterisk

REMEMBER: this option is only applicable when using a Calling Card or ANI/PIN dialing when

the call is already CONNECTED.

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How to set language for Calling Cards

Description

The language in which Calling Card is played can be set in several places in the following

order of priority:

1. Default (en).

2. IVR.

3. DID.

Default language

If Calling Card does not use IVR and the DID language is set to empty, the default language

will be used.

This image shows how to set to "use default language" (when IVR is not used and DID

Language is set to empty):

Language set by IVR

If Calling Card Dial Plan (DP) is using IVR, then the language set in IVR overwrites Default

Language and is used IF DID Language is empty.

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This picture shows how Language (Voice) is set in IVR:

This example changes Language to Spanish (es).

Once again: it overwrites Default Language (en) only when DID Language is empty!

Language set by DID

And most important is DID Language, which can be set in user DID:

This example shows ru language which will be used no matter what Default Language is and

no matter which Language (Voice) is set in IVR.

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Examples

Language/Voice

Default empty empty empty empty en en en en

IVR empty empty es es empty empty es es

DID empty ru empty ru empty ru empty ru

Which one will be used en ru es ru en ru es ru

NOTES:

1. When Default is 'empty' it means that sip.conf file is messed up and is set to language

= .

2. When IVR is 'empty' it means IVR is used but no Language is changed in it, OR IVR is

not used at all.

3. When DID is 'empty' it means the Language field in DID configuration is empty.

Calling Cards Payments

What is a Calling Cards Payments list?

Calling Cards Payments list lets you to see your calling cards payments in one place for each

card.

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How to make a Calling Card payment?

You have to mark Calling card as sold:

1. Go to Calling Card group list (Addons -> Calling Cards -> Cards) and press one of the

marked icons (later in this example always press the same icon):

2. A dialog appears, press the Buy button:

3. A confirmation dialog appears, press Confirm:

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Now having your card sold make a payment for it:

1. Press the same buy icon again:

2. Now enter the amount you want to add to the current card balance

3. Confirm the payment

How to get a list of payments for a specific card?

1. Press on the marked icon and you will get into the payments list for that Calling Card

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2. Calling Card payment list:

Calling Cards Ghost Minutes External Script

This topic is ADVANCED and shows how to MOD MOR. Please do not use this

functionality if you are not sure what you are doing!

What is it for?

This functionality is designed to allow clients to modify the way how MOR Core behaves with

Calling Cards.

MOR itself has some logic which allows to change this from the GUI.

But very often it is necessary to extend or change such behavior.

This functionality is made as external script in which clients (with enough programming skills)

can define the way how MOR reacts to some situations.

With External Script it is possible:

Adjust Time which is told to the Card User

Adjust Balance which is told to the Card User

Adjust real call duration of the Card User

With this script it is possible to achieve various functionality, as example some of them:

Taxation per call

Connection Fee per call

Disconnection Fee per call

Multiple in-progress fees

Post-call surcharges

Adjust call duration surcharges

Random surcharges

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Maintenance/Service charges /Recurring configuration that are applied by system at fix

interval configured in days

Toll Free Surcharge in case of call is from the TOLL FREE number (pass alone to reseller

in this case owner of TF and a calling card owner a.k.a person who dial TF).

Custom rounding increments

How to turn it on?

By default it is OFF:

Turn it ON:

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NOTE: When External Script is ON, Ghost Minutes and Ghost Balance

functionality is NOT used from the GUI. It is defined in External Script.

How to use it?

File which does all the logic is in /usr/src/mor/agi/mor_cc_external.c

It is written in C with a lot of commentaries which explains all the logic INSIDE THE FILE.

Same script can be rewritten in any other programming language by experienced

programmer which knows how to interact with Asterisk over AGI.

Make changes to this file, execute ./install.sh in same folder to compile and move it to correct

place.

How to know it is working?

Make a call using Calling Cards with enabled Tell Time/Balance and watch Asterisk CLI:

This shows the output BEFORE telling the balance to the Card User.

NOTE: default External Script DOES NOT change any values, it is just for demo.

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This shows the output BEFORE telling the time to the Card User.

NOTE: default External Script DOES NOT change any values, it is just for demo.

IMPORTANT: To make sure script works OK, you SHOULD see line:

MOR CC External Script stopped.

Calling Card setup example

Card Group creation

We need to create a Calling Card Group to handle all such Cards.

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We go to SETTINGS –> Addons –> Calling Cards and click New card group. In the new

window, we enter our Card's details:

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After creating a Card Group, we will see all its details:

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Change image for Card Group

Click on and go back to the Group's details and select the image for your card:

Click Upload:

Generate Cards

Now we have a Calling Card Group, we need to generate some Cards. Click Add cards:

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Here we will create ten Cards, starting from 2000000001:

Create a Dial Plan

Now that we have a Calling Card Group and Cards, we need to create rules on how to handle

all these cards.

For that, we need to create a Dial Plan. Let's go to SETTINGS –> Billing –> DIDs –> Dial

Plans and click New Dial Plan.

In the new window, we enter the following details:

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Name is just for informational purposes. Number and PIN length should match the settings

of our newly created Calling Card Group.

We want IVR to ask for the number and PIN, so we leave Ask only PIN unchecked. IVR will tell

the remaining time and balance.

IVR will ask for number/PIN and destination three times at most:

Assign a DID to the Dial Plan

In the last window, we see that we have no DID(s) assigned to our Dial Plan.

We need to go to SETTINGS –> Billing –> DIDs, select some available DIDs and click the

icon:

Here we select our Calling Card Test DP and click the confirmation mark.

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The new window shows the status of our action:

Mark Card as Sold

In order to be able to use Cards, they should be Sold, i.e. ready for usage.

Lets make one Card Sold for testing.

Go back to the Cards list and click the icon near a card:

Click the Buy button:

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... and Confirm the purchase:

Now this Card is marked as Sold and can be used:

Conclusion

That's it. The Calling Card engine is ready to serve calls.

In our example, number 37063042438 is the Access number to be dialed to reach IVR.

When you dial this number (in the example), IVR will ask for the card number/PIN to be

entered and follows all other Calling Card logic.

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Callback Addon

About

This Addon must be purchased separately in order to use a Callback functionality.

Definition

From Wikipedia [http://en.wikipedia.org/wiki/Callback_%28telecommunications%29], the free

encyclopedia

In telecommunications, a callback (also written as call-back) occurs when the originator of a

call is immediately called back in a second call as a response.

In order to use a callback service, a subscriber is allocated a unique number, which must first

be dialled in order to trigger a return call. This is known as a Direct Inbound Dialing (DID)

number (Access Number). Where Caller ID is available, it may be possible to use the same DID

number for many different subscribers, as the callback system will be able to recognize each

subscriber's registered number.

On hearing a ringing tone for a few seconds, the subscriber simply hangs up and awaits the

callback. On receiving this, usually within a few seconds, the customer picks up the phone and

dials the required number.

The cost of making a telephone call via callback consists of two parts, as the caller is

effectively paying for an outbound and an inbound call at the same time. If it costs 20 US

cents a minute to call Nigeria from the US, and 8 cents a minute to call Spain from the US,

then the caller will pay a total of 28 US cents a minute.

NOTE: If you want to use Callback, make sure your DID provider sends correct CallerID

information. Without CallerID, you will not be able to use Callback.

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Common Callback diagram

Implementation in MOR

Callback in MOR is implemented in very simple way. It's enough to tell the system the Access

Number (DID) and the way Callback should act. All of this will be described in following

paragraphs.

Supported Callback methods in MOR, listed by initialization method (which triggers Callback),

are:

Call

Over GUI

Over Email

Over API

Who pays for not authenticated call?

First of all, by the nature of Callback it is important to know that before calling back to the

user, the system does not know who the user is (when not using ANI). For example, it does

not know whether the user is a client, whether he has an account in the system, and so on.

This means that when the system makes the callback to the user and the user picks up, the

call is ANSWERED, but the system still does not know who the user is. Let's say the user

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unsuccessfully tries to enter a PIN several times. Then he hangs up. The call is ANSWERED.

Does the user pay for this call? No. Who pays? You, as the owner of the system. It's sad, but

true.

Callback User, Callback Device and Callback Tariff

In order to keep track of Callback calls, you need to configure MOR in the proper way. For

this we need:

1. Callback User

2. Callback Device

3. Callback Tariff

Callback User is the user who will pay for answered but not authenticated calls to the user.

100% of the time, this user is the system owner. But creating a new user in MOR lets you keep

track of all Callback activity in your system. The other reason to have Callback User (and not

use the System Admin account) is the ability to create and assign a Callback Tariff to this user.

Callback Device is the necessary entity in MOR which will actually 'make' callback calls to

user. The type and other details do not matter. This device is pure virtual.

Callback Tariff – a separate tariff which should be assigned to the Callback User. In this tariff

you enter rates only to those destinations from which you expect the user will make an

Initiation Call. This is important! Make sure you do not have some Inmarsat rates defined

here. Guess how much you will pay if the user initiates a callback from Inmarsat and, when he

gets the call, enters the PIN several times incorrectly and hangs up? In short, this tariff decides

from which destinations you allow users to initiate callback.

Types of Callback in MOR

Callback in MOR can be classified three ways depending on how the user is

authenticated/authorized:

1. By User Device's PIN.

2. Using ANI (Automatic Number Identification).

3. By the Calling Card's Number/PIN (or just the PIN).

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Callback with Device's PIN authentication/authorization

The main part here is that user gets authenticated/authorized when he enters Device PIN.

System checks for such PIN in database and if user is found – user can proceed.

Callback Using ANI (Automatic Number Identification)

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This way of Callback is extended User auth. by PIN Callback. In order to ask user for Device's

PIN system checks user's CallerID (ANI). If such CallerID is found in database – user gets

authenticated/authorized and can proceed entering destination. If CallerID is not found – user

is asked to enter his Device's PIN number.

Attention! It is very easy to fake CallerID so be careful when you are enabling Callback

with ANI. Anybody can put anybodies CallerID and use your service for free.

Callback By Calling Card's Number/PIN (or only PIN)

This type of Callback lets user enter Calling Card's Number/PIN to get

authenticated/authorized. System checks for such Calling Card in database. If call is made – it

is assigned to particular Calling Card.

Rule about Users and Cards:

Users and cards are not related at all. That means:

1. User (as entity in MOR) can't have Cards.

2. Card (as entity in MOR) can't belong to some User.

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Setting Callback in MOR

In order to setup Callback in the MOR you need:

1. Callback User

2. Callback Device

3. Callback Tariff

4. Access Number (DID)

5. Calling Card or Auth. by PIN Dial Plan

The work flow is as follows:

1. Create Callback User

2. Create Callback Device to the Callback User

3. Create Callback Tariff and fill it with rates to destinations from which you want to allow

users to initiate callback

4. Assign Callback Tariff to Callback User

5. Make sure you have Calling Card or Auth. by PIN Dial Plan

6. Create Callback Dial Plan and assign Calling Card or Auth. by PIN Dial-Plan to Callback

Dial Plan

7. Assign Callback Dial Plan to some DID

See also

WEB Callback

Callback over Email

API Callback

Callback setup example

217 Callback not initiated because device not found by ANI

Can't setup callback

Callback and Calling Cards

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WEB Callback

You can find Web Callback in PERSONAL MENU –> Various -> Callback.

The main window lets you enter callback details:

When you enter the source (and destination if you wish), press Execute and you will get a call

to Source. After that, you will be connected to Destination (or asked to enter the destination

and only then connected).

See also

Callback

Callback over Email

Description

This function receives email and initiates callback using details in this email.

A device can have one CallerID from which it is able to initiate Callback.

Admin sets up the email box, which should be monitored for callback execution.

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An SMS to initiate callback can look like:

For CallerID change: change*new_callerid

For callback inititation:

first_number#second_number

first_number - will ask destination

Then SMS gateway should send email to MOR to initiate callback in following format:

CallerID change:

Email subject: change

Email body: old callerid\nnew callerid, (\n) means new line

Callback inititation:

Email subject: callback

Email body:

MOR_PARAM1 auth_callerid

MOR_PARAM2 first_number

MOR_PARAM3 second_number

NOTE: email format should be plain, not html!

Inner mechanics

1. SMS gateway sends email to specified address (example [email protected]) in the

following format:

CallerID change:

Email subject: change

Email body: old callerid\nnew callerid, (\n) means new line

Callback inititation:

Email subject: callback

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Email body:

MOR_PARAM1 auth_callerid

MOR_PARAM2 first_number

MOR_PARAM3 second_number

2. Postfix server gets email and sends to /usr/local/mor/mor_email2callback.sh script

This is configured in /etc/aliases file:

callback: "|/usr/local/mor/mor_email2callback.sh"

3. /usr/local/mor/mor_email2callback.sh script parses email and sends everything to MOR.

wget -o /dev/null -O /dev/null

"http://localhost/billing/emails/email_callback?subject=$subject&param1=$param1&param2

=$param2&param3=$param3"

4. MOR GUI initiates callback using AMI command Originate.

SMS Callback

It is possible to setup SMS Callback using such provider as: https://www.aql.com/sms/sms-to-

email/

MOR API callback

About

Callback Addon is needed in order to use API callback.

Request

http://<YOUR_SERVER_IP>/billing/api/callback?u=<USERNAME>&p=<PASSWORD>&device

=<DEVICE_ID>&src=<SOURCE>&dst=<DESTINATION>

USERNAME/PASSWORD – login and password from the USER, not from the device!

DEVICE_ID – which device should be used to initiate callback and which Caller ID will

be set on the call (Device ID = Device ACC, or accountcode). You can find ACC in

/billing/devices/devices_all).

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SOURCE – which number to dial first.

DESTINATION – will be dialed when SOURCE answers.

You can omit <DESTINATION>. If you do so, SOURCE will be asked to enter it.

Response

<Status>Ok</Status>

Means callback is initiated.

Errors

Not authenticated

<Status>Not authenticated</Status>

User is not present with provided username.

User's password is wrong.

Bad device

<Status>Bad device</Status>

Device does not exist.

Device should belong to user if user does not have 'admin' rights.

No source

<Status>No source</Status>

No source in request.

Call is not received

Check your CDR for a FAILED call and reason why it failed.

Check Asterisk CLI for error.

Use Call Tracing to locate the error and fix it.

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Callback setup example

This example will demonstrate the basic steps needed to make a calling card callback setup:

Before proceeding, make sure you understand the concepts of dial plans in MOR!

In order to accomplish this you will need:

a) Two DID numbers (one can be fake)

b) One tariff for callback

c) One user + device for callback

d) Two dial plans (or more), one for callback and one for Calling Cards or Authorization by

PIN/ANI.

1. Let's create a Callback User:

2. When you create a callback user make sure to assign a correct Tariff for the callback.

Our suggestion is to include only those destinations in the tariff from which clients will be

calling to the callback.

3. Now, let's add a Callback Device:

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The Device's settings are not at all important – leave everything empty. They will not be used,

anyway.

You can even create a Virtual device for this service.

Here you can find Video which demonstrates how to create user and device.

4. Calling Card or Authorization by PIN dial plan should be created depending on your needs:

Open dial plans menu: SETTINGS -> Billing -> DIDs -> Dial Plans

We will create a calling card dial plan:

Assign the dial plan to the DID, through which it will be reached, in the DIDs menu (the DID

can be fake).

Important! This Dial Plan are active by themselves. That is, dialing the associated DID, you

will get the service described by this Dial Plan. Callback is not configured yet.

5. Now we will create a Callback Dial Plan:

1. Enter your callback dial plan name.

2. Assign a DID which will be used after callback is made (to which the call will

be connected).

3. Set the correct callback hangup cause - this allows to work with various

DID providers.

4. Wait - set how many seconds to wait before starting the callback (defined in

seconds).

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5. Select your created autodialer device here.

6. Ask for card - if enabled system will was to enter PIN after callback is

made

6. Save the dial plan and edit the settings, you will be forwarded to this page:

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Important steps here - Assign the correct DID through which the client will reach callback

service

Set the callerID which is needed. In many cases enter the Asterisk sign * - for example using

Pinless services.

7. And in the Dial Plans section, the same information is given in another perspective:

See also

Callback

Auto-Dialer Addon

Auto-Dialer addon lets you call an array of numbers automatically. When they are answered,

it plays back a prerecorded message to the receiving end of the call.

It can be used in telemarketing.

Overview

When Auto-Dialer (AD) is installed, every user can use it.

The user can find the AD menu under PERSONAL MENU -> Various -> Auto Dialer.

The main window shows the user's campaigns:

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The user can Activate or Stop the Campaign by clicking the or button near the

Campaign's status.

A Campaign can only be Activated if it has free (not-called) numbers AND some Actions.

So please, before activating a Campaign, import some Numbers and create Actions.

Add a new Campaign

Auto-Dialer can be found under PERSONAL MENU -> Various -> Auto Dialer.

Add new campaign allows you to add a new campaign:

The important fields here are:

Start/Stop Times – when calls will be made.

Max Retries – how many times to retry an unsuccessful call.

Retry Time – how many seconds the system should wait to retry the call.

Wait Time – how long to wait till the call is answered.

Device – each Campaign should be assigned to a Device in order for MOR to bill the

call.

CallerID - what CallerID to put on calls.

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Numbers

Click on in the Numbers column to check campaign numbers:

The user can import phone numbers from a TXT file. Numbers should be in correct format.

Example of numbers file:

442012345678

442087654321

442054987123

In a new Campaign there are no numbers, so click on Import numbers from file and

select the file to upload:

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After uploading the file, you will see status of imported numbers:

You can delete all numbers by clicking Delete all numbers.

If a number is Completed, it is possible to reactivate it (tell the system to call it again) by

clicking the icon near the number.

Actions

Click in the Actions column to access the Campaign's actions:

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In this window, the user programs the Actions for the Campaign:

Types of Actions:

WAIT – how many seconds to wait.

PLAY – play a file.

IVR - send call to IVR. It can be used to connect to a real person and give the callee a

greater choice. For example, it is possible to play an advertisement and tell the user to

press '1' to talk with a real person about this advertisement or similar.

o NOTE: This action is only available for admin, not for users (they can't have their

own IVRs).

Admin interface

When Auto-Dialer (AD) is installed, Admin can find it under the SETTINGS –> Addons –>

Auto Dialer menu. Here he can get a view of all users' campaigns:

The important information for each Campaign is:

Status – is the Campaign Active or Stopped?

Run time – tells when calls will be made.

All other fields are self-explanatory.

It is possible for admin to see the Actions for users' Campaigns. Just click on the Actions

button.

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Redial all failed Calls

Starting from MOR 9 you can redial all failed calls:

Workflow

Each 5 minutes, "cron job" is activated to check whether any campaigns are activated.

If there are some active campaigns, the script creates call files for the future.

It puts calls in Time Period in equal time periods, to balance the workload for Asterisk.

For example, if you have 10 calls to make, the script will execute them in 6s intervals to start

all calls in 1 min time.

Variables in /var/lib/asterisk/agi-bin/mor.conf:

calls_one_time - how many calls to distribute per one minute. (PLEASE NOTE, this value is not

for simultaneous calls! It has a very different meaning, which is explained below).

cron_interval - the interval of time (in minutes) in which we should 'put' calls.

Example #1:

calls_one_time = 10.

cron_interval = 1.

That means that in 1 minute the script will put 10 calls. So each call will be executed at a 6s

interval:

1st call: 0s

2nd call: 6s

3rd call: 12s

....

9th call: 54s

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10th call: 60s

Example #2:

calls_one_time = 5.

cron_interval = 2.

That means that in 1 minute the script will put 5 calls and it will do so for 2 minutes. So each

call will be executed at a 12s interval:

1st call: 0s

2nd call: 12s

3rd call: 24s

4rd call: 36s

5th call: 48s

6th call: 60s

7th call: 72s

8th call: 84s

9th call: 96s

10th call: 108s

NOTE: Auto-Dialer is not the tool to generate a lot of calls simultaneously. Auto-Dialer

distributes calls in even intervals in a particular time period to minimize the load on Asterisk.

Logs

Autodialer log can be found here:

/var/log/mor/mor_ad_cron.log

Reseller Addon

Description

This addon incorporates basic reseller support into MOR

Currently it lets to create user with type 'reseller'

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When this user logs into the system he is able to:

1. Create tariffs

2. Create users

3. Setup their own tariffs and assign them to their users

4. View calls and other stats for his own users

5. Add own logo to GUI

6. Customize invoice with their details (logo is not supported)

7. Can calculate own profit

When his users are making calls - reseller's balance decreases along with user's balance. If

reseller is prepaid and he is out of balance or postpaid and credit limit is reached - his users

will not be able to make calls even if they have money in their balance.

DIDs

DIDs can be created only by admin. Reseller can assign DIDs available to him to his users.

ADMIN can assign DID to USER directly bypassing RESELLER.

Complete Calls

To get all Calls in selected period for Reseller and his User use following link:

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Manual

This manual is dedicated for resellers to work on MOR

Manual for Resellers

How to change user type to reseller?

1. Go to SETTINGS –> Users:

2. Click the marked icon:

3. Select the required user type from the marked drop down box:

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Web access

Each Reseller has it's own Homepage. URL can be found in MAIN MENU -> Settings Global

tab.

It is possible to have different domain name for each Reseller. Domain name should be

pointed to Reseller's URL in such case.

NOTE: Kolmisoft does not provide domain names, so you need to check with certain domain

provider for such option.

See also

Reseller Pro Addon

MOR Addons

How to show rates for resellers users

Multi-level Reseller support

Reseller Permissions

What is Reseller Permissions?

Reseller Permissions allows system administrator to select which addons can be used by

Reseller Group.

How to set Permissions for Reseller?

To create a Reseller Group, set Permissions and assign reseller to Reseller Group please follow

these steps:

1. Go to ADDONS –> Resellers –> Reseller permissions

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2. Create a new Reseller Group:

3. Click on (edit) icon to edit a new Reseller Group:

4. Change permissions for Reseller Group:

Disabled - reseller is not allowed to use this addon.

Write - reseller is allowed to use and configure this addon. Addon configuration made by

reseller is actual for his account only.

Read - this option is for future addons.

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and click Update.

5. Go to reseller settings (Users -> ) and assign the reseller to a permissions group:

You can create more than one Reseller Group and assign more than one reseller to each

Reseller Group.

Reseller Settings

This page can be found in ADDONS –> Resellers –> Settings

Reseller Settings are global. It means that these settings will be applied for ALL resellers.

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Show Hangup Cause Codes for Resellers - if this option is enabled, resellers are able

to see Hangup Cause Codes in Calls list.

Allow use ZAP devices - allows resellers to use ZAP/DAHDI devices.

Allow use Virtual devices - allows resellers to use Virtual devices.

Resellers can add their own DIDs - allows resellers to add DIDs. If this option is

disabled, only admin can add DIDs by assigning them to resellers.

Default DID Provider for Resellers - which DID provider should be assigned to DIDs

which are created by resellers.

Allow resellers to change device PIN - allows resellers to change their users devices

PINs.

Reseller Pro Addon

What is Reseller Pro Addon?

Reseller Pro Addon is an optional MOR feature that allows Resellers to have own Providers.

Also it lets Admin to mark some of his Providers to be optionally used for Resellers.

Reseller PRO can have his own providers and create his own LCRs, but he is able to change

tariffs for his providers only.

Reseller PRO cannot affect tariff which is assigned for him by admin.

How to allow a Reseller to have own Providers?

1. Go to SETTINGS –> Users, find reseller and click icon.

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2. Check Allow to have own providers option:

Now reseller can add own Providers from Reseller Menu: MAIN MENU ->Routing ->

Providers

How can Reseller add own Providers?

Reseller can add own Providers in the same way as Admin:

http://wiki.kolmisoft.com/index.php/Material_for_testing_users#Add_a_provider_in_MOR

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How Admin providers can be used by reseller?

MOR 10

You have to edit provider settings and select an option common use. All providers with

option Common use set will be visible for all resellers who have Own providers option set in

their user details.

MOR 11

Starting from MOR 11 separate Common use providers can be dedicated to separate

resellers with different tariffs.

Configuration

1. Enable Common use option for a desired provider.

2. Go to ADDONS -> Resellers -> Common Use providers

Now you have two options:

1) Press a button marked with number 1 - this will create associations for all available

common use providers for all resellers, later you can delete unneeded associations. Please

note that when you will add a new reseller to the system - you will need to press this button

again or create associations manually as stated in option 2).

2) Using menu marked with number 2 you can manually assign common use providers for

each reseller with different tariffs.

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Manual for Resellers

Dear MOR prospective reseller, before reading any further, please note that all rates,

and taxes below are only sample rates to show you around our reseller panel but do not

reflect and are not related to our current real rates.

Dear MOR prospective reseller, This is your guide on how to navigate through your reseller

panel page by page with screen caps.

Login

When you first login from the URL http://<YOUR_URL>/billing with your reseller ID and

password, you will go to the following page:

On this page you can see on the Very Top Right, a group of flags, by clicking each flag you

can change the language of your display to your own native language or your preference, try

to click on every flag if you want to change the language.

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Then you see info about Your Balance and Calls made by your

users Today and During this month.

Now please take a look at the Navigation buttons on the left side

one by one from the Top Side Downwards:

As You can see, there is Reseller Menu and Personal menu

Under Reseller Menu you can manage your reseller data, create

and set your rates to destinations and your Tariffs while under

Personal menu you can see your own data and see how much

rates you pay for us and your own personal calls made by you

and not by users.

Tariffs

Now under Reseller Menu, Check out the first button called

Tariffs.

By clicking on this button you can start creating your own Tariffs

and Rates to destinations, and please note the following:

Extensions or devices cannot start calling, if you did not create

Tariffs.

Before creating any users, you must complete this step first, this

step will take time and it is manual, but you only must do it one

time only.

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Now click on the Tariffs:

This is empty list of your Tariffs right now.

Click on New Tariff to create new Tariff:

Enter any name, this is for you. And Purpose of Tariff chose "User Retail" for (demo purposes

now) then click the Create button.

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You will be directed to the page, which means your Tariff was created successfully:

Rates

Name of the Tariff we created for the demo purpose of User Retail is "Test Tariff"

The currency we chose for the tariff is USD.

Before someone can dial-out, rates to the destination any user will use must be filled first.

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To do this, click on the Rates:

On this page you need to enter rates manually one by one. ( Retail Rates can be transformed

from Wholesale Rates also).

Enter all the rates per minute you need to charge your customers for every destination.

Rounded by field, is in seconds, it usually is either, per one second, or per 60 seconds which

will mean the call will be rounded per minute. So make sure you change it to rounded by 60.

And enter your rates per minute for every destination or only the destinations you want.

Then click on the Update button at the end of the page, and you will see your rates have been

filled out and saved.

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More info about Tariffs/Rates can be found here

[http://wiki.kolmisoft.com/index.php/MOR_Manual#Tariffs].

Users

Now you can start creating Users and Devices for them, and those Devices can start calling if

they have enough balance in their account.

Now go to the Reseller Menu again to your left and this time click on Users to create a new

User:

Press on Add new to add new User:

Now you need to start filling out this data for the user you created:

Username/Password - For user to login to this interface to checkout his balance, rates

and later he will be able to pay affiliates automatically using Paypal inside this panel.

Tariff - from this drop down menu, you select which tariff to apply for this user since

you can create unlimited tariff plans.

Balance - enter initial balance (WITHOUT TAXES) for user here

Credit - if user is prepaid - leave it untouched

Prepaid - Make sure you check the user to PREPAID user and NOT postpaid if you do

not trust user

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After you fill in all the fields properly, click on Create button to create your first user:

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Devices

Now this user cannot start calling unless you create a device or extension for him.

To do that you must click on the icon:

This page lists all devices or extensions for this user, while each user can have unlimited

devices, only you can add for them devices and not the users themselves, now as you can see

this user still has no devices created.

To create the first device for this user click on Add new device:

Description field you can type anything you want.

Device group, you cannot edit it, so leave it as primary group.

Type - if you do not know which type to choose, then leave it as SIP, because most of

the devices used today is using SIP protocol and SIP technology

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Once you have filled in the description (Which you can also leave empty if you wish), click on

the Create button:

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Please do not edit any settings in this page, it is better if you do not know what you are

doing, that you leave this page like this, like the default settings, and hit the update button

below.

In this page you can see the device or extension number that users need to put in X-lite (or

other softphone or real phone) to dial out, for this device for example

Device number/extension is 1004

Password is 2g1eu1pa

If you put this info on your X-lite and our SERVER_IP_ADDRESS you will be able to start

making calls from this X-Lite until this User has balance in his account.

Now you can email this user his info and how to use the system, you need to only email him:

His username

His Password

Login to GUI URL http://<SERVER_IP>/billing

Device username

Device password

Devices overview

About

This functionality lists all devices that belongs to reseller's users

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Calls

If you want to check calls made by your users go to RESELLER MENU -> Calls:

Here you can press on icon near each user to check their calls.

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Personal Details

You can change your personal details in PERSONAL MENU -> Details:

Personal Rates

PERSONAL MENU -> Details -> Rates lists rates as given to you by us, this is the rates WE

CHARGE YOU per minute, So this can be your guide when you are creating your own rates for

your users, you can open both windows at the same time and make reference before you add

your rates.

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SMS Addon

Description

SMS addon lets you send SMS from MOR.

SMS addon does not support SMPP protocol.

It can send SMS over Email if Provider supports that.

Addon implements SMS billing and routing.

It is possible to:

Create many SMS Providers

o Send email to provider to initiate SMS

o Native Clickatell [www.clickatell.com] integration

o Send SMS over API

Make route (LCR) from these providers

Make Tariffs, enter Rates for SMS into these Tariffs

Let Users send SMS from their MOR login or from -------> their Mobile Phone!

o iPhone optimized!

How to enable SMS sending from start page?

Assuming that you have already set rates, LCR, etc.

1. Go to SETTINGS –> Setup –> Settings –> Addons

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2. Select the marked checkbox (marked with number 1) and press the button to save changes

(marked with number 2)

3. That is all. SMS dialog should be visible now on start page:

P.S. This SMS form will be visible only for users which have SMS subscription

Troubleshooting

1 - failed, system owner does not have rate for this destination

This means that Provider does not have rate for destination where you are trying to send

SMS.

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Solution - add such rate to appropriate prefix.

See also

MOR Addons

SMS Addon iPhone

SMS Addon - Clickatell configuration

SMS Addon - Providers

SMS Addon - Status codes

Recordings Addon

Functionality

The Recordings Addon gives more options for managing recordings, such as making it

possible to:

Send a recording to email.

Limit space users' space for recordings.

Record calls without user knowledge.

Let users manage their recordings (send to email/listen/download/delete).

Convert recordings from WAV to MP3 format and store them on an external server, to

ease the load on the main server.

NOTE: Resellers do not have the same functions as an administrator - they can't manage

recordings of their users.

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Settings

SETTINGS –> Setup –> Settings –> Addons –> Recordings

Use external Server - should system send recordings to external server?

Recordings server - details such as IP/Port/Login where port is SSH default port 22 and

Login is default root.

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Administrator functionality

ADDONS –> Recordings

Here the System administrator can see all recordings. He can search Recordings by:

Period.

Source (CallerID).

Destination (Dialed number).

User.

Device.

With each recording, it is possible to:

Check its details.

Check Call details.

Add comment to it.

Listen to it.

Download it.

Delete it.

Admin can decide:

Whether or not the user can use Recording Addon functionality (set to record calls,

send recordings to email, etc).

Which user calls should always be recorded (without user knowledge).

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SETTINGS –> Users –> Edit (on selected user)

The Recordings section has the following options:

Allow to use recording functionality - can this user use the Recording addon

functionality?

Forced record calls for this User - record calls for this user (all devices) without user

knowledge.

Server space - how much HDD space to reserve for recordings of this user.

Send deleted recordings to this email. If HDD space is used, the recording will be

deleted and sent to this email. If this email is empty, the recording will be sent to the

default user email.

Device –> Edit window

The following options are available for Device:

Record calls for this Device - record calls for this Device (without user knowledge).

Send Recording to Email - when recording is completed, send to it email.

Recordings Email - send to this email address.

Keep Recordings on Server after sent to Email - should the recording be left on the

server after being sent to email?

Recordings bulk management

About

This feature allows an administrator to quickly delete recordings for selected period or

selected device

Who can use this functionality?

This feature is only available for MOR administrator

How to use this functionality

Go to ADDONS –> Recordings –> Bulk management

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Check an option to delete by device or by date, select device/date and click Delete:

Confirm that you really want to delete recordings of selected period/device:

User settings

The Administrator must authorize the user to use the Recording Addon functionality before

User can use its functions. This can be done by checking Allow to use recording

functionality in the User Edit window.

PERSONAL MENU –> Details

Server space - shows used/reserved HDD space for recordings.

Email for deleted Recordings - to which email recordings should be sent if there is not

enough space for them.

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PERSONAL MENU –> Details –> Devices –> Edit

Record calls for this Device - user can set to record calls for this device or not.

Send Recording to Email - should recording be sent to email?

Recordings Email - the email address email to which recordings should be sent.

Keep Recordings on Server after sent to Email - should recordings be left on the server

after being sent to email?

PERSONAL MENU –> Calls –> Recordings

On this page, the User can manage his recordings.

Guides

How to set up user so he can see his calls

1. As admin, go to User Edit settings and check Allow to use recording functionality.

User will be able to use Recording addon functionality.

2. Log in as this User and in PERSONAL MENU –> Details –> Devices, click for Device and

in the Recordings section check Record calls for this Device: YES.

3. Now the user will be able to see his calls in PERSONAL MENU –> Call –> Recordings.

Mobile Number Portability Addon

Mobile number portability (MNP) enables mobile telephone users to retain their mobile

telephone numbers when changing from one mobile network operator to another.

MNP Addon for MOR allows system owner to check number if it belongs to other network

and if so - route it through different providers or bill it with different price.

Current implementation allows to save numbers in MySQL database on any server (possible

not on local server).

Before each call MOR MNP Addon checks dialed number and if number is found in database

- some prefix is added in front of it.

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By prefix MOR can set different routing(LCR) or billing(Tariff) for this number.

Detailed technique is described here: LCR/Tariff change based on call prefix

MNP Addon for MOR is available from MOR v8

As Kolmisoft cannot make one unique database for all MNP services in all countries all over

the world. MOR admin has to manually enter numbers in MNP database (process described

below) in order for this solution to work. mor_mnp database has to contain only the

necessary data (telephone numbers).

NOTE: It is important to take care of various number formats user can use to dial a number.

As a solution to it is to insert MNP number into DB in several possible formats.

Installation

Do such steps:

1. Upgrade MOR

2. cd /usr/src/mor/sh_scripts

3. ./install_mnp.sh

Upgrade

After Upgrade MOR run /usr/src/mor/upgrade/9/mnp_reenable.sh

Database

Config to DB in: /usr/local/mor/mor_mnp.conf

It should be MySQL database mor_mnp:

DROP TABLE IF EXISTS `numbers`;

CREATE TABLE `numbers` (

`number` varchar(50) NOT NULL,

`prefix` varchar(20) NOT NULL,

PRIMARY KEY (`number`),

UNIQUE KEY `number` (`number`)

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) ENGINE=InnoDB DEFAULT CHARSET=utf8;

Make sure you have set correct MySQL permissions for mor_mnp database and also local or

remote asterisk must have correct settings in /usr/local/mor/mor_mnp.conf . If your number

is still not recognized, double check if you have it in your mor_mnp database!

An example how to set MySQL permissions:

GRANT REPLICATION SLAVE , REPLICATION CLIENT ON * . * TO 'mor'@'localhost' IDENTIFIED

BY 'mor' WITH MAX_QUERIES_PER_HOUR 0

MAX_CONNECTIONS_PER_HOUR 0 MAX_UPDATES_PER_HOUR 0 MAX_USER_CONNECTIONS

0 ;

GRANT ALL PRIVILEGES ON `mor_mnp` . * TO 'mor'@'localhost' WITH GRANT OPTION;

Payment Gateway Addon

2Checkout.com

http://www.2checkout.com

2checkout needs some additional configuration on your side. In order to make it work, you

should login to your vendor account in 2checkout's site and do the following:

1. Go to "Account" -> "Site Management"

2. Set "Demo Setting" option to "Parameter".

3. Set "URL*" setting to your MOR GUI address, that is: "http://YOUR_IP/billing/"

4. Verify that "After completing an order, buyers should be:" setting is set to "Given links

back to my website".

5. Set "Approved URL" to:

"http://YOUR_IP/billing/payment_gateways/integrations/two_checkout/notify".

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Set Pending URL to

"http://YOUR_IP/billing/payment_gateways/integrations/two_checkout/pending"

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Authorize.net

http://www.authorize.net/

What is Authorize.net?

Authorize.net in Wikipedia

Where to find Authorize.net settings?

1. Navigate to ADDONS -> Payment gateways

How can I setup Authorize.net?

This example is made using ING bank as an example. Control panels of other banks may differ

but the process should be similar.

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This is Authorize.net setup page in MOR:

Enabled - Check this box to make Authorize.net available for your clients.

Login - Authorize.net account login.

Password - Authorize.net account password.

Tax in amount - "Include" tax into amount that user has to pay or "Exclude" if you

want to exclude the tax from the amount that is added to user's balance

Minimum amount - Minimum payment amount

Maximum amount - Maximum payment amount.

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Payment confirmation by you

o Not required - No confirmation is required. Every payment is added to clients

balance immediately.

o Required for suspicious payments - currently Authorize.net doesn't provide

enough info to calculate suspiciousness.

o Required for all payments - When payments are completed they must be

confirmed manually.

Notify by email - Check to receive notifications by email.

Test mode - Turns on the test mode. Use this option only if you know what you are

doing. In test mode MOR will send payment requests to bank's test environment, so

the payments will not transfer actual money. On the other hand MOR still counts these

payments as real and updates user's balance.

Logo - You can select and upload a custom logo picture. MOR accepts images that are

*.jpg, *.png or *.gif and less that 100Kb in size.

Google Checkout

http://checkout.google.com

What do you need before begining?

Before proceeding you need a valid (not self signed) SSL certificate

[http://support.google.com/checkout/sell/bin/topic.py?hl=en&topic=12156] implemented in

MOR.

All the valid certificates recognized by Google checkout are listed here

[http://support.google.com/checkout/sell/bin/answer.py?hl=en&answer=134466&from=578

56&rd=1].

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What is Google Checkout?

Google Checkout is an online payment processing service provided by Google aimed at

simplifying the process of paying for online purchases.

More information can be found in google checkout page at wiki.

How to configure Google Checkout to work with MOR?

1. Go to Google Checkout and configure it as follows:

API Callback URL must be set to:

http://YOUR_SERVER_IP/billing/payment_gateways/google_checkout/google_checkout/notify

Notification as XML must also be selected

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2. Select the marked setting:

How to configure MOR to work with Google Checkout?

1. Go to Payment gateways: ADDONS -> Payment gateways

2. Select Google Checkout

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3. The marked settings are the most important:

Tick the checkbox to enable Google Checkout payment gatewai

Enter merchant ID and key - they are visible at Google Checkout integration menu

(look at the previous picture)

Select are required currency. It is very important to select the same currency as in

Google Checkout here.

4. Save the settings

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More examples of configuration

Here are more examples of correct configuration:

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Moneybookers

http://www.moneybookers.com

What is Moneyboookers?

Moneybookers in Wikipedia [http://en.wikipedia.org/wiki/Moneybookers]

Where to find Moneybookers settings?

1. Navigate to ADDONS -> Payment gateways

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How can I setup Moneybookers?

This is Moneybookers setup page in MOR:

Enabled - Check this box to make Moneybookers available for your clients.

Login - Moneybookers account login.

Tax in amount - "Include" tax into amount that user has to pay or "Exclude" if you

want to exclude the tax from the amount that is added to user's balance

Minimum amount - Minimum payment amount

Maximum amount - Maximum payment amount.

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Payment confirmation by you

o Not required - No confirmation is required. Every payment is added to clients

balance immediately.

o Required for suspicious payments - currently Moneybookers doesn't provide

enough info to calculate suspiciousness.

o Required for all payments - When payments are completed they must be

confirmed manually.

Notify by email - Check to receive notifications by email.

Test mode - Turns on the test mode. Use this option only if you know what you are

doing. In test mode MOR will send payment requests to bank's test environment, so

the payments will not transfer actual money. On the other hand MOR still counts these

payments as real and updates user's balance.

Logo - You can select and upload a custom logo picture. MOR accepts images that are

*.jpg, *.png or *.gif and less that 100Kb in size.

How to make a payment with moneybookers?

Click on "Moneybookers" icon. You can find it on a Start page:

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Enter an amount of money, specify currency and click Confirm:

Click Confirm:

Enter your personal and card information:

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Click Continue:

Enter Card SecureCode and click Submit:

Now your payment will be confirmed.

Payment is done!

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Payment is done!

PayPal PRO

http://www.paypal.com/cgi-bin/webscr?cmd=_wp-pro-overview-outside

iDeal

iDeal [http://ideal.nl/?lang=eng-GB] [English]

iDeal [http://ideal.nl/?lang=dut-NL] [Dutch-Nederlands]

What is iDeal?

iDEAL is an Internet payment method in The Netherlands, based on online banking.

Introduced in 2005, this payment method allows customers to buy securely on the Internet

using direct online transfers from their bank account. iDEAL processed 4,5 million transfers in

2006, 15 million transfers in 2007, 28 million transfers in 2008 and 45.4 million in 2009[1].

iDEAL is owned by the Dutch organization Currence, which also owns PIN and Chipknip.

Wikipedia [http://en.wikipedia.org/wiki/IDEAL]

Supported Banks

ING advanced. Basic account type currently is not supported.

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More details about banks

The bank selection in iDeal is 2 ways.

Firstly it is the merchant (your) bank. This is where we send payment requests and that

requires configuration. Currently we have configuration for only one bank (ING).

Secondly there is a client bank (your customers). iDeal implementation fetches those

on the runtime so it always contain full list of iDeal banks.

Where to find iDeal settings?

1. Navigate to ADDONS -> Payment gateways

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2. Select the marked iDeal logo:

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How can I setup iDeal?

This example is made using ING bank as an example. Control panels of other banks may differ

but the process should be similar.

This is iDeal setup page in MOR:

Enabled - Check this box to make iDeal available for your clients.

Acquirer bank - Select bank that issued your iDeal account.

Merchant ID - Your merchant ID from iDEAL. You can find it in iDeal control panel.

Passphrase - Enter passphrase that you used when you generated you certificates. To

get more information about generating certificates refer to the section Generating

Certificates

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Payment description - Custom textual description that describes what your customer

is paying for.

Private key (*.pem) - Upload private key that you have generated. More information

in "Generate certificates"

Private certificate (*.cer) - Upload a certificate that you have generated. More

information can be found in "Generate certificates"

Ideal certificate (*.cer) - Select certificate file that can download from your iDeal

provider.

Tax in amount - "Include" tax into amount that user has to pay or "Exclude" if you

want to exclude the tax from the amount that is added to user's balance

Minimum amount - Minimum payment amount

Maximum amount - Maximum payment amount.

Client pays transaction fee - Check to additionally charge the client for the

transaction fee.

Transaction fee amount - Enter transaction fee amount if you have enabled "Client

pays transaction fee"

Payment confirmation by you

o Not required - No confirmation is required. Every payment is added to clients

balance immediately.

o Required for suspicious payments - currently iDeal doesn't provide enough

info to calculate suspiciousness.

o Required for all payments - When payments are completed they must be

confirmed manually.

Notify by email - Check to receive notifications by email.

Test mode - Turns on the test mode. Use this option only if you know what you are

doing. In test mode MOR will send payment requests to bank's test environment, so

the payments will not transfer actual money. On the other hand MOR still counts these

payments as real and updates user's balance.

Logo - You can select and upload a custom logo picture. MOR accepts images that are

*.jpg, *.png or *.gif and less that 100Kb in size.

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Generating certificates

In Linux console type these commands, but replace the_passphrase with your desired

passphrase. Do not forget your passphrase because you will have to enter it in iDeal

configuration window.

/usr/bin/openssl genrsa -des3 -out private_key.pem -passout pass:the_passphrase 1024

/usr/bin/openssl req -x509 -new -key private_key.pem -passin pass:the_passphrase -days

3650 -out private_certificate.cer

More information can be found here [http://dovadi.com/ideal].

Transferring certificates to another server

To find your iDeal Private key, Private certificate and Ideal certificate issue these commands:

mysql mor;

Ideal certificate (*.cer):

select * from conflines where name like "%ideal_ideal_ideal_certificate_file%";

Private key (*.pem):

select * from conflines where name like "%ideal_ideal_private_key_file%";

Private certificate (*.cer):

select * from conflines where name like "%ideal_ideal_private_certificate_file%";

Copy from: -----BEGIN CERTIFICATE-----

till: -----END CERTIFICATE-----

please note that these commands can retrieve more than 1 certificate, the one you

need is that matches owner_id field to the user id in users list.

paste to an empty text files and save. Then you can import them in MOR as normaly.

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Payment status

Successful:

Completed - Payment is made and completed, user's balance was increased.

Waiting for confirmation - Payment was completed by iDeal and is currently waiting

for your confirmation. Money will be added to user's balance after you confirm the

payment

Unsuccessful:

Canceled - Payment was canceled.

Expired - Payment expired. Usually because your customers haven't completed the

payment in 60 minutes.

Failure - There was an error in iDeal when dealing with transaction.

Troubleshooting

"Select bank" is empty.

1. Check all configuration options

2. Check certificates.

"Select bank" shows only "Issuer Simulator" as an available bank

Test mode should be turned off.

Payments "Confirmed" column displays "No (waiting_response)" and

payment is not being completed.

If iDeal fails to deliver money instantly (lag between issuer and acquirer banks) then MOR

marks this payment as "waiting_response" and then waits for another hour and checks iDeal

again, so be patient. If waiting_response stays like that for several hours then contact our

Support.

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OSMP

Technical details can be found in osmp technical details page at wiki

www.osmp.ru [Russian]

What is OSMP

OSMP is a payment gateway that lets your clients to top up their balance. The main difference

from other payment gateways is that OSMP allows you to pay using physical OSMP payment

terminals rather than MOR GUI.

How OSMP works?

Client enters his username and amount into OSMP payment terminal.

OSMP checks payment details with MOR (is there such user in MOR; amount sizes;

other technical technical details )

o User is checked buy matching user name provided by user in OSMP terminal

with MOR username. Example. if we have user with username "admin" in MOR

he should enter "admin" as his user name in OSMP.

o Other details include:

Is this transaction unique and it is not a hacking attempt.

Is amount greater that Min Amount (if Min Amount is provided in

configuration)

Is amount smaller that Max Amount (if Max Amount is provided in

configuration)

If payment can be created inside MOR and no errors occur.

If everything is correct then user pays using OSMP terminal.

After the payment has been successfully completed OSMP announces MOR about this

and MOR tops up users balance.

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In what countries OSMP is available

OSMP is only available in these coutries: Belarus, Tajikistan, Uzbekistan, Kazakhstan,

Georgia, Moldova, Ukraine, China, Russia

Where to find OSMP settings in MOR?

1. Navigate to ADDONS - Payment gateways

2. Select the OSMP gateway:

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How to configure SSL for OSMP?

In order to use OSMP securely SSL certificates are being used. Certificate configuration and

payment gateway testing is being done by OSMP staff.

If you want to do it by yourself the configuration file /etc/httpd/conf/httpd.conf should

look like this:

NameVirtualHost *:80

NameVirtualHost *:443

<VirtualHost *:80>

RewriteCond %{REQUEST_URI} !^/billing/public

RewriteRule ^/billing(/.*)?$ /billing/public

<Directory /var/www/billing/public/>

Options ExecCGI FollowSymLinks

AllowOverride All

Allow from all

Order allow,deny

</Directory>

<Directory /var/www/html/stats/>

AddHandler cgi-script .pl

Options +ExecCGI

DirectoryIndex index.pl

</Directory>

</VirtualHost>

<VirtualHost *:443>

SSLEngine on

SSLCipherSuite

ALL:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL

SSLCertificateFile /etc/ssl/billing.crt

SSLCertificateKeyFile /etc/ssl/billing.key

SSLCACertificateFile /etc/ssl/comodo.intermediate.crt

RewriteCond %{REQUEST_URI} !^/billing/public

RewriteRule ^/billing(/.*)?$ /billing/public

<Directory /var/www/billing/public/>

Options ExecCGI FollowSymLinks

AllowOverride All

Allow from all

Order allow,deny

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</Directory>

<Location /billing/payment_gateways/osmp/osmp/notify>

Allow from 195.22.241.194 195.22.228.238 89.28.59.10 #contact OSMP for exact details

which IP's must be allowed

Deny from All

</Location>

<Directory /var/www/html/stats/>

AddHandler cgi-script .pl

Options +ExecCGI

DirectoryIndex index.pl

</Directory>

</VirtualHost>

Advices

You should enable manual confirmation for this payment gateway

I found a bug, OSMP logo cannot be clicked like other payment gateway logos!

This is not a BUG, that's because of the way OSMP works - payments are made from physical

terminals, please read ―what is OSMP― section in osmp technical details page

HSBC

http://www.hsbc.com/

How can I setup HSBC?

Enabled - Check this box to make HSBC available for your clients.

Login - HSBC account login.

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Client ID - Your client ID from HSBC. You can find it in HSBC control panel.

Password - HSBC account password.

XML url - The URL to send the request in XML.

Tax in amount - "Include" tax into amount that user has to pay or "Exclude" if you

want to exclude the tax from the amount that is added to user's balance

Minimum amount - Minimum payment amount

Maximum amount - Maximum payment amount.

Payment confirmation by you

o Not required - No confirmation is required. Every payment is added to clients

balance immediately.

o Required for suspicious payments - currently iDeal doesn't provide enough

info to calculate suspiciousness.

o Required for all payments - When payments are completed they must be

confirmed manually.

Notify by email - Check to receive notifications by email.

Logo - You can select and upload a custom logo picture. MOR accepts images that are

*.jpg, *.png or *.gif and less that 100Kb in size.

Test mode - Turns on the test mode. Use this option only if you know what you are

doing. In test mode MOR will send payment requests to bank's test environment, so

the payments will not transfer actual money. On the other hand MOR still counts these

payments as real and updates user's balance.

Other Notes

About SSL Certificates in Russian

Van's Apache SSL/TLS mini-HOWTO

Monitorings Addon

What is Monitorings Addon?

Monitorings Addon is an optional MOR feature that:

protects you from high money losses.

o blocks suspicious users on various activity

o informs administrators about suspicious activity

allows admin and resellers to spy their user calls on real-time

allows to see leg A and leg B codecs used and post-dial delay in Active calls.

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You can find monitorings in ADDONS -> Monitorings.

Remember, you will only find this addon if you have purchased it.

How Monitorings Addon work?

Monitoring Addon crawls the MOR database and counts user's call price sum over specified

period. If user's call price sum over specified period will be higher or lower than the amount

specified in monitoring - an action will be taken. Possible action are:

Only notify admin (no action is taken against user)

Block user (user is automatically blocked, no notification is sent to admin)

Block user and send notification for admin

You can create differently configured monitorings for:

All users

Postpaid users

Prepaid users

1 user (separate monitorings for each user, also called personal monitorings)

Also it is possible to create 2 or more monitorings for each group, for example:

First monitoring only reports admin about quickly increasing sum, let's say 50 Euro

If an administrator takes no action and someone tries to abuse the system when call

sum for that user reaches for example 100 Euro the second monitoring can block that

user

System's administrator can turn on monitorings for resellers. If they are off, resellers' users are

under the terms set by administrator.

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Monitorings for resellers

As it is mentioned above, administrator can turn on monitorings for resellers. If you want to

do it go to ADDONS -> Resellers -> Reseller permissions. Choose or create a group and

click the Edit icon. You will see this window:

Choose whether you want to permit reading writing or to keep monitorings for resellers

disabled (this is default).

How to create a Monitoring?

Global Monitoring

Global Monitoring will be applied to all users which meet the requirements (this will be

explained later)

1. Go to ADDONS section -> Monitorings.

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2. Select the desired options for new global Monitoring:

Period - this option let's you specify how often the monitoring must run and check all

users balance.

Amount - this option let's you specify the amount of money and whether it is

maximum limit or minimum call price limit. When user's balance drops to or

reaches the amount you specified an action will be taken against him depending on

other parameters.

Block user - if this option is set to Block - user will be blocked. He will not be able to

make any calls.

Send Email to Admin - here you may choose to be informed about the action by

email.

User type - this options specifies a user group an action must be taken against.

o All - Monitoring will be applied to All users

o Postpaid - Monitoring will be applied to Postpaid users

o Prepaid -Monitoring will be applied to Prepaid users

3. Press Create to create the Monitoring

Additional Notes

You can create as many Monitorings for any group as you want. For example you can

create 2 monitorings for Postpaid users:

o 1 monitoring could just inform the administrator about the high user's balance

(in this case select these options: enter the amount and choose Increases more,

choose Do not block, choose Send, choose Postpaid)

o 2 monitoring could block the user with high balance if the administrator is

sleeping and didn't saw the warning email (in this case enter a bigger amount

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than in the first monitoring and choose Increases more, choose Block,

choose Send, choose User type: Postpaid)

o 3 monitoring could inform the administrator that some users stoped calling for

some reason, maybe because of some technical problems, that otherwise would

not be noticed so quickly (in this case enter some amount and choose Drops

bellow, choose Do not block, choose Send, choose User type: Postpaid)

Do not add too much monitorings, because this functionality is database

intensive. A separate database server could be dedicated for monitorings.

Personal Monitorings

What is a Personal Monitoring?

A Personal Monitoring is applied only to a specific user.

How to create a Personal Monitoring?

1. Navigate to SETTINGS -> Users

2. Select the marked icon:

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3. Select the marked icon:

4. Depending on your scenario enter correct settings for a monitoring. Some information

about marked fields:

1. Must never be equal 0

2. or 3. must be set to block or to send. You may select both, but you cannot leave

them both Do not block and Do not send

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5. Press the marked button to create the monitoring:

6. You will get a confirmation that monitoring was created:

Monitoring alerts when usage drops

In MOR it is also possible to monitor if users call price sum is under some certain certain sum.

This is useful to detect that something is wrong with user's settings, providers assigned to

him, etc.. Go to ADDONS -> Monitorings if you want to make a monitoring for users' group

or SETTINGS -> Users if you want to make a monitoring for single users.

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You will see this window:

Select Drops below, choose other settings and click Create. The text next to the information

icon below shows what actions will be taken under what conditions depending on your

choices.

Disable global monitorings for 1 or more users

In some situations you might need to disable global monitorings for some users leaving only

personal monitorings. You can do that by navigating to user settings and enabling the

marked setting:

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Recommended hardware

Minimum requirements:

20GB of free space on HDD

2GB of RAM

See also

User Blocking

Spy real-time calls

About

This functionality is a part of Monitorings Addon and is used to spy user's real-time calls. This

functionality allows call-center managers to assess their employees work. It is also used in

prisons where such functionality is also necessary.

MOR user's which are allowed to use this functionality:

Admin

Reseller (active calls must be enabled for resellers)

How this functionality works?

1. The manager selects a device where he will receive calls from the system when he wants to

spy his user (this has to be done only once).

2. Manager configures his phone to be able to answer it.

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3. Manager goes to active calls and presses a Spy icon on a desired call:

4. MOR system sends a call to a device set in step 1.

5. Manager answer's the call from the system and is able only to listen.

Please note that it is not possible to spy calls if their Server IDs do not match Spy Device

Server ID of the user (admin/reseller) who is viewing the calls. Icons for such calls are fainted:

Enable/disable this functionality

Enable

1. Go to reseller's or admin's user settings

2. Select a desired device where you want the system to send a call being spied:

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Disable

You can also globally disable ChanSpy. Go to ADDONS -> Monitorings -> Settings. You will

see this window:

Select Chanspy is disabled globaly: and click Save changes.

What is needed to use this feature?

Admin:

System must have Monitorings Addon

A spied device cannot have recordings enabled at the same time. If this condition is

not met - you will not be able to spy this device.

Reseller:

System must have Monitorings Addon

Active calls must be enabled for resellers

A spied device cannot have recordings enabled at the same time. If this condition is

not met - you will not be able to spy this device.

Other setttings

When spying you can press # to increase/decrease sound volume.

Available sound levels:

1 2 3 4 -4 -3 -2 -1 0(default)

Here:

-4 - is the quietest

4 - is the loudest

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Limitations

If you have 2 Asterisk servers in your system, both devices for the "Spy action" have to

registered to the same Asterisk server.

Call Shop Addon

What is Call Shop?

A callshop is a business that physically provides phones for the purpose of making long-

distance telephone calls. More information can be found in Call shop page at wiki.

What is a Call Shop Addon?

It is an additional functionality which allows to have and manage Call Shops for Admin and

Resellers.

Download Call Shop Brochure here

[http://www.kolmisoft.com/uploads/files/Call_Shop_Brochure.pdf].

Typical Call Shop Scenarios

Prepaid

Customer visits the call shop.

Customer pre-pays the operator for the call.

The operator will then activate a phone booth for the customer using billing software.

The customer will go to the phone booth and dial the destination number.

Billing software will record the call details and the corresponding call charges for

future reporting needs.

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Postpaid

Customer visits the call shop.

Customer chooses a vacant phone booth and dials the destination number.

Billing software keeps track of each phone booth's call details and the corresponding

call charges for invoicing and future reporting needs.

When the customer has completed his/her calls, the operator generates an invoice for

the customer's calls.

Manager's view

Call Shop Management in Admin's/Reseller's view

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Terms

Reseller

Reseller is a Call Shop owner. Reseller can have many Call Shops.

Manager

Manager is a person who works for reseller. He works in a Call Shop and does these tasks:

Takes money from users

Gives change to a user

Prepares a Call Booth for a user, so he would be able to call

Manager user belongs to reseller

Call booth

Call booth is a special user in MOR billing system dedicated for use in a Call Shop.

Call booth user belongs to reseller.

How to create a Call Shop?

1. You have to be Logged in as an admin or a reseller

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2. Navigate to ADDONS -> Call Shop:

3. Select the marked icon:

4. Fill the required fields and press Create:

Name - the name for Call Shop

Manager's interface language - this will be the default language for Call Shop manager (the

person who will accept payments, enables call booths, etc.)

Invoice details (as seen in the invoice header) - the information you will enter here will be

displayed in the invoice.

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5. Now you have to assign some call booths, select the marked icon to do that:

6. Select a required user from dropdown box (marked with number 1) and press a Add user as

booth (marked with number 2):

Very important! You have to dedicate user's for call shop booths, that means you must

not use these users anywhere else in MOR.

In this way add as many users as you need.

When there is no manager assigned in Call Shop when adding new call booth you will be

offered to mark it as a manager. Please do so. If you skip this step you can mark a booth as a

manager by pressing a marked icon:

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Now you can see, that user's type was changed to manager and you can see a new icon:

Login as manager. You can use this icon to quickly access the manager's view to help him

solve problems.

P.S. The manager has his own way to access his manager view. Check „How can manager

access his management interface?― section in Call shop addoin page for more info.

Call Shop manager interface

How can manager access his management interface?

1. The manager has to enter the server address in his browser (the same way as you access

MOR gui). Most commonly it would look like this: http://www.example.com/billing or

http://x.x.x.x/billing, where x.x.x.x is an IP address.

2. In open window the manager has to enter his Login and Password:

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3. If Login and Password is correct the manager is redirected straight to his manager

interface:

How can the manager assign a call booth for a user?

To assign a call booth the manager has to complete these steps:

1. Press the marked icon:

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2. After pressing the start button the following settings window will appear:

3. Now the manager has 2 options - to mark the user as prepaid or postpaid:

Prepaid - user pays before a call.

1) To mark a user as Prepaid select the marked check box (number 1).

2) In box 2 enter the amount of money the user has paid.

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3) Press the button marked with number 3

Postpaid - user pays after a call.

1) To mark a user as Postpaid select the marked check box (number 1)

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2) Press the marked button with number 2

The field Comment can be used to remind yourself some specific details about a user like

"The man with a hat" or/and to include some personal info about this user like his Name,

Surname, Driver's licence number etc. as this is a requirement in some countries. This field is

not limited by any size limits.

4. The booth status changes to reserved (user didn't called any number yet):

The user can make a call now

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How can the manager stop the user call session, write invoice and take

money?

1. Firstly you need to press the marked icon:

2. The window expands and you will see a lot of valuable information about the current user

call session. You can see how much money the user has paid and how much money should

be returned to user if his call session type was Prepaid.

The session is NOT ENDED until you press the marked button:

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3. When you press the marked button (End Session and Generate Invoice) the call session is

stopped (the user will not be able to make any calls) and an invoice is generated. You can find

the generated invoice in Invoices section - press the marked button:

4. To view and print an invoice press the marked icon:

User type is prepaid, how can I top up his balance during the call?

1. Click the marked icon:

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2. Enter the required amount (marked with number 1) and choose to increase (marked with

number 2) or to decrease (marked with number 3) the current user balance.

3. Depending on what you have chosen (Increase or Decrease) you will see a similar

confirmation:

Press the marked button to confirm.

4. The balance get's changed.

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Call Shop taxes

Taxes are counted and shown in Call Shop invoices only if they are set in that call

booth's user settings (this allows you to have different call booths with different taxes)

How to enable taxes for a call booth?

1. Go to ADDONS -> Call Shop:

2. Go to Call booths:

3. Now select a booth you want to set tax:

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4. Find a displayed section in settings and set tax for a call booth:

More information about taxes can be found in multi-tax system page at wiki.

5. Save the changes

Now when you will end a calling session an invoice will be generated with taxes:

Rate search

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Internal information on how Call Shop addon works

When an ordinary user is added to callshop - it is automatically blocked so he would not be

able to make calls. When calling session is started by the manager - it is unblocked (can make

calls now). When the session is ended - the user account is blocked again. You can always

check current booth blocked/unblocked status from admin or reseller in that booth details.

Please do not interfere with MOR system and do not change Blocked/Not blocked

status for call booth users.

When Call Shop Manager creates Prepaid or Postpaid session in Call Shop - MOR system also

changes Postpaid/Prepaid option and balance for that call booth user. Please do not edit

these options manually - it is not necessary.

Features which might be implemented in the future

1. Information about a number: you enter the number and get the information about the

price, destination, pricing type (per minute, per second, etc)

2. Possibility for admin to hide some reseller features if needed:

Hide Settings

3. Possibility to hide Comment field from manager view

4. Possibility to hide Price/Min field from manager view

5. Auto drop a call after a predefined amount of minutes. This setting will allow to drop a call

for example after 2 minutes if user want's to pay only for a two minutes.

6. Statistics for each Call Shop: Money collected, Calls made, etc.

Other notes

1 call booth (phone) = 1 MOR user with 1 device. If there are more than 1 device

assigned for a user - please leave only 1.

The number of call booths (phones) is unlimited for each Call Shop

Questions and Answers

Which user should have the credit in order Call Shop could make outgoing calls

That depends. If reseller uses his own providers - he must have enough balance (if he is

prepaid) on the provider side to make calls. If reseller is using admin's providers -

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then reseller must have enough balance in MOR in order Call Shop call booths could make

calls.

Call Shop manager and a call booth does not need to have any balance. Zero balance is OK.

When the call session is terminated by Call Shop manager - call booth balance is reseted to

0.00

Which browser should I use for manager view for best user experience?

For best experience use Mozilla Firefox. You can get it here [www.mozilla.com]

How many users should I add to a Call Shop?

You should add 1 manager (you will not be able to see this user in Call Shop

manager's view as it is needed only for administrating purposes)

You should add as many ordinary users as you need. This number depends on how

many phone booths (to simplify: phones) you have in your 1 call shop.

For example: You have 8 call booths (phones) in your call shop - then you need to create 9

users. 1 user account for a manager and 8 for call booths (phones).

How to setup tariffs (prices) for a Call Shop

Please refer to tariffs section in MOR manual to answer this question in detail. All information

is valid for Call Shop also, because Call Shop users are the same MOR user's just logically

grouped into Call Shops.

Basic scenario how admin/reseller should setup a Call Shop with tariff

1. Create rates (tariff import). More information about: rates

2. Assign that tariff to the users he will use in Call Shop

3. Create Call Shop and assign users to Call Shop as booths

Done, now Call Shop manager (the person who sits in Call Shop and collects money) enables

call booths from manager interface by assigning them credit/balance.

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Other notes

Admin/Reseller can change Call Shop prices by assigning a different tariff for a call

both (user).

Call Shop booths (users) cannot change prices.

Admin can check his resellers rates by loging in as reseller.

How to setup a call booth that it would bill in increments: per second, per minute or

preferential?

This can be achieved by creating a separate tariff with such settings. Then you will have

to assign this tariff for each Call Shop call booth (phone).

I would like that Call Shop owner would be able to import tariffs for his Call Shops, how

to do that?

Create a reseller account for each Call Shop owner.

I want to separate billing statistics per Callshop

Create a Reseller account per each Call Shop and configure Call Shops on these

Resellers. Billing information will be separated per Reseller.

See also

What is a reseller?

What is a Call Shop?

Tariffs

MOR Addons

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Skype Addon

IMPORTANT! Skype channels for Asterisk are no longer available for sale after Microsoft

bought Skype, so this addon is not usable for this moment. For existing users, who have

bought Skype channels, Skype for Asterisk will be supported till July 26, 2013.

What is Skype?

Skype is a software application that allows users to make voice calls and chats over the

Internet. Read more about at wikipedia here [http://en.wikipedia.org/wiki/Skype].

What I need in order MOR and Skype to work together?

You need to buy a Skype Addon from Kolmisoft [http://www.kolmisoft.com/contacts]

You need to buy Skype channels from Digium.

[http://store.digium.com/productview.php?product_code=1SFA0001]

MOR version must be 10 or greater

Asterisk version must be 1.4.37-rc1 or higher (it is default in MOR 10)

Use of Skype for Asterisk requires a Skype Manager account. Please see Skype

Business Account creation for more details.

What configurations are possible with a Skype addon?

OUTGOING (as a device)

Through Skype call to Skype network to another Skype user

Skype as a Device

OUTGOING (Skype as a provider)

Through Skype call to outside - PSTN.

Calling through it to SkypeOut if you have enough money in Skype account.

INCOMING (Skype as a provider)

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Calling to Skype account from any other Skype account.

Calling from Skype and reaching Asterisk - then you can do something with that call.

For example you can set up an IVR which will pass the call to right person in your

company.

Where to find Skype Addon in MOR?

If you have bought Skype Addon you will find it in ADDONS section in MOR menu: ADDONS

–> Skype

Skype configuration

Skype provider

In order to use Skype you must configure a provider of type Skype in MOR:

1. Go to SETTINGS –> Billing –> Providers:

2. Press New Provider

3. Fill all fields and select Skype as Technology:

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4. Fill Skype Name and Password, choose audio Codec (we recommend G.711 u-law when

possible):

5. Press Edit to save changes

Skype Device

In order to use Skype in MOR you have to create (please note that you can't edit existing

device and change Technology to Skype, you must create a new one) a device of type

Skype for a desired user:

1. Go to user's device list

2. Press Add new device

3. Select device type Skype:

4. Press Create

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5. Enter a correct Skype Name:

Skype FAQ

Q. Is it possible to make world-wide PSTN calls to landline and mobile phones

(SkypeOut)?

Yes, but please note: Skype consumer subscription prices for unlimited calling do not

apply.

Q. Is it possible to use simple Skype plans which gives discounts

No.

See also

Skype for Asterisk admin manual

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Skype channel installation guide

How to buy Skype credits from Digium?

1. Go to https://www.digium.com/en/login/register.php and register an account by filling

these and other required field by Digium:

2. After submitting the registration you will get an email from Digium. Validate you email and

activate the account by clicking the link in email.

3. Now Log in with an account details you registered with Digium:

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4. Go to Digium store

5. Select how many channels you would like to buy and press "Add to Cart":

6. If all provided information is correct - checkout:

7. Complete the transaction by paying for selected amount of Skype channels via your

preferred payment method (Visa, Mastercard, American Express, PayPal are supported as of

time of writing this article).

Skype installation

cd /usr/src

mkdir -p digium

cd digium

wget http://downloads.digium.com/pub/register/linux/register

chmod 777 register

# run the command

./register

# now answer the questions in the following order:

# 1. Choose 1.

# 2. Choose 7.

# 3. Enter the key

# 4. Press y for yes

# 5. Press y for yes

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After completing these steps you will see a similar output:

Wrote license to /var/lib/asterisk/licenses/S4A-D3Z8TJAQZNCQ.lic

Now download Skype For Asterisk from

http://downloads.digium.com/pub/telephony/skypeforasterisk:

wget

http://downloads.digium.com/pub/telephony/skypeforasterisk/asterisk-1.4/x86-

64/skypeforasterisk-1.4_1.1.2-x86_64.tar.gz

Extract the archive and install:

tar -xvzf skypeforasterisk-1.4_1.1.2-x86_64.tar.gz

cd skypeforasterisk-1.4_1.1.2-x86_64

make

make install

make samples

Now check in Asterisk CLI how many licensed channels you have:

ns356744*CLI> skype show licenses

Skype For Asterisk Licensing Information

========================================

Total licensed channels: 1

Licenses Found:

File: S4A-D3Z8TJAQZNCQ.lic -- Key: S4A-D3Z8TJAQZNCQ -- Expires: 2031-03-08 --

Host-ID: 25:d3:8b:cd:0c:0d:65:ec:a9:f8:87:7a:52:cd:aa:f9:a9:8b:a1:a1 -- Channels:

1 (OK)

Configuration

Configurations for Skype are located in /etc/asterisk/chan_skype.conf

Skype Business Account creation

1. Create a Skype Business Account at http://skype.com/business

2. Log in to https://manager.skype.com and answer all questions.

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3. You will receive an email from Skype. Activate your Skype Business account by visiting the

provided activation link in email.

4. Create Skype accounts as described in this manual

Create Skype accounts for calling

Go to https://manager.skype.com and follow the steps:

1.

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2.

3.

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4.

5.

DO NOT CREATE ACCOUNTS IN ANY OTHER METHOD - THEY WILL NOT WORK WITH

Skype For Asterisk

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Skype troubleshooting

Skype For Asterisk does not compile successfully

[root@ns356744 skypeforasterisk-1.4_1.1.2-x86_64]# make

gcc -I/usr/include -o chan_skype.o -c chan_skype.c -pipe -O3 -fPIC -

DAST_MODULE=\"chan_skype\" -MD -MT chan_skype.o -MF .chan_skype.o.d -MP

chan_skype.c: In function 'indicate':

chan_skype.c:457: error: 'AST_CONTROL_SRCUPDATE' undeclared (first use in this

function)

chan_skype.c:457: error: (Each undeclared identifier is reported only once

chan_skype.c:457: error: for each function it appears in.)

make: *** [chan_skype.o] Error 1

Your Asterisk version is too old. Please contact Kolmisoft to upgrade it to 1.4.37 or later

I cannot load channel

ns356744*CLI> load chan_skype.so

[2011-03-10 11:21:53] WARNING[27423]: loader.c:434 load_dynamic_module: Error

loading module 'chan_skype.so': /usr/lib/asterisk/modules/chan_skype.so: undefined

symbol: sfa_license_iterator_destroy

[2011-03-10 11:21:53] WARNING[27423]: loader.c:777 load_resource: Module

'chan_skype.so' could not be loaded.

Do not load channel, just restart asterisk

When chan_skype.so attempts to load, I get an error

When chan_skype.so attempts to load, I get an error similar to:

ERROR[1702]: core.cpp:1553 sfa_startup: Skype engine failed to start.

The skyhost binary (which shows up in the process list as skypeforasterisk)

failed to start.

This could be because a previous instance of this binary is running and was not properly shut

down. Since any new instances try to bind to the same port, they fail. If 'ps aux|grep

skypeforasterisk' returns any skypeforasterisk processes when Asterisk is not running, kill

them and restart asterisk.

Another reason for failure could be that disable_tcpauto is set to no (the default) and can not

bind to ports 80 and 443 due to asterisk running as a non-root user or other services being

bound to those ports.

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Failure can also occur if the engine_directory is set to a directory which is mounted with the

'noexec' flag or to a directory where the user does not have execute permission.

Cannot restore segment prot after reloc: Permission denied

cannot restore segment prot after reloc: Permission denied‖

You have to disable Selinux

I just created an account to manage my Business Control Panel

and I can't get that account to log in via SFA (Skype For

Asterisk).

Only Skype accounts that are actually created from inside the Skype Business Control Panel

can be used. The account that was created to admin the BCP was not created from inside the

BCP, so it will not work with SFA. Inside the BCP, go to Add Members and click "Create a

Business account" to create an account that can be used with SFA.

To create Skype accounts please follow this guide

[http://wiki.kolmisoft.com/index.php/Skype_channel_installation_guide#Create_Skype_accoun

ts_for_calling].

Provider billing addon

Description

Provider billing addon allows reversed billing - it counts provider's balance which increases

when admin or reseller makes a call through it.

Balances

Provider has balance which increases when call goes over this Provider. E.g. Provider receives

money for the calls over him.

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If Provider is assigned to the User then such Provider can make calls. When Provider (which is

assigned to the User) makes calls, such calls goes over other Providers and User's (to which

Provider is assigned) balance is decreased.

IMPORTANT! Provider's balance and User's balance are two different values. They are not

connected and do not impact each other. They change independently of each other.

Manual payment for Provider balance

To add manual payment for provider in main provider billing addon window click and you

will be lead to this window:

Click Proceed:

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Click Confirm. You will be redirected to Payments window and see this payment if everything

is correct.

Please note that adding positive Amount will decrease provider's balance and adding

negative will increase it. Balance - amount = new balance.

Reseller can use this feature only on that reseller's providers.

Using the addon

To reach the provider billing go to ADDONS -> Provider billing. You will see this window:

By clicking Edit you will be lead to provider's edit window.

Limit to device or provider per day

After you were lead to provider's edition page you can select a minute limit per day for

provider. You will find it in Advanced block.

If the limit is exceeded that provider is disabled which means it is not available to make calls

through it anymore. If you add a value of zero, no limit will be applied.

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To apply this functionality to device go to SETTINGS -> Setup -> Default device and

see Advanced block.

This functionality is available only if provider billing addon is enabled.

Tutorials

Usual configuration sequence

1) Make a provider's tariff, then import it.

2) Make a provider, then assign the provider's tariff to the provider itself.

3) Make a user tariff by adding a % on top of the provider tariff.

4) Make a user.

5) Make at least one device for user.

6) Make LCR.

7) Add your provider to LCR and assign it to your user.

8) It is a good idea to check whether everything is working by using call tracing.

9) Register your device to MOR using your user's device details (username and password (not

PIN)) and then make a call.

If you have any questions or suggestions, or can't solve a problem on your own, please use

our forum or support system, or just send an email to [email protected]

Conference with MOR + Trixbox

IT IS NOT POSSIBLE RUN CONFERENCES ONLY WITH MOR. YOU NEED

EXTERNAL SOFTWARE, SUCH AS TRIXBOX.

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Tested with 0.7, 0.8 MORs and 2.6.2.3 Trixbox.

Settings for MOR

First of all you need a user and device in MOR [http://www.kolmisoft.com/billing-and-

routing/] for a DID channel. Make sure you use IP authentication and use the exact settings in

the image below (change 192.168.0.165 to Trixbox's IP address, and Trunk? should be set to

Yes).

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Create a DID and map it to the device.

Settings for Trixbox

Log in as administrator to Trixbox and create a Trunk with the following settings: Change

192.168.0.148 to your MOR server IP address and 1005 username and secret to the

appropriate settings.

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Create a Conference with these settings:

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Then create a DID (Inbound route):

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Then try to call from any device to 8000 in MOR. You should hear a woman's voice and

instructions on what you need to do.

Of course, if you change settings everywhere from 8000 to your real DID, it will work as well.

A few hints

For each new conference room you will need a separate device and a DID in MOR.

How to configure 911 service

Description

This guide will show how to configure 911, 411 or any other similar service.

You can read more about it at: VoIP 911 Services

[http://www.myvoipprovider.com/VoIP_911_Service].

First of all, you need a provider that will allow you to use such a service. Some such providers

may be found here [http://www.voip-info.org/wiki/view/VOIP+911+Service+Providers].

When you buy a service from such a provider, he will give you an access number to dial.

As an example, let's take the number 12125551212, which should be dialed to access the

service.

Short guide

In order to configure number 911, we will need to:

Create a User/Virtual Device for the 911 service.

Forward a Virtual Device to 12125551212.

Create a DID 911.

Assign a Virtual Device to DID 911.

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Detailed guide

Create User/ Virtual Device for 911 service:

Forward Virtual Device to 12125551212:

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Create a DID 911 and assign a Virtual Device to it:

Another way

Create User 911.

Create Virtual Device with Extension 911 for this user.

Make correct LCR for this user (with correct Provider which can handle 911 calls).

In Devices Call Flow, Forward the call to the External number given by your Provider.

How to make several connections from one IP

Description

In order to separate traffic, it is sometimes necessary to make several connections to MOR

from one IP (usually a server).

The main rules for how to do this are:

1. Devices should be DYNAMIC and should REGISTER to MOR.

2. In the INVITE packet from the external server, which device is dialing should be visible

in the From: line. (If the external server is Asterisk, then in the MOR descriptions (in

sip.conf file) on the external server, there should be fromuser= lines that point to

which device is dialing. Check explanation in details below).

If the devices are not configured properly, then a call from external server to MOR is possible,

but only one device will be billed all the time or only one trunk will be active (it depends on

the misconfiguration).

Possible error on misconfiguration:

WARNING[2836]: chan_sip.c:8385 check_auth: username mismatch, have <XXXX>, digest has

<YYYY>

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SIP INVITE packet

As described previously, the correct SIP INVITE packet should be received in order for MOR to

authenticate the user correctly.

Good SIP INVITE packet

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 82.215.xxx.xxx:5060;branch=z9hG4bK50ef8b9b;rport

From: "39081761xxxxx" <sip:[email protected]>;tag=as3b117c95

To: <sip:[email protected]>

Contact <sip:[email protected]>

Call-ID: [email protected]

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Fri, 19 Feb 2010 12:28:03 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 242

Bad SIP INVITE packet

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 82.215.xxx.xxx:5060;branch=z9hG4bK7edd0b9e;rport

From: "3908176xxxxx" <sip:[email protected]>;tag=as1b7e883e

To: <sip:[email protected]>

Contact: <sip:[email protected]>

Call-ID: [email protected]

CSeq: 103 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Proxy-Authorization: Digest username="1028", realm="asterisk", algorithm=MD5,

uri="sip:[email protected]", nonce="0f96c765",

response="df7b502821c03ad43222d007b20dbf84", opaque=""

Date: Fri, 19 Feb 2010 12:17:41 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 242

Example

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Let's say we have MOR and another Asterisk server from which we want to configure two

connections to MOR.

MOR configuration

First, on MOR we need to describe such connections as Devices for different Users.

We will create two devices, 1028 and 1009, for these connections:

(Important parts are marked in RED)

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Remote server configuration

The remote server in our example is Asterisk, so its configuration should be in sip.conf file,

which describes both connections and registers to MOR:

[general]

register => 1028:[email protected]/1028

register => 1009:[email protected]/1009

[authentication]

[mor1028]

canreinvite=no

host=82.215.xxx.xxx

user=1028

username=1028

secret=ewpjxxxx

type=peer

fromuser=1028

[mor1009]

canreinvite=no

host=82.215.xxx.xxx

user=1009

username=1009

secret=ca5hxxxx

type=peer

fromuser=1009

Limitations

It is not possible to send CallerID from a remote server this way, because "fromuser"

overwrites the CallerID field.

You can only hardcode CallerID in MOR for every user account.

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How to forward calls when device is offline

1. Go to SETTINGS -> Users -> Devices and from opened device list choose a device.

Click icon which will lead you to Call Flow page.

2. In the Call Flow window choose Failed and click icon:

3. In the Action dropdown choose Forward and click Change:

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4. In Data section select whether you want it to be forwarded to a local device (choose one

from the dropdown) or an external number (enter the number), choose CallerID settings and

click Change:

Call flow is ready!

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