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MOR v0.6 Manual KolmiSoft 2006-2008 1 MOR v0.6 MANUAL
MOR v0.6 Manual
KolmiSoft 2006-2008 1
MOR v0.6 MANUAL
MOR v0.6 Manual
KolmiSoft 2006-2008 2
Table of Contents
Introduction .......................................................................................................................................................................... 12
Who is this product designed for ..................................................................................................................................... 12
Supported Business Models ............................................................................................................................................. 13
Concepts ............................................................................................................................................................................... 14
Asterisk Realtime .............................................................................................................................................................. 14
VAT .................................................................................................................................................................................... 14
E.164 format ..................................................................................................................................................................... 14
Rounding ........................................................................................................................................................................... 15
Setting up VSP ....................................................................................................................................................................... 15
Configuration ........................................................................................................................................................................ 16
mor.conf ........................................................................................................................................................................... 19
environment.rb ................................................................................................................................................................. 19
Logo change ...................................................................................................................................................................... 20
VoiceMail configuration ................................................................................................................................................... 20
Currencies ......................................................................................................................................................................... 21
Translations ...................................................................................................................................................................... 22
Billing .................................................................................................................................................................................... 23
Concepts ........................................................................................................................................................................... 23
What is increment ........................................................................................................................................................ 24
Links .................................................................................................................................................................................. 25
Logic .................................................................................................................................................................................. 25
Providers ........................................................................................................................................................................... 26
Provider's settings ........................................................................................................................................................ 28
Registration ................................................................................................................................................................... 30
Several IPs from Provider ............................................................................................................................................. 30
How to create Zap/H323 providers .............................................................................................................................. 30
H323 connection configuration example ..................................................................................................................... 31
Broadvoice example ..................................................................................................................................................... 32
Provider with ANI .......................................................................................................................................................... 34
LCR .................................................................................................................................................................................... 35
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LCR Logic ....................................................................................................................................................................... 37
Directions and Destinations .............................................................................................................................................. 39
Destination Groups ....................................................................................................................................................... 40
Tariffs ................................................................................................................................................................................ 42
Day setup ...................................................................................................................................................................... 43
Advanced rates for users .............................................................................................................................................. 44
Custom rates ................................................................................................................................................................. 46
Entering rates for user retail tariff ................................................................................................................................ 47
Entering rates for provider tariff .................................................................................................................................. 49
User wholesale tariff ..................................................................................................................................................... 51
Rate import from CSV ................................................................................................................................................... 51
Destination to create from CSV .................................................................................................................................... 57
Importing Tariffs from CSV with wrong Regional Settings ........................................................................................... 57
Tariff transformation .................................................................................................................................................... 65
Export to CSV/PDF ........................................................................................................................................................ 66
Sell minutes to only one country .................................................................................................................................. 66
Tariff analysis ................................................................................................................................................................ 66
DIDs ................................................................................................................................................................................... 67
How DIDs are handled? ................................................................................................................................................ 71
Usable DIDs from providers .......................................................................................................................................... 72
DID billing ...................................................................................................................................................................... 73
Example - Configuring DID to ring some Device ........................................................................................................... 74
Number manipulation ...................................................................................................................................................... 78
Localization ................................................................................................................................................................... 79
DID handling by Localization ......................................................................................................................................... 82
Provider Rules ............................................................................................................................................................... 83
Workflow to setup Localization and Provider Rules ..................................................................................................... 84
CDR ................................................................................................................................................................................... 86
Import ........................................................................................................................................................................... 86
Export ............................................................................................................................................................................ 88
Rerating ......................................................................................................................................................................... 89
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Users ................................................................................................................................................................................. 91
User details ................................................................................................................................................................... 92
User blocking ................................................................................................................................................................ 93
Hide users ..................................................................................................................................................................... 93
Balance and credit ........................................................................................................................................................ 94
Grouping ....................................................................................................................................................................... 95
Devices .............................................................................................................................................................................. 99
Settings ....................................................................................................................................................................... 100
Which codecs should I choose for devices? ............................................................................................................... 103
Process SIPCHANINFO ................................................................................................................................................. 104
How to create Zap/H323 devices ............................................................................................................................... 105
Extension lines ............................................................................................................................................................ 105
CLIs .............................................................................................................................................................................. 106
Trunks ......................................................................................................................................................................... 106
Fax device ................................................................................................................................................................... 107
Virtual device .............................................................................................................................................................. 107
Device groups ............................................................................................................................................................. 107
Call Flow .......................................................................................................................................................................... 109
Forward ....................................................................................................................................................................... 110
Voicemail .................................................................................................................................................................... 110
Fax detect ................................................................................................................................................................... 111
Additional functionality ...................................................................................................................................................... 112
Online registration .......................................................................................................................................................... 112
Recordings ...................................................................................................................................................................... 115
Call Shop functionality .................................................................................................................................................... 117
Setup ........................................................................................................................................................................... 117
Operation .................................................................................................................................................................... 120
Phonebook ...................................................................................................................................................................... 122
Global PhoneBook ...................................................................................................................................................... 122
Simple PhoneBook ...................................................................................................................................................... 123
Using phonebook ........................................................................................................................................................ 123
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Fax2Email ........................................................................................................................................................................ 124
Mass mailing ................................................................................................................................................................... 126
Dial plans......................................................................................................................................................................... 128
Auth. by PIN ................................................................................................................................................................ 128
Auth. by ANI ................................................................................................................................................................ 130
Calling Card dial-plan .................................................................................................................................................. 131
Callback ........................................................................................................................................................................... 132
Common Callback diagram ......................................................................................................................................... 132
Implementation in MOR ............................................................................................................................................. 132
Who pays for not authenticated call? ........................................................................................................................ 133
Callback User, Callback Device and Callback Tariff ..................................................................................................... 133
Types of Callback in MOR ........................................................................................................................................... 133
Callback with Device's PIN authentication/authorization .......................................................................................... 134
Callback Using ANI (Automatic Number Identification) ............................................................................................. 134
Callback By Calling Card's Number/PIN (or only PIN) ................................................................................................. 135
Setting Callback in MOR .............................................................................................................................................. 136
Step-by-step example ................................................................................................................................................. 137
WEB Callback .................................................................................................................................................................. 141
Callback from HTTP request ........................................................................................................................................... 141
Click2Call ......................................................................................................................................................................... 142
Quick Forwards ............................................................................................................................................................... 143
Accounting .......................................................................................................................................................................... 144
Services ........................................................................................................................................................................... 144
Subscriptions................................................................................................................................................................... 144
Invoices ........................................................................................................................................................................... 145
Invoice by CallerID ...................................................................................................................................................... 147
Manual payments ........................................................................................................................................................... 147
Vouchers ......................................................................................................................................................................... 149
ERROR: Vouchers are not accepted ............................................................................................................................ 150
PayPal payments ............................................................................................................................................................. 151
WebMoney payments .................................................................................................................................................... 152
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Reporting ............................................................................................................................................................................ 153
Provider's calls ................................................................................................................................................................ 153
Provider's stats ............................................................................................................................................................... 154
User's calls ...................................................................................................................................................................... 155
User call's stats ............................................................................................................................................................... 155
New call's stats ............................................................................................................................................................... 157
Loss making calls ............................................................................................................................................................. 158
Hangupcause codes ........................................................................................................................................................ 159
Payment stats ................................................................................................................................................................. 160
Finances .......................................................................................................................................................................... 161
Profit stats ....................................................................................................................................................................... 162
Country stats ................................................................................................................................................................... 163
Fax2Email stats ............................................................................................................................................................... 164
Realtime user stats ......................................................................................................................................................... 165
How to view user's stats in Realtime Stats ................................................................................................................. 165
Realtime Stats Usage .................................................................................................................................................. 166
Active Calls ...................................................................................................................................................................... 167
User login stats ............................................................................................................................................................... 167
Quick stats ...................................................................................................................................................................... 168
Addons ................................................................................................................................................................................ 169
Calling Cards ................................................................................................................................................................... 169
Short description ........................................................................................................................................................ 169
Differences between Users and Cards in MOR ........................................................................................................... 169
Work-flow ................................................................................................................................................................... 170
Card Groups ................................................................................................................................................................ 171
Card Group Management ........................................................................................................................................... 172
Cards in CSV file .......................................................................................................................................................... 172
Calling Card Groups Image.......................................................................................................................................... 173
Cards ........................................................................................................................................................................... 173
Adding new Cards ....................................................................................................................................................... 173
Create Dial Plan........................................................................................................................................................... 174
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Assign DID to Calling Card Group Dial Plan ................................................................................................................. 175
Online Calling Card Shop ............................................................................................................................................ 176
Calling Card Owner portal ........................................................................................................................................... 177
Step-by-Step example ................................................................................................................................................. 178
Auto-Dialer ...................................................................................................................................................................... 183
User interface ............................................................................................................................................................. 183
Numbers ..................................................................................................................................................................... 184
Actions ........................................................................................................................................................................ 185
Admin interface .......................................................................................................................................................... 186
Reseller Addon ................................................................................................................................................................ 187
Description .................................................................................................................................................................. 187
DIDs ............................................................................................................................................................................. 187
PayPal .......................................................................................................................................................................... 187
Manual ........................................................................................................................................................................ 187
Click2Call Addon ............................................................................................................................................................. 188
Concepts ..................................................................................................................................................................... 188
System setup ............................................................................................................................................................... 189
Call destination control ............................................................................................................................................... 190
Call .............................................................................................................................................................................. 190
Failed call handling ..................................................................................................................................................... 190
Statistics ...................................................................................................................................................................... 190
API ....................................................................................................................................................................................... 191
Login................................................................................................................................................................................ 191
login ............................................................................................................................................................................ 191
Logout ......................................................................................................................................................................... 191
Callback ........................................................................................................................................................................... 192
callback ....................................................................................................................................................................... 192
c2c_callback ................................................................................................................................................................ 193
Accounting ...................................................................................................................................................................... 194
invoices ....................................................................................................................................................................... 194
Other ................................................................................................................................................................................... 195
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Branding .......................................................................................................................................................................... 195
Security ........................................................................................................................................................................... 195
How to change admin password ................................................................................................................................ 195
Change DB name......................................................................................................................................................... 196
Various ............................................................................................................................................................................ 196
How to restart Asterisk server .................................................................................................................................... 196
Asterisk CLI .................................................................................................................................................................. 197
App mor.so version ..................................................................................................................................................... 197
Ports which should be opened ................................................................................................................................... 197
Asterisk under NAT ..................................................................................................................................................... 197
Useful links ...................................................................................................................................................................... 198
Guides ................................................................................................................................................................................. 199
PBX connection to MOR ................................................................................................................................................. 199
Common connection diagram .................................................................................................................................... 199
Dialing from PBX ......................................................................................................................................................... 200
Dialing to PBX .............................................................................................................................................................. 200
MOR and Trixbox ............................................................................................................................................................ 201
Connection diagram.................................................................................................................................................... 201
Dialing from phone to MOR ........................................................................................................................................ 202
Dialing to phone using DID ......................................................................................................................................... 207
Final notes ................................................................................................................................................................... 209
Configuring more phones ........................................................................................................................................... 210
Limitations .................................................................................................................................................................. 210
Troubleshooting .................................................................................................................................................................. 211
Integrity Check ................................................................................................................................................................ 211
Destinations not assigned to Destination Groups ...................................................................................................... 211
Call Tracing ...................................................................................................................................................................... 212
User testing ................................................................................................................................................................. 212
Device testing ............................................................................................................................................................. 215
Asterisk related errors .................................................................................................................................................... 218
SIP/2.0 401 Unauthorized ........................................................................................................................................... 218
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SIP/2.0 403 Forbidden ................................................................................................................................................ 218
SIP/2.0 404 Not Found ................................................................................................................................................ 218
SIP/2.0 407 Proxy Authentication Required ............................................................................................................... 218
SIP/2.0 486 Busy Here................................................................................................................................................. 219
SIP/2.0 503 Service Unavailable ................................................................................................................................. 219
SIP/2.0 603 Declined ................................................................................................................................................... 219
Call rejected by IP: No authority found ...................................................................................................................... 219
CallerID not passed ..................................................................................................................................................... 220
DID is not working ....................................................................................................................................................... 220
Fax is not received ...................................................................................................................................................... 221
MOR can't determine who is calling. Make sure accountcode is set for caller (Provider or Device)......................... 221
Retrieved 0 adv.rates, max adv.rate: 0.000000, total event price: 0.000000 ............................................................ 222
One-way audio or not audio at all .............................................................................................................................. 222
No sound on Voicemail or Playback ........................................................................................................................... 222
Dropping extra frame of G.729 since we already have a VAD frame at the end ....................................................... 222
Comfort noise support incomplete in Asterisk (RFC 3389) ........................................................................................ 222
mysql_reconnect: mor: Unknown connection error: (2006) MySQL server has gone away. .................................... 223
Call dropped upon connection ................................................................................................................................... 223
Starting asterisk: Cannot find your TTY ...................................................................................................................... 223
False ANSWER ............................................................................................................................................................. 224
Unable to allocate RTCP socket: Too many open files in system ............................................................................... 224
Echo ............................................................................................................................................................................ 225
Asterisk stops working with Internet loss ................................................................................................................... 225
Can't register device ................................................................................................................................................... 225
Registration from 'SOME DEVICE' failed for 'IP' - Not a local domain ........................................................................ 226
Unable to find key 'XXX' in family 'SIP/Registry' ......................................................................................................... 226
Registration from 'sip:[email protected]' failed for '34.56.67.89' - No matching peer found ................................... 226
Device can't dial out ................................................................................................................................................... 226
No path to translate from XXX to YYY ......................................................................................................................... 226
Call rejected by IP: No such context/extension .......................................................................................................... 226
Mor authorize: Too low balance for more simultaneus calls!!! ................................................................................. 227
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Device can't be reached .............................................................................................................................................. 227
Unable to create channel of type 'XXX' (cause 3 - No route to destination) .............................................................. 227
Device is not receiving video ...................................................................................................................................... 227
Zaptel YELLOW alarm .................................................................................................................................................. 228
Unable to open master device '/dev/zap/ctl' ............................................................................................................. 228
GUI problems .................................................................................................................................................................. 229
We're sorry, but something went wrong ................................................................................................................... 229
Troubleshooting .......................................................................................................................................................... 229
Check GUI log .............................................................................................................................................................. 229
Forbidden You don't have permission to access /mor/ on this server. ...................................................................... 230
LoadError (Expected /home/mor/app/controllers/callc_controller.rb to define CallcController) ............................ 230
LoadError (Expected /home/mor/app/controllers/stats_controller.rb to define StatsController) ........................... 230
rrno::EACCES (Permission denied - /tmp/mor_debug.txt) ......................................................................................... 230
NoMethodError (undefined method `in_place_edit_for' for CallcController:Class) .................................................. 231
FCGI info is shown in text format ............................................................................................................................... 231
Errno::EACCES (Permission denied - /tmp/mor_cf_xxxx or /var/spool/asterisk/outgoing/mor_cf_xxxx) ................ 232
ActiveRecord::RecordNotFound (Couldn't find User with ID=0) ................................................................................ 232
Interlaced PNG images are not currently supported .................................................................................................. 232
No route matches "/" with {:method=>:get} .............................................................................................................. 232
uninitialized constant Gem::GemRunner ................................................................................................................... 232
Folder content is shown ............................................................................................................................................. 233
No such file or directory - /tmp/mysql.sock ............................................................................................................... 233
NoMethodError (undefined method `exchange_rate' for nil:NilClass) ...................................................................... 233
TypeError (PNG alpha channel transparency is not supported; only palette transparency is supported.) ............... 233
ActiveRecord::StatementInvalid (Mysql::Error: Got a packet bigger than 'max_allowed_packet' bytes: ................. 233
Can't open phpMyAdmin ............................................................................................................................................ 234
Email problems ............................................................................................................................................................... 235
Default settings ........................................................................................................................................................... 235
Net::SMTPAuthenticationError (535 auth failure) or Net::SMTPAuthenticationError (535 invalid user ID or
password) ................................................................................................................................................................... 235
NameError (undefined local variable or method `xxxxxx' for #) ................................................................................ 235
Timeout::Error (execution expired) /usr/lib/ruby/1.8/timeout.rb:54:in `open' ........................................................ 236
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SocketError (getaddrinfo: Name or service not known) ............................................................................................ 236
501 Syntactically invalid HELO argument(s) ............................................................................................................... 236
Can't initiate web callback .............................................................................................................................................. 237
Installation errors ........................................................................................................................................................... 237
Rails: command not found .......................................................................................................................................... 237
Yaml.rb:133: BUG Segmentation fault........................................................................................................................ 238
Various ............................................................................................................................................................................ 239
Asterisk crashing ......................................................................................................................................................... 239
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Introduction KolmiSoft VoIP Billing Software MOR PRO is an advanced web-based application 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 Billing Software integrates the core AAA functions Authentication, Authorization, and Accounting with vital functions that facilitate the effective management of all billing-related processes:
Customers
Services
Finance
Services distribution
Reporting
Traffic partners
User rights
On the business side, KolmiSoft Software 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 Software the absolute billing and customer care solution for Voice Over Broadband/ IP businesses worldwide.
Who is this product designed for KolmiSoft VoIP Billing Software 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 VoIP Billing System 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
Call Shop
Corporate Clients
Residential Clients
Internet services
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Concepts Some concepts should be explained before going further.
Asterisk Realtime This feature is very well described in www.voip-info.org website here. Please read about it.
We can tell that MOR uses Asterisk Realtime to configure everything – users/devices/providers/etc. All configuration is in the database. From it, a very important rule follows:
VERY IMPORTANT RULE: MOR DOES NOT UPDATE/EDIT .CONF FILES! NO NEED
TO EDIT THEM! As with every rule – this one has exceptions. Provider registration and Zap/H323 providers/devices should be
configured in .conf files. But everything else is in the database.
VAT
The principles of how MOR handles VAT are explained here. All prices and rates in MOR are without VAT. That‟s a rule. Each user has VAT percent. By that VAT percent he gets charged when he makes payments or views his rate.
For example, if user wants to increase his balance by PayPal and enters amount $10.00 and his VAT is 17.5%, so total he will have to pay $11.75 in PayPal. His balance will be updated by $10.00 and $1.75 will be counted as VAT.
E.164 format What is E.164 format number? In short it's the number without international prefix which starts with country code, for more details check Wikipedia. Every destination inside MOR should be in this format. That's one of the most important rules of the MOR. If
destination is not in E.164 format – it can't be billed correctly. All DIDs in MOR should be in this format also. Do not include international prefixes for DIDs. Enter them in E.164 format.
Example: For instance, the local number 08 6311 3000 in Australia would be converted to E164 as 61863113000.
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Rounding Rounding in MOR GUI is done only when showing final result. All values are not rounded in database. Example: 0.027 + 0.027 = 0.054 rounded is 0.05, if you are rounding each step, then you get: 0.03 + 0.03 = 0.06 which is not so right as 0.05.
Setting up VSP
In order to start using MOR it should be properly configured. Here we will cover main steps which should be done after fresh MOR installation.
These steps covers basic VSP (VoIP Service Provider) configuration. After these steps fully configured system will be ready to use.
1. Configuration of GUI – change logo/company name/other details 2. Add providers 3. Create Provider Rules for your Providers 4. Create LCR and add your Providers to these LCR 5. Create tariffs for providers and users
6. Populate tariffs with rates
7. Create Locations and rules for these Locations 8. Create Users 9. Assign LCR/Tariffs for users 10. Create Devices for users 11. Configure real devices and connect them to the server 12. Use Call Tracing to test 13. Resolve any problems 14. Make test calls
These are the steps for basic outgoing call configuration.
All other functionality is additional.
All these steps are covered in following chapters of this manual.
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Configuration Most of MOR configuration is saved in the database. Some part in environment.rb file. It is very easy to change configuration using GUI.
You can find configuration in SETTINGS – Setup – Settings. Settings are divided amongst the groups:
Global – representative values for GUI mostly
Registration – settings for online registration
Invoices – invoice formatting settings
Vouchers – settings for vouchers
WEB Callback – settings for WEB Callback
PayPal - settings for PayPal
Emails – email sending options and SMTP server's connection properties
Various – all other configuration
Global
Company – Name of the company
Company email – email 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
Registration
Enabled? – do we want to let users to register online automatically?
Tariff for new users – select tariff for just registered users
LCR – select LCR for just selected users
Default VAT percent – what VAT percent should be for just registered users
Default country
Asterisk server IP – what IP user should connect his softphone/server
Default CallerID name – the CID name to send for new users
Default CallerID number – the CID number to send for new users
Send email to user after registration – just yes or no
Send email to admin after registration – just yes or no
Allow user to enter VAT – some want to let users do this
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Invoices
Number start – how 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
Address format – changes field positioning in the ivoice for the address
Address line 1-4 – Your address info
Bank details line 1-5 – Your bank details info
End title – some title for your invoice's end
Vouchers
Vouchers enabled – yes or no
Number length – how long vouchers number should be – make it at least 10 digits long
Disable time (minutes) – if user several times fails to enter correct voucher number – for how long should we
disable posibillity for him to try again
Attempts to enter – how many time user can try to enter voucher number
WEB Callback
Active? – enabled or disabled
Temporary directory – where call files should be stored – not possible to change – just for info
Spool directory – where call files should be sent in order to activate them – not possible to change - just for
info
MaxRetries – how many time try to redial if call failes
RetryTime – in what intervals (in seconds) try to redial
WaitTime – timeout value for the call – how long to wait till the call is answered
PayPal
PayPal Enabled – yes or no
PayPal Email – account in PayPal where you want to get payments
Default currency – select currency which matches you 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 – minimal amount user can send you
Test – if checked all amounts will be equal to 0.01 – good for testing with real account
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Emails
Sending enabled – if checked you will be able to use Mass Mailing and system will send emails upon user
registration
SMTP server – hostname of the server
Domain – you can leave it localhost.localdomain
Login – username to your SMTP server
Password – password to your SMTP server
Batch size – value for Mass Mailing – how many emails to send at once
Various
Click2Call active? – yes or no
User wholesale tariff enabled – yes or no
Days for DID to close – when user uses DID and stops using it, we reserve this DID for this user for this
amount of days in case user will change his mind and wants this DID back
Agreement number length – user agreement number length
Number digits – how many digits to show in numbers in the GUI and PDF reports
Items per page – value for GUI, active in all pages
Device PIN length – length of device's PIN
Fax device enabled – enable if you want Fax2Email functionality
Change Zap – on or off. Sometimes it is wanted 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.
Zhange Zap to – to what value change "Zap" in GUI
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mor.conf This file is in /etc/asterisk/ and describes some values for app_mor.so application which is the main MOR core working in Asterisk environment. hostname, dbname, user, password, port, sock – all these values describes connection to database;
min_frozen – Minimum credit left to allow simultaneous calls. If users balance is lower – he will not be able to make simultaneous calls (default 10); frozen_time – How much balance to freeze in minutes if user is prepaid (default 30). If user makes several simultaneous calls and he is prepaid – we need to „freeze“ some amount to be safe;
show_notice, show_warning, show_error – sets debug level for app_mor.so application in Asterisk CLI; zap_change – change ZAP into something else in Asterisk to let use 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
environment.rb Some important configuration is in /home/mor/config/environment.rb file. Debug_File = "/tmp/mor_debug.txt" - file in which some debug info is placed
Please_Login_Context = "please_login" - do not change this Default_Context = "mor" - do not change this Recordings_Folder = "http://<YOUR IP>/mor/recordings/" – change YOUR IP to your IP
Actual_Dir = "/home/mor" - actual dir where mor resides
Web_Dir = "/mor" - Virtual dir for Apache Web_URL = "http://<YOUR IP>" - change to your IP Main_User_ID = 0 – System administrators ID – do not change it CC_Active = 0 - enable/disable Calling Card module. If it's not installed – no need to enable CC_Single_Login = 1 - Let user enter Card's number and PIN in one line when he wants to login
AD_Active = 0 - enable/disable Auto-Dialer module. If it's not installed – no need to enable AD_sounds_path = "/var/lib/asterisk/sounds/mor/ad" - AD file's folder It is recommended to change only Recordings_Folder and Web_URL and leave all other values as it is. IMPORTANT! Rules how to work with environment.rb file:
1. Before editing environment.rb file make backup
2. Edit environment.rb file and save it
3. Restart apache
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4. Check for changes
5. If everything ok – proceed to other changes or finish work with this file
6. If you get some error on GUI – means you did some mistake.
a. Copy backup back
b. Restart Apache
c. Start from 2.
Logo change Go to SETTINGS - Setup - Logo and browse for your new logo. Relogin to see your new logo.
VoiceMail configuration This section can be found in SETUP - Settings - VoiceMail
VoiceMail Extension - describes by which local extensions local user can dial to retrieve his voicemail (system
will ask him to enter his voicemail password which can be set in device's details)
If you want to setup VoiceMail (VM) on different server, then following settings are important:
VoiceMail Server active - turns VM server on/off, when on - all VM request will go to distabd VM server, when
off - VM will be handled locally on Asterisk
VoiceMail Server connection/device - describes connection to VM server
VoiceMail Retrieve Extension - which extension to dial on distant VM server to retrieve voicemail for some
extension
Example:
Let's say we have our server on IP 23.45.67.89 connected with SIP to our Asterisk.
We create separe 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 = 999
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First action is when VM is recorded
So now when VM server is ON and user with device 201 get's a VM, call is redirected like this: Dial(SIP/vm_device/201)
That means server on 23.45.67.89 will receive call to extension 201. Server SHOULD BE configured that way, that it will process VM correctly when it receives such call.
Second action - retrieving VM
When user want to retrieve his VM on distant machine he dials VoiceMail Extension. He is redirected like this: Dial(SIP/vm_device/999#201)
Again - distant server should know how to process such a request. Let's say we have distant VM server based on Asterisk. So in our example it's 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
Currencies Multicurrency is implemented in some parts of the MOR. Mainly it is for Tariffs. E.g. it is possible to have tariffs in different currencies. Currencies can be configured in SETTINGS – Setup – Currencies:
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At the bottom you can enter new Currency by pressing icon.
Next to each icon you can press to edit selected currency or to delete it. IMPORTANT! Be sure you enter correct exchange rate for all currencies. Note that Exchange rate ==
Default currency DIVIDED by selected currency. Not vice-versa! If you want to change Default currency – just rename the existing default currency to new one. Do not forget to recalculate and fix exchange rates for all other currencies!
Translations If you want to change translation of MOR GUI you can press 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 – Translations. Here you will see window with all available translations. In the first column you can Drag & Drop translation to its new place. This column is dedicated to manage the order of
translations.
Second column is for Activating/deactivating translation. Simply click on desired translation to activate/deactivate it. When it's disabled – it is in grey color.
If you want to see changes instantly press icon at the bottom. Page will be reloaded and you will not need to
logout/login to see changed translation order at the upper right corner of your window where translations are represented as flags. The first translation is DEFAULT translation. If you want to change default translation – Drag&Drop other
translation to the very top of translations list.
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Billing
For the introduction to the billing process in MOR let's review following items:
Concepts
Links - how concepts depend on each other
Logic - how these concepts work together
Concepts This section is dedicated entirely for billing system. Here is configured the actual call-flow from user to provider(s). Billing system is quite complex and there are a lot of concepts to grasp.
There are main parts from which billing in MOR consists. The most important are:
Providers
LCR
Directions
Destinations
Rates
Tariffs
Users
Devices
Providers - In old PSTN world they are called Trunks. These are your routes to outside world (out of your VoIP network). Providers can be several types: Zap, SIP, IAX2, and H323. Type determines which technology is used to connect to Provider.
LCR - Least Cost Route. It's an entity which describes how calls should go through several Providers and in which order. Order can be determined by calls route's price, quality or manual priority.
Directions - In other words - Countries. Could also be satellite networks. They just describe direction to which the call is routed.
Destinations - these are more concrete concepts of the calls course. Let's say we are dialing number 37068547771. From this number MOR knows, that Direction is Lithuania, but Destination is Mobile network Bite. Destinations are determined by Prefix. Each destination has unique Prefix. The Prefix from the example is 370685.
Destination groups - in order to make life easier and to manage prices for destinations, destinations are grouped. And one price applies for a group of destinations not for one destination separately. (Group can contain one destination also.) Destinations groups are used for users, not for providers, because providers need to have very
exact rates for each destination.
Rates - these are the prices for Destinations. They have many properties from which the way they are calculated
depends.
Tariffs - they are groups of Rates. Sometimes they are called Price Lists.
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Users and devices - User is the actual user of the system – the client which dials out or receives calls. User can represent one person or whole company. User is main unit in the system. Most of the settings are associated with user. Each user can have many devices. Device represents end-point where voice traffic should be sent. As example device can represent:
VoIP phone
VoIP softphone
PSTN phone (connected through FXS port)
What is increment
Increment is amount of time in which the call will be billed. It is counted in seconds.
Example:
Imagine call length is 15s
Rate for 1 min is 1 EUR/min (for simplicity reasons)
Then:
If increment is 1, then call length will be counted as 15s and call price will be 0.25 EUR (1/60*15) - increment is 1, so call can be 1, 2, 3... in length. 15s is exact length of call.
If increment is 10, then call length will be counted as 20s and call price will be 0.333.. EUR (1/60*20) - increment is 10, so call can be 10, 20, 30... in length. 20s is nearest to 15.
If increment is 20, then call length will be counted as 20s and call price will be 0.333.. EUR (1/60*20) - increment is 20, so call can be 20, 40, 60... in length. 20s is nearest to 15.
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Links
There are links between main concepts of MOR:
LCR can have many Providers
Direction can have many Destinations
Tariff can have many Rates
Provider can have 1 Tariff
User can have 1 Tariff
User can have 1 LCR
Destination can have 1 Rate for special Tariff
Logic
In order to successfully use MOR for dialing out and billing some rules must be met.
Best way to show this is by example.
User A dials number trying to reach User B.
System takes User's A Tariff and LCR.
Then takes all Providers which belong to that LCR.
For every Provider system takes their Tariff.
From every Tariff based on Prefix - Rate is taken.
If LCR's type is 'Price' then all Providers are sorted that first Provider would have lowest Rate to that Destination.
Then system tries to dial number using first Provider.
If call successful - system does billing and finishes session.
If call unsuccessful - system tries to dial next Provider in a row.
It repeats until call is successful or no Providers left.
The whole process in MOR is much more complex. This example covers bare minimum to take grasp of how the billing
system works.
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Providers
IMPORTANT NOTE: MOR uses Asterisk Realtime – that means all configuration
is saved into database. Config files are not modified. After creating provider or device – do not search for them in .conf files – they are in database. Commands
such as 'sip show peers' do not work with Asterisk Realtime (when caching is turned off – what is recommended).
In old PSTN world they are called Trunks. These are your routes to outside world (out of your VoIP network). Providers can be several types: Zap, SIP, IAX2, and H323. Type determines which technology is used to connect to Provider.
At the very start you have to create Providers through which you will dial out to PSTN or other VoIP networks. Every Provider will charge you some money for calling through him (rate could be 0 also) to various destinations. And to every destination rate is different. Following this logic we need to have Tariff (or Price List) for Provider. This Tariff will tel l our billing what price we pay to some destination if we are using this Provider's services. In other words - that would be our Selfcost.
Providers' configuration can by found in SETTINGS – Billing – Providers.
- shows Provider Rules
- Provider calls
- This icon let's you edit Provider's settings
- This icon let's you delete Provider if it is not used in any LCR
In order ro create provider press on New provider:
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Here enter the name for new provider, select its type, tariff and press Create. NOTE: You can't create new Provider if there're no Tariffs available for Providers. You will go to Provider's settings page. There are a lot of settings. We will describe them here section by section.
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Provider's settings
General
Name – name for provider, for informational purposes
Technology – SIP/IAX2/ZAP/H323 – choose which one is used by your provider, e.g. in which way do you
connect to it
Tariff – the rate list provider charges you
DTMF Mode – available options: inband, info, RFC2833, auto – choose which one your provider uses
Location – which Localization rules set to apply to INCOMING calls COMING FROM this provider
Device ID – informational data
Timeout - This setting lets set how long this provider should be dialed before giving up. Default value is 60
seconds. Minimum value is 30 seconds.
Authentication
For ZAP providers:
Channel – which channel (or channel group) to use on PRI/BRI/PSTN card. Channels and groups should be
configured in zapata.conf.
For all other providers:
Login – username if your provider asks for it
Password – password used for authentication by your provider
Or Authenticate by IP – just check the checkbox
o Make sure you put correct Hostname/IP adress!
CallerID
CallerID – consists of two parts: name and number. The "Number" part gets transferred by default by all technologies: GSM, PSTN, SIP etc, the "Name" part only on some. The number you see on your mobile phone when
somebody is calling you is "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 Provider. It should be like this almost always.
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Network related
MOR v0.4 Manual-2007 18.
Hostname – hostname of the provider
Server IP – provider's IP, the value can be "dynamic", which means that provider can change its. To discover
this value do ping to the provider's hostname.
Port – port used to connect to provider (default: 5060 for SIP, 4569 for IAX2)
Media control – canreinvite/transfer – do you want your server to stay in media path between your clients and
your provider
NAT – available options: yes, no, never, route. For detailed explanation about these settings please refer
here.
Qualify – how often "ask" provider about availability. More details here for SIP providers and here for IAX2
providers.
Codecs
Choose the codecs your provider uses.
NOTE: When no fields are checked – that means that all codecs are available. E.g. settings in sip.conf or iax.conf are effective.
Allowed addresses
IP, MASK – permit traffic from these IP only. You can find detailed explanation here.
Advanced
Fromuser – info here.
Fromdomain – info here.
Trustrpid – info here.
Sendrpid – info here.
Insecure – info here.
Progressinband – info here.
Video support – does your provider support Video over IP? More info here.
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Registration
If provider asks you to register to your server, you need to do that in sip.conf or iax.conf file manually. Use register line: register => username:password@server:port Save conf file and reload Asterisk.
To check for registration enter:
sip show registry or iax2 show registry. Add register… line in [globals] section. If you add it to the end of file - it will not be active.
If IP of your device can change - then you need to register. You don't need to register when you know static IP's of your devices/providers.
Several IPs from Provider
If your provider providers different IP/PORT for Incomming calls - you should create such device with these settings in
/etc/asterisk/sip.conf (for SIP).
How to create Zap/H323 providers
Zap and H323 does not have Realtime support in Asterisk. In order to configure them you should do following steps:
1. Create provider in GUI
2. Write down just created provider's device accountcode which can be found in provider's edit window in Device
ID: field.
3. In zapata.conf or h323.conf (ooh323.conf or similar) file look for context field. As example (in zapata.conf):
;Sangoma A101 port 1 [slot:2 bus:2 span: 1] switchtype=euroisdn context=from_pstn group=0 signalling=pri_cpe
channel => 1-15 channel => 17-31
NOTE: context=from_pstn
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4. In extensions.conf
[from_pstn] exten => _X.,1,Set(CDR(ACCOUNTCODE)=X) exten => _X.,2,Goto(mor,${EXTEN},1)
Where X(variable) is Device ID from Provider's settings, example:
Here X == 1. (It is example! In your configuration X will have other value!)
NOTE: [from-pstn] is same context as in zapata.conf
5. Add accountcode = X to just created provider (peer/friend) in .conf file where X is the Device ID created in GUI
6. Reload Asterisk and you are ready to go!
H323 connection configuration example
Example configuration for 2 H323 gateways: MERA and QUINTUM
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/etc/asterisk/h323.conf
[global]
userbyalias=no
[mera]
type=friend
host=23.45.67.89
fastStart=no
context=from_mera
[quintum]
type=friend
host=192.168.10.5
fastStart=no
context=from_quintum
/etc/asterisk/extensions.conf
[from_mera]
exten => _X.,2,Set(CDR(ACCOUNTCODE)=X)
exten => _X.,3,Goto(mor,${EXTEN},1)
[from_quintum]
exten => _X.,2,Set(CDR(ACCOUNTCODE)=Y)
exten => _X.,3,Goto(mor,${EXTEN},1)
Note accountcodes X and Y - they should be equal to devices ID in MOR GUI to tie this configuration with MOR.
NOTE: If you hear "Congratulation from asterisk ...." message that means you forget to set context = mor in zapata.conf or h323.conf file for your provider.
Broadvoice example
As an example we will show you how to configure Broadvoice as provider. In /etc/asterisk/sip.conf in [general] section we enter: register => <phone number>:<register password>@sip.broadvoice.com save this file. Reload Asterisk.
And in MOR GUI we create new provider and enter such settings:
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Provider with ANI
Automatic Number Identification for provider is used ONLY (at least we don't know any other situations) when
provider gives you some access number after which (IMPORTANT!) after which dialer enters destination number WITHOUT PAUSES. E.g. WHOLE number is sent to provider instantly. Explanation:
Let's say provider gave you access number 1234
When somebody wants to dial some destination, lets say: 22234567890, he enters on his phone:
123422234567890
22234567890 from provider comes to your MOR box
Because Provider is with ANI, we are checking user's CLI
If CLI is found - we know which user is dialing
Before he dials out, we change CallerID of this call by CallerID found in our database - that's minor job, but it's
part of algorithm
After that 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 access number as DID, you can enter p(pause) after your DID calling to the provider and then enter destination. But that's totally different scenario. It should be configured with DID + Auth. by ANI DialPlan, because destination is send to MOR by DTMF, not by provider as in previous example.
Example:
If your DID is: 234567890 and you want to dial dst: 987654321 in your mobile phone (example) you enter:
234567890p987654321
That way 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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LCR LCR is set of Providers which should be used to send call. LCR stands for Least Cost Routing. E.g. cheapest rate is selected when trying to dial any destinaiton. If one provider fails – next one is dialed. This is completely invisible for dialing user and let's you increase the chances for user to reach destination. This function is called Fail-Over - if one Provider is not capable of delivering a call - system will try to use remaining Providers.
LCRs' window can by found in SETTINGS – Billing – LCR:
- This icon let's you edit LCR's settings.
- This icon let's you delete LCR if it is not used by any user.
Providers – shows Providers assigned to this LCR. Number in () shows how many providers are in this LCR. NOTE: Same providers can belong to different LCRs at the same time.
Press on New LCR to add new LCR:
Here enter name for new LCR and press Create.
Now when we have created our LCR, let's add some Providers to it. Press on Providers:
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Select one of available Providers from the list and press Add. There's no difference in which order we will add Providers to LCR. System will group them cheapest first and tries to dial one by one.
In field Active? you can press:
- to make this Provider active in this LCR (this provider will be used in dialing), or
- to disable this Provider and not use in LCR and in dialing
Note the reverse order of icons – you press on red cross if you want to enable ir and vice-versa. That's
not mistake. Icon represents current state of the Provider in this LCR.
You can press if you want to remove Provider from this LCR.
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LCR Logic
Example:
We have 2 providers A and B. Dialed destination is 921XXXXXXXXXX Provider A has such rates for prefixes(destinations):
92 rate 0.1
No other destination starts with 92
Provider B has such rates for prefixes:
92 rate 0.05
921 rate 0.2
So LCR in MOR will group this providers in following order:
Provider A with rate 0.1 (92)
Provider B with rate 0.2 (921)
It first dials through Provider A, if call fails, Provider B will be used to send the call.
If call is ANSWERED, BUSY or there's NO ANSWER - then call ends. And Provider B is not used. It is used ONLY when call fails - CONGESTION. In following schema LCR logic is presented in graphical way:
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Directions and Destinations
Directions - in other words - Countries. Could also be satellite networks. They just describe direction to which the call is routed.
They can be found in SETTINGS – Billing – Directions. In the main window you can see all Directions with their details:
- This icon let's you edit direction's details.
- This icon let's you delete direction if it does not have any Destinations.
New direction – add new Direction
Destinations – View all Destinations for this Direction:
Destinations - these are more concrete concepts of the calls course. Let's say we are dialing number 37068547771.
From this number MOR knows, that Direction is Lithuania, but Destination is Mobile network Bite. Destinations are determined by Prefix. Each destination has unique Prefix. The Prefix from the example is 370685.
- This icon let's you edit destination's details.
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- This icon let's you delete destination if it is not assigned to any rate.
New destination – add new Destination
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 into some Destination Group -
otherwise you will get corrupted invoices if user uses retail tariff.
QUESTION: How to import destinations from file?
ANSWER: You can import them using Tariff import script. Just create dummy tariff, add dummy rates and import. Destinations will be created automatically. Later you can delete tariff. Just remember to
enter prefixes in E164 format. NOTE: You can get CSV file with all Destinations in Tariffs window. It is suitable for rate entering and importing back
to MOR.
Destination Groups
In order to make life easier and to manage prices for destinations, destinations are grouped. And one price applies for a group of destinations not for one destination separately. (Group can contain one destination also.) Destinations groups are used 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.
The main window shows main details of Destination Groups:
- This icon let's you edit destination group's details.
- This icon let's you delete destination group if it is not have no rates
New destination group – add new Destination group
View – shows how many Destinations are grouped under this group.
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NOTE: Destination can belong to one group at the same time only.
- This icon let's you remove destination from this destination group:
Press on New destinations to add new Destinations to this Destination Group:
Check desired Destinations and press Add destinations button.
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NOTE: If you want to split some Destination Group, remove all Destinations from it, delete old Destination Group and create new ones. Then add removed Destinations to just created Destination groups.
Tariffs Tariffs are sets of rates (prices) to different destinations. There are three types of tariffs: providers wholesale (inbound), users wholesale (outbound) and users retail (outbound). The difference between wholesale and retail tariffs are in the way how rates are set for destinations. In wholesale tariffs rate can be set to every destination possible. In retail tariff destinations are grouped which lets
have less rates. They are used in different situations and each of them has their strong sides and weak. Strong side of wholesale providers is possibility to enter different rate for different destination. It is useful to set this tariff when you resell services to other providers. If your clients are retail end users – it is better to use retail tariff. They are easier to manage and understand for final
user.
Tariffs' window can by found in SETTINGS – Billing – Tariffs:
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Rates - This icon shows Tariffs' rates
Import from CSV – let's you import rates all at once from CSV file
Delete all rates – empty the tariff
Make user retail tariff – transforms provider tariff to user retail tariff with some modifications
Exports tariff's rates to PDF
Exports tariff's rates to CSV
- This icon let's you edit Tariff's settings.
- This icon let's you delete Tariff if it is not used by any User, Provider, LCR or Calling Card Group.
Press on New tariff to add new Tariff. Here just select type and enter name for the tariff.
Destinations CSV - This link in Tariffs window gets you CSV file with all Destinations/Directions. It is suitable for
rate entering and importing back to MOR.
Day setup
MOR let's you set different dates for different types of days. Days can be 2 types: work days (WD) or free days (FD).
Free days by default are counted as Saturday and Sunday. Work days are Mondays – Fridays. National holidays and other days which do not meet 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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Advanced rates for users
In order to manage rates for destinations easier, destinations are grouped. And one rate applies for a group of
destinations not for one destination separately. (Group can contain one destination also if it is necessary.) Destinations groups are used by users, not by providers because providers need to have very exact rates for every destination. Each group of destinations consists of several destinations described by prefixes. Let‟s take for example Albania MOB: here we have 3 destinations described by prefixes: 35538, 35568 and 35569. Each of these destinations is Albania MOB. Usually users do not care to which MOB network they are calling and usually for not popular countries providers
put one price for all destinations to that direction. Same here. We can put one price for all these 3 destinations grouped to group Albania MOB.
Each rate for some destination group can have many rate details. Which means: destination group has one rate, but this rate has many rate details (sub-rates). Each rate detail describes one interval of the call. If first rate detail is RD1, second: RD2, and so on, so all call is RD1 + RD2 + .. + RDn. Number is finite. Rate detail can be seen as time
interval for all duration of the call. This interval could be zero length also. It is explained further. Each rate detail consists of such parts:
FROM – from which second rate detail starts
DURATION – what is the length of this rate detail (interval)
TYPE - minute or event
ROUND BY – in what intervals round time
RATE – price for minute in this rate interval
The type 'minute' describes interval which last some time. This time is billed by rate and rounded accordingly.
Interval starts at FROM and lasts for DURATION seconds. The type 'event' describes some kind of zero-length interval which we call EVENT. As example it could be connection fee. If type is 'event' rate detail has no DURATION (it‟s always zero) and ROUND BY fields. Examples:
1. The call should have 0.2 connection fee (currency does not matter here) and all other time must be billed by
6 second intervals with rate 0.1 per minute.
Such rate should be described like this:
FROM DURATION TYPE ROUND BY RATE
1 0 event 0.2
1 ∞ minute 6 0.1
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 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
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Same example can be described like this also:
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
3. The rate for call is 0.01 per minute rounded by 1s. The max price for call can be 0.1.
FROM DURATION TYPE ROUND BY RATE
1 600 minute 1 0.1
601 ∞ minute 1 0
NOTE: The same rules apply to Custom Rates for users.
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Custom rates
Custom Rates are such rates which are different from default user's tariff.
Example: if you have users A and B, and they both have tariff 'GOLD', but customer A has rate to Russia 0.5 and other user has tariff to Spain for 0.6 ('GOLD' rates for these destinations are 0.7 and 0.8), so you just put new values for each customer in their Custom Rates and they will be used instead of default tariff rates. This helps to assign different rates for diferent customers not creating new tariff for everyone.
Most of the time customers are interested in calling only to few destinations. Nobody needs complete A-Z destinations.
You can reach Custom Rates by pressing in Users window:
From drop-down list select the Destination Group you want to create Custom Rate and press :
Now this user has Custom Rate to Australia MOB, but rate for this destination groups is 0.0. To fix it you need to
press on icon. Here you can create Custom Rate Details. The rules to create them are the same as for Advanced
rates for user tariffs.
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Entering rates for user retail tariff
Each user must have a tariff. User pays the rates described in his tariff. In order to add/edit rates for user retail tariff
press on Rates next to tariff.
Here you can enter Rate and Round by values for every rate. If you finished entering what you want, press
at the bottom of the page. All rates will be updated.
View – shows you the Destinations which are grouped to this particular group.
Press on and you will be able to edit rates in Advanced mode:
Here you can press on Split to Work Days and Free Days to do what it tells:
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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 time in the day. For that we need to create hour interval. In the column
End Time select some time and press on to do this:
Now we can set different rate for Free Day period from 00:00:00 till 12:59:59 and from 13:00:00 till 23:59:59.
In order to set rate press on icon:
And here you should add/edit/delete Advanced rates as described in Advanced Rates.
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Entering rates for provider tariff
This method is used to enter rates one by one by hand. It is primary used for editing rates because it's very time consuming to enter thousands of rates by hand. For such purpose CSV import is available.
Press on Rates:
In this window you can view already entered rates. If you want to add new rate press on New rate:
In this window enter all rates you want to enter and press Add rates button at the bottom.
When you are back in main rates window you can press:
- to delete selected rate
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Details - to view rate's details:
Here you can Split to Work Days and Free Days, delete rate or edit it:
Here you can add different End Time for different rates in time period and change other values which are more simpler then Advanced Rates.
NOTE: if there are no rates for some destinations – user will not be able to dial these destinations using provider to
which 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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User wholesale tariff
This tariff is used by users and it has no difference with provider wholesale tariff. All functions are the same – so if
you need to do something – just look into the provider section how to manage this tariff. This tariff type can be enabled or disabled in Settings. When wholesale tariff type is assigned to user – he will not be able to see all rates online. He will be able to download then in PDF or CSV format.
NOTE: you can't have Custom rates for user if he is using Wholesale tariff.
Rate import from CSV
It's very tedious work to add several thousands rates by hand so you can import them from CSV file. There‟re 7 steps for importing rates from 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 them one by one.
IMPORTANT NOTE: When importing - prefixes should be in E.164 format!!! E.g.
no international prefixes!
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1. File upload
Select Import from CSV for the Tariff you want to import rates.
Then press on Browse and select CSV file with rates. Press Upload file:
2. Column assignment
File is uploaded and 5 first line of this file is available to check:
This helps us in next step, to assign CSV file‟s columns to MOR‟s variables. We have following window:
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Here we must to assign columns from 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 want, you can enter manually values for them. Manually entered values have highest priority. If you omit them, then MOR insert values from CSV file, if there‟re no assigned column for these fields, then default values will be added to database (DB).
There‟re group of values under name USA specifics. These are only for USA/Canada. Users from these countries need to use these fields with their CSV files.
In the following example we selected fields which match our CSV file. Also entered some manual values for
Increment and Min.Time. We will count time by 30s increments and minimum time for the call will be 60s with
these settings:
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Such way of selecting CSV rows and assigning them to MOR values lets us import various CSV file formats.
3. Column confirmation
Next step is to review our selection and confirm if everything is correct. We can go Back or forward to the next step:
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4. Analysis
It will take some time to accomplish this step. Now MOR is checking a lot of things from DB and CSV files: comparing rates, searching for duplicates, fixing small errors and so on. We get something like this after analysis is completed:
The most interesting line is 3 bad destination(s). We press 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. Possible that there are no directions for them, MOR can‟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 whole file. Here you can stop and fix your CSV file or you can ignore these errors and continue.
We will continue due to demo purposes pressing on Proceed.
Following steps are separated to divide big processing power if rate file is huge. It allows browser not to crash and successfully import 60.000+ destinations from one CSV file. (Tested).
5. 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:
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6. Updating rates
Here MOR updates rates with the values from CSV files. These rates already exist in DB. Just press Proceed:
7. Creating new rates
Finally MOR creates all rates and we are all done!
Notes If there‟re a lot of rates please be patient and let MOR finish its job. Sometimes it gets some time to count everything and come back to life. At the moments it could seem that browser is not responding – that‟s not true – it just works very hard importing your very own rates from CSV file. If you get timeout error – try to import file in smaller pieces.
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Destination to create from CSV
When importing rates from CSV file now you can check which Destinations will be created automatically. You can view them in separate window when you press List in step 4 in field Destinations to create. Such destinations do not exist in MOR database and MOR will create them when importing rate file.
IMPORTANT! DO NOT FORGET TO ADD THEM TO APPROPRIATE DESTINATION GROUP LATER!!!
When you import some csv file and new Destinations are created they do not belong to any Destination Group. In order Call Tracing, Invoice generation and Dialing to work properly - you need to add these Destinations to appropriate Destination Groups. Use Integrity Check to do that. Otherwise:
1. Call can be dropped
2. Detailed invoice will have errors
3. Call Tracing will report bad call (when it is good)
Importing Tariffs from CSV with wrong Regional Settings
In many computers Regional Settings affects how CSV is saved from XLS file.
MOR accepts CSV file where values are separated by comma "," and decimal places by point "."
Very often values are separated by ";" and decimal places by comma ","
In order to import such files, we need to fix them a little.
FIle in Excel looks like this:
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Let's save it to CSV format:
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CSV looks like this and we see incorrect file format for MOR import (';' and ',' instead of ',' and '.'):
Let's open this file in some editor (MS Word in this example - any other will be good also):
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We see same picture in MS Word also:
In Find and Replace window enter Find what: ',' and Replace with '.', e.g. we are fixing decimal values from ',' to '.':
We replaced a lot of values:
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We see some improvement:
Again in Find and Replace window enter Find what: ';' and Replace with ',', e.g. we are fixing value separation from ';' to ',':
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Even more fixes done:
We see correct file format:
Let's save it to another correct file:
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Now you are ready to import this CSV file into MOR tariffs.
Possible errors
International prefix
If all your destinations are BAD - make sure you import them in E.164 format - e.g. WITHOUT international prefix.
Example of bad file:
Same file but fixed:
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Country instead of Destination
If you have fields like
Afghanistan Mobile
Then in Step 2 this column should be assigned to Destination/OCN Name instead of Country/Direction.
If you will make mistake in this step you will get a lot of BAD DESTINATIONS in Step 4.
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Tariff transformation
Wholesale tariffs can be transformed to user retail tariffs. It can be done in main tariffs window pressing on Make
user retail tariff (this link appears when tariff has at least one rate):
Enter values you want to add to values from wholesale tariff to retail tariff. 3 cases possible: Let's say X is wholesale rate
1. You enter value Y into Add amount field then:
Retail rate will be X + Y
2. You enter value Z into Add percent field then:
Retail rate will be X + (X/100 * Z)
3. You enter value Y into Add amount field AND Z into Add percent field then:
Retail rate will be (X + Y) + ((X + Y) / 100 * Z)
NOTE: Largest value from wholesale tariff is taken for Destination group in retail tariff (by prefix) when doing this
transformation. HINT: this function can be used when importing user retail tariff. At first you create fake provider, import rate for him and transform to user retail tariff with values 0/0.
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Export to CSV/PDF
In main tariff window you can export tariff to PDF or CSV.
Exports tariff's rates to PDF
Exports tariff's rates to CSV
Sell minutes to only one country
If you want to sell traffic to only 1 country (or several) create tariff for user which has rates for these destinations only.
Tariff analysis
Using this tool you can select the tariffs (wholesale) which you want to compare. Click on desired tariffs and select currency in which you want to compare tariffs. When you selected tariffs press on Generate CSV icon to get CSV file.
In this file you can see all selected tariffs with their rates for ALL destinations. In 4 last columns you will find MIN and MAX rates for each destination with their providers. This let's you compare tariffs on the fly. The final section shows
Total/Min/Max rates for each tariff. Using these values let's you see which tariff is the most expensive/cheap. NOTE: Min + Max can be not equal to Total because same provider can have MIN and MAX rates for some prefix if there're no more providers for this prefix (destination).
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DIDs What is DID? In short – it‟s external number to make incoming calls to your VoIP network to reach some device. (It‟s a very simplified explanation which suits our needs right now).
For more extensive explanation please check here .
VERY IMPORTANT RULE: DIDs MUST be in E.164 format. You can reach DIDs window in SETTINGS – Billing – DIDs:
Here you can see all DIDs and their info.
Status is very important. For every just created DID status is Free. DID can have several Statuses:
Free – This status means, that DID is free to be reserved for any user/device. You can also Terminate it.
Reserved – DID is reserved but NOT actually used for some user. It‟s useful in negotiation state before signing a contract when user decides to sign up – then we will Activate this DID. We can also make this DID Free. Active – DID is assigned to some Device and it’s functional. We can Close it.
Closed – DID is no more functional. E.g. all calls going to this DID will get Hangup. One important feature of this state: when DID is Closed, he will be closed for some time (default – 90 days) in case user will change his mind and will decide to use this DID again. (Useful when user is not paying for services, so we can suspend DID using
just Closing it). MOR will make it Free after time of Closing will run out. You can manually make it Free or Terminate this DID.
Terminated – DID is no more used in MOR. It‟s here just for historical purposes. You can make it Free if you want.
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User/Dial Plan and Device shows who will answer when dialing this DID.
Pressing on user or device brings you to it's settings. If you press on icon you will go to device's Call Flow
window.
If you want to edit DIDs settings press on icon:
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This window is for Free DID. You can:
1. Change Provider and language for this DID. Language is for voice prompts in Asterisk dialplan.
2. Reserve DID for user
3. Assign to Dial Plan
4. Terminate DID
When DID is assigned to some device you have little different choices:
Here you can:
1. Change provider/language
2. Assign DID to different device (for same user)
3. Stop DID subscription and reserve this DID for 90 days (by default) to this user incase he will change hsi
mind and will want to get back this DID
When DID is assigned to Dial Plan choices are almost identical.
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In the main DIDs window if you want to add new DID(s) press on New DID:
In this window you can add one DID or series of DIDs. Also you should select provider which sold you this DID. QUESTION: How to delete DID?
ANSWER: At first mark it as not used (free). Then terminate it. When you do it – you will see next to DID in
main DID window. Pressing on it you can delete DID. QUESTION: How to edit DID's number? ANSWER: It is not possible to edit DID number. You can only delete old one and create new one.
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How DIDs are handled?
Technical explanation:
1. Let's say call comes through some Provider. The connection type is not important. So it could be
PSTN/SIP/IAX2 whatever.
2. Call -> Provider -> PSTN/SIP/IAX2 -> Your Asterisk
3. Then call hits dialplan [mor] exten => _X.,1,mor:
4. Call -> Provider -> PSTN/SIP/IAX2 -> Your Asterisk -> MOR
5. MOR checks incoming DID with it's DB, if it's found, it sees where to route the call. In MOR GUI you can select
to route DID to some special Dial Plan (CallBack, Calling Cards, Authorization by PIN/ANI) or some local
device.
6. Local device in it's Call Flow can forward the call to:
a) VoiceMail,
b) another Local Device
c) External number
7. So summed up, MOR can forward DID to:
a) Dial Plan (CallBack, Calling Cards, Authorization by PIN)
b) Local Device
c) VoiceMail
d) External Number
8. Call -> Provider -> PSTN/SIP/IAX2 -> Your Asterisk -> MOR -> (a or b or c or d)
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Usable DIDs from providers
When you are buying DID from some provider make sure it has several necessary properties:
1. Incoming call from provider carries DID information
2. Incoming call carries CallerID information
The most important is first one. If your DID (provider) does not suit second property – that means you can use DID for incoming call but you can't use this DID for Callback. Let's go in details here:
DID or destination information in the call is number to which call should be sent when received by your server. Very often providers send call without such information. When call is received by your server – it does not carry destination information. E.g. – server does not know where to route the call, to which device. If provider uses Asterisk, then we can give an example how he sends the call:
exten => DID,1,Dial(SIP/your_server) correct way to send call is: exten => DID,1,Dial(SIP/your_server/DID)
If call is sent without destination – in Asterisk CLI you will see call coming to 's' extension. That indicates that your provider sends call without destination information. You can "educate" provider to fix his configuration to make proper way of sending calls. Maybe you will get lucky and he will understand his error. Often they sell these DIDs only for devices to connect. That means – no PBX or Softswitch will be able to use these
DIDs in proper way without a lot of .conf file editing. TECHNICAL NOTE: it is possible to make such DIDs work when for each connection from provider different context is created. But it's not practical way to do it. Because such configuration will fail when you want to buy second DID from same provider.
Second case is when provider does not send CallerID information. That way you can't decide who calls you and off course you can't use Callback to call caller back. If you need Callback solution – make sure your DID provider sends correct CallerID information. Without it you will not be able to use Callback.
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DID billing
DIDs can be billed. DID billing is quite complex for DIDs. They have 3 types of rates. That means there are 3 prices
counted for 1 call. DID's rates can be reached in main DIDs window pressing on Rates icon for selected DID:
Here you see 3 different rates:
1. Provider's rate - this rate is paid to or by provider for using this DID
2. Incoming rate - this rate is paid by user who calls this DID
3. Owner rate - this rate is paid by owner of this DID if somebody is calling to this DID
Enter values for these rates and bill your DIDs!
NOTE: If local user e.g. user from within the system is dialing DID – all rates are always 0.
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Example - Configuring DID to ring some Device
Primary situation
We have:
DID - external number provided by some Provider: 02180995165
Number is from UK
Connection to provider is via IAX2, IP: 23.45.67.89:4569
Devices we have: SIP/201 and SIP/201
We want:
Dial DID (02180995165) using our mobile phone and ring Device SIP/201
Dial DID (02180995165) using local Device SIP/202 and ring Device SIP/201 (local SIP-SIP call not going to PSTN)
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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
Configure Localization for Provider
As DID is not in E164 format, we must configure MOR to handle that.
That means that DID from provider comes in non-E164 format: 02180995165
We must create Location for this provider and add rule to it.
We create Location: UK and add rule to convert non-E164 DID to E164 DID: 02180995165 -> 4402180995165
Create Provider
Create Provider in MOR GUI (no need to edit any of .conf files!!!)
Do not forget to set Location: UK for this Provider.
Register Provider
IF your Provider asks you to register, register it in /etc/asterisk/iax.conf (or sip.conf if you connect to your Provider via SIP).
Create Devices SIP/201 and SIP/202
Create Devices as usual (no need to edit any of .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
Assign DID to Device SIP/201
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That‟s it.
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Number manipulation Number manipulation is necessary when receiving and sending number. The main idea is very simple but very important – it's a rule we will follow all the time and we can split it into 2 parts:
1. Incoming number should be made E.164 compatible (done by Localization) 2. Outgoing number should be made out of E.164 number (done by Provider rules)
What is E.164 format number? In short it's the number without international prefix which starts with country code, in long - check Wikipedia. The first part which makes any incoming number E.164 compatible is done by Localization. The second one which changes E.164 number is Provider Rules. The basic schema for all this process can be simplified as:
The second most important rule is:
All numbers inside the MOR are E.164 compatible! What does it mean? If you need other format number – you should use Localization or Provider Rules to achieve this. Otherwise it's E.164 everywhere. This is very important!
International prefixes, local number formats are handled by Localization. Provider technical prefixes and special number formats for providers are handled by Provider Rules. That means:
if you receive number from user with international prefix – you need to strip it and make number E.164 compatible.
if you receive number from user in some local number with some special digit(s) in front of it – you need to strip it, add country code make it E.164 compatible using Localization
if your provider requests some fancy technical prefix – you need to add this prefix using Provider Rules. Add prefix to E.164 number. It can occur that you need to cut some digits (let's say country code) before adding technical prefix – do it with Provider Rules.
And similar more examples…
Let's repeat again, because it is extremely important:
When you receive number from user – make it E.164 compatible with Localization
When you are sending number to provider – use Provider Rules to change E.164 number
to provider expected format
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Localization
Localization transforms received number to E.164 format number.
Main idea of Localization: No matter how user dials 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
caller. Often it is important to let users to dial different format numbers to reach same destination. It depends on the
location where user (or his device) is located. For that purpose each device has field Location. It describes the physical location of device and rules how this device can dial destination. In other words – each device IS (belongs) in some Location where some specific dialing rules apply.
Location describes where caller IS - not where he is calling! By default device has location 'Global'. Global rules apply to ALL devices. If user belongs to some other location and his dialed destination has some rule for his destination then global rules are not activated – they have lower priority upon 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 Global location. You can't delete this location. It's default for all devices.
You can add new Location by entering this Location's name in field at the bottom of table and pressing :
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In the column Rules press on to access Location's rules:
Following settings should be explained for rule: Name – just for informational purposes Cut – what prefix from the number should be cat
Add – what prefix to the number should be added Min Len – what's the minimum length of number we are trying to apply this rule Max Len – what's the maximum length of number we are trying to apply this rule
When you enter all values (Cut or Add can be left empty – but just one of them at a time) press .
Example:
Normal Lithuania MOB prefix is: 3706 User's device is located in Lithuania Vilnius
By current Lithuanian regulations local customer can dial 9 digit numbers starting with 86 to reach Mobile subscribers
To describe this situation we will create following rule:
Rules can be enabled or disabled pressing icon or accordingly in the "Enabled?" column.
- This icon let's you edit rule's details.
- This icon let's you delete rule
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Now in the main Localization window:
Press on in Devices column to access Devices in Global Location:
Here all Devices with selected location are listed. Choose other location for selected Device and press on icon to activate this change. Device will be assigned to the selected Location.
NOTE: It's possible to change Device's Location in Device's settings (SETTINGS – Users – Devices).
Now our device belongs to Location Lithuania Vilnius. And if we dial mobile number in Lithuanian format for example:
863042438, MOR cuts 86 (remaining part 3042438) and adds 3706. Result is: 37063042438 which is E.164 compatible and MOR can bill it correctly.
NOTE: make sure that after localization your number always is E.164 compatible. Call Tracing can help you do this.
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DID handling by Localization
When call comes in to your system from the Provider – they should be localized to E.164 format for DID search. In
other words – as DIDs are in E.164 format, whatever comes from provider – should be localized to E.164 to find correct DID. To do this each Provider has Location also. That means that Location rules can be created and applied for Provider. You can do this in Provider settings window:
Example: you have DID 37052058393 and provider sends you 0037052058393. You need to strip this international prefix with following rule:
And do not forget to apply correct Location to the Provider!
Special case: In some countries when E1 is connected to the server, only 4 last digits of the number reach MOR. Localization should be used here to add remaining part of the number.
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Provider Rules
Provider Rules transforms E.164 compatible number to any number format Provider requires. Different providers expect different number formats from you. You can send them whatever they want using Provider Rules.
They can be reached pressing icon next to some provider in Providers window:
We see similar window as with Localization. The rules to manage Provider Rules are identical to Localization management so we will not cover these here again.
Let's say our provider expects to get number with technical prefix 89765#. We can do this in following way:
That's it – very simple. Now every number which will be sent to this provider will be transformed adding our created tech. prefix.
Example: 37063042438 should be sent to this provider, so this rule will be activated and number 89765#37063042438 will be sent to provider.
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Workflow to setup Localization and Provider Rules
To setup correct Localization and Provider Rules you need to do following things:
1. To know what number format user will be dialing 2. Create appropriate Location and assign user to that Location 3. Create rules for specific Location to handle user entered numbers 4. To know what number provider expects 5. Create rule for provider which handles number transformation to providers expected number
For the training let's cover full example using the scheme we had before:
1. User dials some number: 863042438 2. Number is made E.164 compatible by Localization: we cut 86, add 3706 and we have: 37063042438 3. Billing: MOR uses 37063042438 to find correct destination and rates 4. E.164 number is transformed to provider format by Provider Rules: number 37063042438 is transformed to
number format which provider expects by adding 89765#, so we have: 89765#37063042438
5. Call is sent to provider: 89765#37063042438 Hint: We should remind you to use Call Tracing to debug all this. Hint: World Telephone Numbering Guide can help you a lot handling different number formats.
Example
If calling to USA/Canada, Provider accepts number in format: 1+number
If calling INTERNATIONAL number (everything else except USA/Canada), Provider accepts number in format: 011 + E164 format number (country code + number)
In Provider Rules should be such rules:
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Explanation:
It is IMPORTANT to remember that number from MOR to Provider Rules (pior going to Provider) comes in E164 format
Rules are grouped by LENGTH of CUT field - it is important!
Let's say user dials 123456789012. Number length is 12 (12 digits). So both rules apply. But USA/Canada Specific rule has higher priority, because CUT field LENGTH is 1 (1 digit) and in Int.prefix it is empty (0 digits). So USA/Canada Specific rule will be applied.
When any other number comes, starting not from 1, e.g. 23456789012, only Int.Prefix rule is applied. Because number does not starts with '1' as described in USA/Canada Specific rule.
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CDR
Import
Sometimes MOR can be used as separate billing software – not doing realtime billing. For that purpose it is possible to import CDRs from other switches (can be not Asterisk) and bill these CDR's.
CDR Import can be found in SETTINGS – Billing – Functions – CDR Import.
First step is to select some CSV file to import:
Press on Upload file and you will get next window where you will have to assign columns to CDR values:
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On step 2 you can see necessary fields to select. They are marked by *(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 import process.
Select appropriate settings and press Assign Columns:
Here you will be presented with parsed file. If something is not right – press Back button. If it's ok – Confirm columns:
Here we select the User to which we want to assign all these calls.
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Here we select Device to which all these imported calls will be assigned and Provider.
Finally file is imported.
Export
You can export CDR in PDF or CSV formats. It can be done in Call Stats page (STATISTICS – Calls - Calls) and other places where you see:
Export to PDF
Export to CSV
Just press on one of these links and you will get CDR in selected period.
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Rerating
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:
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:
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Here you will see the Difference after rerating.
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Users User is the actual user of the system – the client which dials out or receives calls. User can represent one person or whole company. User is main unit in the system. Most of the settings are associated with user.
The user management window is in SETTINGS – Users:
In this window you can check all users with their main details:
and shows is user logged into GUI or is it offline. Do not confuse this with device login to the server. These are different meanings.
- let's you login as this user without entering his password
- initiates Call Tracing for this user
– shows device groups – check Device Groups chapter
- Devices – shows user's devices
- Custom Rates – shows user's Custom Rates
- shows user's Subscriptions
- Recordings – please check Recordings chapter
- This icon let's you edit user's settings
- This icon let's you delete user
NOTE: System Admin is main user and cannot be deleted. NOTE: User which has calls cannot be deleted – that's against billing phylosophy – all calls made within system should be logged.
QUESTION: How many users can I create with this MOR installed on my server?
ANSWER: User number is unlimited
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User details
Press on to edit user's settings:
In this window you will se a lot of user's settings. There're several groups: General, Details and Registration Address. General
Username – username for user to login to MOR GUI Password – password for user to login to MOR GUI User type – by default it's "user" LCR – LCR which user's devices will use to dial out
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Tariff – prices for user Balance – how much YOU owe to user (-10.00 means user owes you 10.00) Credit – how low user's balance can go Postpaid/Prepaid – Type of user Not Blocked/Blocked – should we block this user from dialing out? Primary device – just primary device
NOTE: for detailed explanation about Balance/Credit check appropriate topic. Details and Registration Address are self-explanatory.
User blocking
For various reasons it is possible to block user. He will be unable to dial out but he will be able to get incoming calls. In order to disallow user to get incoming calls it is necessary remove it from the system. When user is blocked and he tries to call, his call goes to dialing application and here it gets dropped. Users balance will not be touched.
Blocked users can dial local users. Also they can receive calls.
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 to them to have clear control panel.
For that reason we just Hide them. They remain active - nothing changes for them - except we do not see them in the GUI.
Later you can find them in SETTINGS - Users - Hidden. You can unhide them if you wish anytime using icon .
To hide a user go to his details and select Hidden. To unhide - Not hidden.
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Balance and credit
There are two fields which describes money situation for user: balance and credit.
Balance is the amount of money you owe user. Credit – how much money you allow user to be in debt before you. These fields act and have slightly different meaning for different types of users: postpaid and prepaid: Postpaid users
Balance – can be any value:
< 0 – means user owns you money for your services, he will get invoice at the end of the month and when he pays for that invoice and his balance will increase
> 0 – user has overpaid you (you owe to user) == 0 – nobody owes nothing to anybody (this is usual when user is new or just paid his invoice and did
not made a call or used other services)
Credit for postpaid user can be: > 0 – that means 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) for total amount of 10 USD. At that time his balance will start at 0 and go down to -10 USD. When balance reaches -10 USD user will not be able to make any more calls. == ∞ - user can make unlimited calls and use any services without any restrictions. All amounts will be
included into invoice.
Prepaid users Balance for prepaid users ALWAYS should be more then zero. Otherwise 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. E.g. if user has several devices and his balance/credit does not allow to make any more calls – that means he will be unable to use any of his devices.
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Grouping
Groups in MOR are groups of users which have one or more supervisors or managers over group. Managers possess
administrative powers over users – like mark users as logged or logged off (thus allowing or preventing calls by the user).
Group Creation
Only administrators can create Groups. It‟s done under SETTINGS – Users - Groups. Just enter Group‟s name to
create Group. You can select type for group from 'simple' or 'callshop'. It describes how you will manage group later.
Group's details
There are some options you can modify with Groups:
Rename a Group
Delete a Group
Modify Group Members
Adding Users to Group
Only administrators can add Users to Groups. Press on Members for a certain group and you will get following
window:
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From drop-down list at the bottom select desired user and press „Add member‟ button. User will be added to the group.
Managers and Users
Each Group can (and must) have one or more manager. The manager is the account that can control users. Like login a user or logoff a user (allow or disallow users‟ calls). You can change the user type pressing on „Change type‟ button for a user.
Manager's actions
If a user is manager of some group, they will see following section in their interface: Under groups is a list of all the groups that manager can manage, or administrate.
Group's details for managers
By pressing on Groups user (manager) will be presented with details about the group:
Pressing on „Members‟ button near some group manager will get list of users:
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Pressing „Login‟ or „Logout‟ buttons manager will change login state of user accordingly. This change will be put into database for further statistic and reference.
Member stats
Pressing on Member stats you will open window where you can setup some values for users which mostly are used in Call Centers:
All these values are used in Realtime statistics window.
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Login stats checking
Login statistics can be viewed by the manager under STATISTICS - Users - Logins (for each user). Or by pressing
icon in previous picture.
NOTE: Callshop Functionality is based on User Grouping. It has slightly different function set but the principles remains the same.
NOTE: If you can't see group just created in your left side menu – make sure you are manager of that group.
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Devices
IMPORTANT NOTE: MOR uses Asterisk Realtime – that means all configuration is saved into
database. Config files are not modified. After creating provider or device – do not search for
them in .conf files – they are in database. Commands such as sip show peers do not work with
Asterisk Realtime (when caching is turned off – what is recommended).
In main user window press on to check user's devices. Also you can select SETTINGS – Users – Devices to
check all devices.
Each user can have many devices. Device represents end-point where voice traffic should be sent. As example device can represent:
VoIP phone
VoIP softphone
PSTN phone (connected through FXS port)
PBX
Softswitch
NOTE: It is important to remember that some settings apply to users and others to separate devices. That means that if setting applies to user, same setting applies to all his devices. For example balance. It is field for user. But if
balance is too low – user will be unable to dial out any of his devices.
In this window you see some details for user's devices. Choices for you are:
Call Flow – see appropriate chapter
DIDs – shows DIDs assigned to this device
CLIs – CallerIDs for this device
- Edit device's settings
- Delete device
Icons near device shows it's status or type:
- Trunk
- Trunk with ANI
- Virtual Fax device
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Settings
When you press icon in order to edit Device you will get almost similar settings as for Provider:
All explained here:
General
Accountcode – id and accountcode of the device
Description – name for device, for informational purposes PIN – device's PIN for authentication/authorization Type – SIP/IAX2/ZAP/H323/FAX – choose which one is used by your device, e.g. in which way do you connect
to it Extension – short number by which this device can be reached DTMF Mode – available options: inband, info, RFC2833, auto – choose which one your provider uses
Works not logged – does user need to log to MOR to be able to dial out? Location – default Global, choose appropriate one Trunk – No/Yes/Yes with ANI – see appropriate chapter Allow loss calls - this setting let's you decide - do you want to allow user to use some provider if provider's
rate to some destination is lower than user's rate. Allowing such call you will pay more to provider then user pays you for such call - that means you will take loss on such call. If this setting is no (default) - cheaper providers will not be used in dialing.
Timeout - It describes how long device should be trying to be reached before timeouting the call. Default
value is 60 seconds. Minimum value is 10 seconds.
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Authentication For ZAP devices:
Channel – which channel (or channel group) to use on PRI/BRI/PSTN card. Channels and groups should be configured in zapata.conf.
For all other devices:
Username – username you enter in your device Password – password you enter in your device Or Authenticate by IP – just check the checkbox
o Make sure you put correct Hostname/IP adress!
CallerID
CallerID – consists of two parts: name and number. The "Number" part gets transferred by default by all technologies: GSM, PSTN, SIP etc, the "Name" part only on some. The number you see on your mobile phone when somebody is calling you is "Number" part.
Name – the "Name" part of CallerID
Number – the "Number" part of CallerID
NOTE: if you leave these fields empty – user can set CallerID by himself. Usually is not a good way to let it do it for user. CID control by DIDs – only available when this device has DID(s) This setting let's you control what CID user can enter in his devices. These CID numbers should be from the set of
device's DIDs. It is mainly used when user's PBX is connected over trunk and many DIDs are routed to this Trunk. Calls coming out from this PBX must have CallerID Number from the set of DIDs assigned to this Trunk (PBX). If CID from DIDs is checked - system checks if 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 user enter any CID number he wants. If CID from DIDs is checked and user enters any CID (not from his DIDs) - system will change his CID by his first available DID (selected from database by lowest id).
Network related
Host – device's IP, the value can be "dynamic", which means that device should register (it should be like this
most of the time). Port – port used to connect to device (default: 5060 for SIP, 4569 for IAX2). If host = dynamic this field is
updated by Asterisk when device registers. It is port from where call is coming (from device). Media control – canreinvite/transfer – do you want your server to stay in media path between your clients and
your provider NAT – available options: yes, no, never, route. For detailed explanation about these settings please refer
here. Qualify – how often "ask" device about availability. More details here for SIP devices and here for IAX2
devices.
Groups
Call Group – to which Call Group this device belongs Pickup Group – which Cal Groups this device can pickup
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Voicemail
Email – to which send received Voicemail
Password – digit password user should enter when he calls Voicemail number to listen for
messages
Codecs
Choose the codecs your provider uses. NOTE: When no fields are checked – that means that all codecs are available. E.g. settings in sip.conf or iax.conf are effective.
Allowed addresses
IP, MASK – permit traffic from these IP only. You can find detailed explanation here.
Advanced
Fromuser – info here. Fromdomain – info here. Trustrpid – info here. Sendrpid – info here. Insecure – info here. Progressinband – info here.
Video support – does your provider support Video over IP? More info here.
Tell options
Tell balance – should MOR tell balance to user when he is trying to dial (every time) – default no
Tell time - should MOR tell remaining time to user when he is trying to dial (every time) – default no
Tell remaining time when left – when some time is left MOR will tell remaining time to talk (in seconds)
Repeat remaining time when left – repeats remaining time when left some time (in seconds)
Device details
This window shows all fields from database table devices to some particular device. It is only read-only.
It can help to troubleshoot login problems for device.
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Which codecs should I choose for devices?
It depends.
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.
G729
Little worse quality but is often used because it saves traffic (~8kbps + ~20kbps overhead ~= 28kbps to one direction).
It needs to be installed separately and it is not free.
Video
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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.
Notes
Make sure you have always selected audio codec! Otherwise you will not be able to place calls!
Process SIPCHANINFO
Can be enabled per SIP device basis. It is useful in debugging. And it is saved to calls table when it is enabled.
When it is on and Device Type is SIP, then in Asterisk CLI you can see something like:
============== SIPCHANINFO ===============
Peer IP: 213.164.109.128
Source IP: 213.164.109.128
From: sip:212@prov1
Contact: sip:[email protected]:5060
Useragent: X-PRO build 1082
Peername: 212
T38Passthrough: 0
==========================================
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How to create Zap/H323 devices
Zap and H323 does not have Realtime support in Asterisk. In order to configure them you should do following steps:
1. Create devices in GUI
2. Write down just created device accountcode (Acc id)
3. Create device in zapata.conf or h323.conf (ooh323.conf or similar) file
4. Add accountcode = X to just created device in .conf file where X is the Acc id of device created in GUI
5. Reload Asterisk and you are ready to go!
Extension lines
In device setting you can select Extension lines. It will show you Asterisk extension lines:
This info is for informational purposes.
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CLIs
CLI or CallerID is number from which user can dial to MOR. These values are used when user is dialing into MOR
through Trunk/Provider with ANI (Automatic Number Identification).
You can reach main window of CLIs in device settings selecting CLIs:
Here as usual you can add new CLI, edit existing one or delete it.
IMPORANT! CLIs should be in E.164 format!
What is the difference between CallerID and CLI
CallerID is for outgoing calls, CLI are for incoming calls when call comes over Trunk/Provider with ANI.
Trunks
Under Trunk we have in mind other server/PBX which can be connected to the MOR system. When we are calling to such device, we send full number which was dialed before.
Example: when dialing simple device, we would use: Dial(SIP/some_device), and if this device is Trunk we use: Dial(SIP/some_device/${EXTEN}). This lets Trunk to decide to which device (connected to the Trunk) to route the call. If you want to make device Trunk, just mark is at Trunk in device details:
No/Yes/Yes with ANI
Trunk with ANI
It is sometimes necessary not to bill whole trunk as one client but to be able to bill separate clients connected to the
server/pbx (trunk) separately. In order to do this, you should mark some device as Trunk with ANI. Now when some other device connected to that Trunk dials through MOR, MOR check this device's CallerID and if it's present for some device in the MOR – then MOR recognizes the device as valid device on the MOR and assigns call to him instead of Trunk.
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Fax device
This is special type of device. It can be enabled/disabled in Settings. When call comes to this device he is expected to be fax call. MOR then receives fax, converts to PDF and sends to appropriate email(s). This is covered in details in Additional Functionality – Fax2Email chapter.
Virtual device
This kind of device is mainly used for virtual users authenticated by ANI.
Device groups
It is possible to group devices. This represent's companies branches – e.g. each device belongs to only one group – companies branch.
To access groups for each user in users window press on Dev.Groups:
In this window you see all device groups for selected user. Under each group you see devices assigned to this group.
Press on Add new device Group to add new group:
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Each group has its own address – that's for invoices.
When new group is added you can delete it. Primary group cannot be deleted.
Edit Details – let's change group's name and address.
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Call Flow Call Flow is set of rules how call should behave when it is trying to reach some device.
It can be reached by pressing on Call Flow link in device details:
Call has several states which are visible from this image:
Before Call - call has not reached device
Call – call to device is in progress
Answered – call was answered
No Answer – call was not answered and ended after timeout
Busy – callee was busy
Failed – call to device failed
We can take actions for several call states: Before Call, No Answer, Busy and Failed. The possible solutions are:
Empty
Forward
Voicemail
Fax detect (only for Before Call state)
Empty – means no action should be taken. All others we will discuss in more details.
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Forward
Call can be forwarded to Local device or external number:
System can select any device he likes. If user is editing his devices Call Flow – he can select only his devices.
Using forward function it is easy to implement:
Unconditional forward – when forward is used in Before Call state
Forward on no answer/busy/failed – when forward is used in No Answer, Busy or Failed states
Voicemail
This sends call directly to voicemail. It's simple. Put in Before Call – and call never reach device – it will go directly to voicemail.
You can put it in No Answer/Busy/Failed state to achieve different functionality. Global Voicemail settings are configured in Settings menu.
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Fax detect
This action can be set ONLY in Before Call state and only if Fax device is enabled in settings:
You can select only FAX device to route call if Fax is detected. MOR will use fax detection when this action is set. And
if fax is detected – it routes call to set fax device – otherwise – to basic device. More details can be found in Additional Functionality – Email2Fax chapter.
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Additional functionality This chapter will cover additional functionality of MOR PRO.
Online registration
This function lets user to register online. It can be accessed in MOR's entry page 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 page Registration part.
In following window user should enter his details. Fields marked by * are mandatory.
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After pressing on Sign up user will get registered, his account is created and he will get connection info on his screen:
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If email sending is enabled (in Settings) and Email SMTP server's details are valid – user will get Email with all the settings. Same to administrator – he will get informational email about this user that he registered to the system. These two settings should also be enabled. If you set every setting to send email to admin when new user registers but no email arrives - check if you have
email set for admin user.
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Recordings
It‟s possible to record selected user‟s calls for monitoring purposes. Recording works only for outgoing calls.
In order to use Recording you have to setup which user‟s calls you want to record. Go to SETTINGS – Users – Recordings and you will see such window:
Here you can check Record? field for the users you want to record and press Change at the bottom for the changes to take effect.
In the Asterisk CLI you should see something like this at the start of the call for the user which calls we do not record:
And such image if we selected to record this user‟s calls:
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If you get following lines:
That means you do not have installed php4-cli or php5-cli package or there are no /usr/bin/php file (or symlink). The recorded calls are stored in mp3 format to save space on hard disk.
You can view/listen/download these recorded calls by selecting Show in the Recordings window. You will see
something like this:
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Call Shop functionality
This chapter describes how to use MOR in Call Shop environment.
The main idea is to have several Phone Booths (we will call them „Users‟) which are allowed to dial out.
After that they get billed for the calls they made.
Setup
First of all we need to create new Group for Call Shop.
Select SETTINGS – Groups – New group
Enter name for Call Shop (“CallShop” in example) and select Type as “callshop”. Press Create.
Now we have group which will be use for Call Shop. We need to add some members to this group.
Member is User in MOR terms. Let‟s add them.
Press on Members for the CallShop group:
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Each group can have different types of members. Member can be “Manager” and “User”. Manager will
manage the group. Group can have several managers.
Let‟s add manager for our Call Shop. Usually that will be “System Admin”.
Make sure in selection list Add new member “admin – System Admin” is selected and press Add
member button:
When new member is added, it‟s type is „user‟, let‟s change it to „manager‟
Press on Change type for our only user:
Now we have manager for our group:
Also instantly we have new controls for the „CallShop’ group under MANAGEMENT,
because we are logged in as „System Admin‟
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Now we need some users for full Call Shop functionality. These users will do actual calling.
We add them same way as our manager but we leave their type as „user‟:
And second:
Now we have 3 members: 1 manager and 2 users:
The setup is complete. If users have correct setup (devices, tariffs, LCRs, etc) and they can dialout, then
we can start using our Call Shop.
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Operation
Now we go to MANAGEMENT – Groups – CallShop and we see our main Call Shop management
window:
1. Here you can select the period you want to check. It‟s saved in ypur browser‟s memeory and if ypu
go to other pages, you will have this period selected automatically.
2. Calls/Duration are for the selected period.
3. Login shows last login time of the user. If it‟s empty – that means user is not logged in and he
can‟t call. If you press on login time, you will get detailed list for that login time till current
moment.
4. Login duration – how long user is logged in.
5. Here‟re controls which let‟s you to login/logout user, at the same time to allow him to call or
disallow him to call.
6. By pressing on this link you will get detailed list of calls in selected period.
7. Here you will get list of logins in selected period grouped by days.
The main procedure is as following:
1. New person comes to your callshop and wants to call.
2. You select free user in Call Shop management window and press on button “Login” (5) allowing
calls from this Call Booth.
3. Person goes to this Call Booth which represents some user on management interface and starts
calling.
4. When he finishes he comes back to you (manager).
5. You press on Login time (3) for this user and you get a list with all user‟s calls.
6. You can get PDF report for user‟s calls, print it and give it to user.
7. User pays for the calls.
8. You press “Logout” (5) for his Phone Booth ending this session.
When you press on Login time (3)
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you will get call list from this login time till current moment:
Here you can press on Export to PDF to get same result in PDF format:
When you press on “Logins” (7) you will get Logins window:
Here you will find graphical representation and all logins – sessions with ability to get call list for all of
them.
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Phonebook PhoneBook is for information purposes only. It searches for the name in database based on incoming number (Caller‟s number/A number/Source).
The PhoneBook structure in MOR is as following:
• There‟re two PhoneBooks: Global and Simple • There‟s only 1 Global PhoneBook
• Each user has his own Simple PhoneBook • User‟s Simple PhoneBook has higher priority than global PhoneBook – e.g. if record is found in Global and in
Simple PhoneBook – the record from the Simple PhoneBook will be used
Global PhoneBook
Only System Administrator can edit Global PhoneBook. It can be found under SETTINGS – PhoneBook:
In PhoneBook management window you will see all necessary info to operate it. Number, Name and Added fields are self explanatory.
By pressing (3) you can delete appropriate field.
If you want to enter new record fill field (1) and (2) and press (4).
If you want to edit existing records, move mouse over the field you want to edit and press left mouse button once on it when it changes color to yellow:
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And you will be able to edit this field. Press OK to accept changes or cancel to discard them:
That‟s it for Global PhoneBook.
Simple PhoneBook
Each user has its own Simple PhoneBook which has higher priority than Global PhoneBook.
It could be found under PERSONAL MENU – PhoneBook:
Here you see almost same as in Global PhoneBook except there‟re records from Global PhoneBook. They do not have
delete option instead they are marked as ‘Global’. All other functionality is the same. Same rules apply to add/edit/delete records.
Using phonebook
When user with the number which is in PhoneBook calls-in, MOR checks his number and sets CALLERID(NAME) to the value from DB. It‟s visible on (soft)phones which support CALLERID(NAME):
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Fax2Email
This feature let's user receive normal fax and get it to email in PDF format. The following image shows common implementation of Fax2Email functionality:
In order to start using this feature it should be enabled in Settings – Various – Fax2Email enabled.
When it's enabled you need to create some device and make it FAX type. It can be done in device settings:
When device's type is FAX it is marked as FAX device everywhere with icon .
Then you need to add emails to which received fax should be sent. You can enter unlimited emails here.
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When fax device is created you have two possibilities to use it.
1. It is possible to route call directly to fax device
2. You can use fax-detection, so when fax is detected – it will be routed to fax device and if it is not detected –
to nornal device.
Second choice lets you save one DID – e.g. lets receive calls and faxes on same number. First method is very easy to implement – you just need to assign FAX device to some DID:
When you want to do fax-detection you need to set this in some device's Call Flow. In Before Call state:
Here when call comes system will try to decide is it fax – if so – call is routed to FAX/108 device and sent to appropriate emails. If call is simple call – it will gets sent to IAX2/104 device.
For received faxes and various stats check Fax statistics in STATISTICS section.
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Mass mailing This function is used to send bunch of emails to your clients. It can be used to send some commercials, notes about service unavailability or anything else – maybe greetings for Christmas. This function can be found in SETTINGS – Setup – Emails:
Here you see 2 first emails which are templates for email notification on registration. 3rd is created for demo purposes
for mass-mailing.
If you want to add new email you should press on Add new email and on to delete email.
We will press on to edit created email. Creating and editing are very similar. We will show how to edit:
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Name and Subject fields are simple. The most important field is Message text. Here you enter message for the user. Also you can use such variables here: <%= server_ip %> - Server IP where your users should register their SIP/IAX2 devices <%= device_type %> - Used on user registration - shows device's type user had selected <%= device_username %> - Used on user registration - shows 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 After all press Change to save it.
In main window you can press on icon to select to which recipients to send this email:
Check the desired users and press on Send. In main window if you press on digit in the column Sent to users you will get list of users this email was sent for:
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Dial plans Dial plan is name for specific functionality in MOR. Currently there're 4 types of it:
Authentication/Authorization by PIN
Authentication/Authorization by ANI
Calling Card dial plan
Callback
We will cover each of these.
You can find all dial plans in SETTINGS – Billing – DIDs – Dial Plans.
Auth. by PIN
This function is for customers who can dial some access number and enter their device PIN number get a dial tone to
enter some destination number. It is used in some countries to get cheaper calls abroad. The algorithm for Auth. by PIN dial plan is visible in following image:
The main things to do in order to configure such functionality are:
1. Create appropriate Dial Plan
2. Assign some DID to this Dial Plan.
At first in main Dial Plan window we press on New Dial Plan and enter values to create Auth. by PIN dial plan:
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IMPORTANT! Check the Use ANI? field – otherwise it will different type of Dial Plan.
Ask Number/PIN (times) – how many times user can try to enter PIN
Ask destination (times) – how many times user can try to enter destination
Press on to create this dial plan. First part is done – we have created our dial plan:
Now we need to assign some DID to this Dial Plan. For this in DID's edit window we assign selected DID to this Dial Plan:
And in main DID's window we see that this DID is pointing to our Dial Plan:
That's it. Very simple. Now dialing 37063042439 we reach our Dial Plan. Here we will hear message which asks to
enter device PIN. If PIN is correct – we can dial desired destination.
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Auth. by ANI
This method is variation of Auth. by PIN method. There's one setting which changes the behavior of the system to act
in different way. When you check Use ANI? field in Dial Plan creation window - then when user will dial in his CallerID will be checked. Only if CallerID is not found – he will be asked to enter his PIN.
CallerIDs are entered for every device in device settings using icon.
ANI - Automatic Number Identification
ANI is action which is used to authenticate who is calling by CallerID.
When call comes into MOR, CallerID is checked with all available CLI's in DB. If match is found - call is authorized and assigned to the device which CLI is matched against CallerID of the call.
All other details are the same as for Auth. by PIN dial plan. We are creating this dial plan same way, just check Use ANI?:
And in main Dial Plan window we have:
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Note the difference between Auth. by PIN and Auth. by ANI dial plans.
Assigning DID is similar as to Auth. by PIN method.
ATTENTION! Be careful because it is very easy to fake CallerID.
Calling Card dial-plan
This dial plan covers Calling Card functionality. It is covered in appropriate chapter.
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Callback From Wikipedia, 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 in, 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 recognise 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 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.
Common Callback diagram
Implementation in MOR
Callback in MOR is implemented in very simple way. It's enough to tell system Access Number (DID) and the way how Callback should act. All of this will be described in following paragraphs.
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Who pays for not authenticated call?
First of all by the nature of Callback it is important to know, that before calling back to User System does not know
who is the User (when not using ANI) – e.g. is it client, has he an account in the system and so on. That means – when system makes callback to the user and user pickups – the call is ANSWERED, but still – system does not know who the user is. Let's say user tries to enter PIN several time but unsuccessfully. Then he hang-ups. Call is ANSWERED. Does user pays for this call? No. Who pays? You, as owner of the system. Sad but that's true about Callback.
Callback User, Callback Device and Callback Tariff
In order to keep track of Callback calls you need to configure MOR in proper way. For that 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. In 100% of the time this user is system owner. But creating new user in MOR let's you keep track of all Callback activity in your system. The
other reason to have Callback User (and not use System Admin account) is ability to create and assign Callback Tariff to this user.
Callback Device is the necessary entity in MOR which will actually 'make' callback calls to user. Type and all details do not matter. This device is pure virtual. Callback Tariff – separate tariff which should be assigned to Callback User. In this tariff you enter rates only to
these destinations from which you expect user will make Initiation Call. This is important! Make sure you do not have some Inmarsat rates defined here. Guess how much you will pay when user initiates callback from Inmarsat and when he gets call he enters PIN several times incorrectly and hang-ups? In short – this tariff decides from which destinations you allow users to initiate callback.
Types of Callback in MOR
Callback in MOR can be divided in three separate parts by the way user is authenticated/authorized: 1. By User Device's PIN
2. Using ANI (Automatic Number Identification)
3. By Calling Card's Number/PIN (or only PIN)
We will show how to configure all three ways.
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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 last type of Callback let's 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.
Let's mention again: 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 workflow 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
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Step-by-step example
This example will demonstrate all steps outlined before to create all 3 types of Callbacks.
1. Let's create Callback User:
2. Then Callback Device:
Device's settings are not important at all – leave everything empty. They will not be used anyway.
3. Callback Tariff:
It is worth to note that Callback Tariff can be Retail or Wholesale. Here in example we made Retail tariff.
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Here we added rate only to Lithuania MOB because we will test it with it. That means – we are allowing user to receive callback only to Lithuania MOB number.
4. Assign Callback Tariff to Callback User:
5. Calling Card and Auth. by PIN Dial Plans should be created in order to proceed with Callback.
Here we will create 3 different Dial Plans: each one for different Callback type. Each Dial Plan should have DID assigned to him.
Here we have 3 different Dial Plans. Each one for different type of calling.
Calling Card Test DP is associated with DID 3703721028. Dialing this DID we will be able to use our Calling
Card.
Auth. by ANI DP is associated with DID 3703721027. Dialing this DID system will try to
authenticate/authorize user by his CallerID. If it does not succeeds – will ask for PIN.
Auth. by PIN DP is associated with DID 3703721026. Dialing this DID system will ask for a PIN.
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In DIDs section these Dial Plans look like this:
Important! These Dial Plans are active by themselves. E.g. Dialing associated DIDs you will get service described by these Dial Plans. Callback is not configured yet.
6. Now we will create 3 Callback Dial Plans to demonstrate each of 3 Callback types:
Here we created 3 Callback Dial Plans and associated each of them with different Calling Card/Auth. by ANI/Auth. by PIN Dial Plan.
Wait Time means how many seconds to wait after Initiation Call before calling back to user. Do not forget to assign Callback Device to Initial Callback device as shown in example!
7. Assign DIDs to Callback Dial Plans
Currently we have 3 different Callback Dial Plans but we can't reach them because we do not have Access Number (DID) for each of them. Let's add them in DIDs section:
And in Dial Plan section same info in other perspective:
In Assigned DID(s) section you can see assigned DIDs for Callback Dial Plans. What does it mean?
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Let's review what we had done just now. Callback for PIN line shows us that when user dials number 37041201026 he will initiate callback. He gets called back and is connected to DID 37037210226 which is Auth. by PIN Dial Plan. Here he will be asked for PIN. That's Callback with Auth. by PIN.
Callback for ANI line shows us that when user dials number 37041201027 he will initiate callback. He gets called back and is connected to DID 37037210227 which is Auth. by ANI Dial Plan. Here user's CallerID will be checked and if not recognized he will be asked for PIN. That's Callback with Auth. by ANI. Callback for CC line shows us that when user dials number 37041201028 he will initiate callback. He gets called back and is connected to DID 37037210228 which is Calling Card Dial Plan. Here he will be asked for Number/PIN of
the Card. That's Callback with Calling Card. This finishes example about various types of Callback and how to setup them.
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WEB Callback You can find Web Callback in PERSONAL MENU – Various - Callback.
Main window let's you enter callback details:
When you enter source (and destination if you wish) press Execute and you will get call to Source after that you will be connected to Destination (or asked to enter destination and only then connected).
Callback from HTTP request
Second callback initiation method is through HTTP request.
It is useful for integration callback service to other Web portals, customer pages and similar.
Just use line in format:
http://<YOUR_SERVER_IP>/mor/functions/callback_from_url?acc=<DEVICE_ID>&src=<SOURCE>&dst= <DESTINATION>&secret=<DEVICE_SECRET>
Enter appropriate values where applicable. You can omit <DESTINATION>. Source will be asked to enter it if you do so.
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Click2Call Click2Call works in following way:
1. You press on the icon next to phone number. 2. You phone rings 3. When you pickup you phone – you are connected with number you pressed.
When you press on the icon next to phone number the small popup window appears:
It shows you which number you will be dialing when you will pickup your ringing phone. You Primary device should
ring. It is marked with icon in your device list.
If you get such message:
that means something is very wrong. You can try to logout/login – that could fix possible cookie problem. Also this error can occur if you do not have selected primary device. These small windows automatically disappear after 5 seconds.
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Quick Forwards
Some DIDs can be assigned to Quick Forwards functionality. When it is done each user can enter different number to be forwarded when he calls some DID.
Main advantage of this function is to use limited number of DIDs to let 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
Now we have 2 users with such CallerIDs:
User A with CallerID 8888888888
User B with CallerID 9999999999
User A set such forwards:
From DID 1111111111 forward to 3333333333
From DID 2222222222 forward to 4444444444
User B set such forwards:
From DID 1111111111 forward to 5555555555
From DID 2222222222 forward to 6666666666
When User A dials DID 1111111111 his CallerID (8888888888) is checked, user is recognized and call is
automatically forwarded to number 3333333333
When User A dials DID 2222222222 his CallerID (8888888888) is checked, user is recognized and call is automatically forwarded to number 4444444444
When User B dials DID 1111111111 his CallerID (9999999999) is checked, user is recognized and call is
automatically forwarded to number 5555555555
When User B dials DID 2222222222 his CallerID (9999999999) is checked, user is recognized and call is automatically forwarded to number 6666666666
IMPORTANT: Correct CallerIDs should be set for users in order to recognize them. Only system admin can do that.
If system can't recognize the caller it asks to enter PIN number of the device.
If user is dialing DID which has not assigned Forward Number, user is asked to enter such number.
User can manipulate his Quick Forwards in his menu. He needs to login to GUI to change numbers.
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Accounting Accounting in MOR is implemented to keep track of all payment activity. Also it is possible to create various services make subscriptions for them and generate invoices for the clients.
Services Service is 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
You can create unlimited number of Services. All of them can be found in SETTINGS – Accounting – Services:
Currently only service could be paid in every month. That represents Type = Periodic fee and Period = month. Each Service should have a price.
As usual in order to create new Service press on New service, to edit – on and to delete it.
Subscriptions Users can subscribe for some services. When they do so – they get Subscription for some service. Subscriptions for
the services can be found for each user in Users main window by pressing icon for some selected user:
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Subscription can have some Memo (if you sell DID for example – you can write it's number here). Also it has important fields – Activation start and Activation end. This data interval will be billed in invoice. E.g. date before and after of this interval will not be billed.
You can add/ edit/ delete subscriptions for user.
It is not recommendet to delete them – otherwise user will not be billed for that subscription. If user stops using this service – make Activation end value to the date when he does it.
NOTE: service is billed from Activation Start till Activation End. And from invoice period start till end. Time period which fits in both previous periods is actual period which will be billed.
Invoices
All settings for Invoices can be set in Configuration window Invoices section.
Invoices can be generated for Postpaid users.
Invoices can be viewed and created in SETTINGS – Accounting – Invoices. In the main window you can see all invoices:
Icons or shows is this invoice Paid or Unpaid. When you mark invoice as paid – new payment is created for
that user and his balance updated by invoice's amount. When you mark invoice as not paid – payment for that invoice is destroyed and user's balance is updated accordingly.
Press on icon to get details for invoice:
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Here you can view all invoices' details and press on or icon to generate invoice files in PDF or CSV formats accordingly.
In main Invoices window press on Generate Invoices to generate additional invoices.
ATTENTION! This should be done only on first days of new month to cover all costs for previous month.
Here we select for which month we want to generate invoices.
If trying to open PDF invoice you get error - make sure your logo PNG file is without ALPHA layer - that means without transparency layer.
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Invoice by CallerID
This type of invoice groups all calls by CallerID. 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 VAT)
Total: 150.4886 LTL (Without VAT)
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 VAT)
Total: 26.9805 LTL (Without VAT)
Monthly fee - test memo2 10.0000 LTL (Without VAT)
TOTAL TO PAY: 187.4692 LTL
Manual payments Sometimes customers send money through bank or by other ways. Or using some online gateway not implemented into MOR. To handle such transactions MOR has Manual Payments. When user sends you as an owner some amount you need to enter this amount into MOR using Manual Payment window which can be found in SETTINGS – Accounting – Payments – Add manual payment:
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First window lets you choose user and enter amount (with VAT) in selected currency:
After we press on Proceed we see calculated values for this Payment:
We need to press on Confirm to enter this payment which can be seen on Payments window:
After payment is entered users account will increase and he will see such payment in his Payment sections. Administrator can delete this payment for user. After deleting the user's account will decrease accordingly to payments amount.
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Vouchers Vouchers are used to fill the balance of some user. You can sell vouchers printed on paper of over internet. When user logins and enters voucher number – his balance will increase by the value issued for this voucher. All voucher configuration options are in Configuration window Vouchers section.
You can view created vouchers in SETTINGS – Accounting – Vouchers:
This window show status of voucher: Active? – shows if we can use it. All other info is self-explanatory.
In order to add new vouchers press on Add vouchers:
Here you can enter necessary information to generate array of vouchers.
Tag field is used just to mark series of vouchers – it is just for informational purposes or easier search. When user logins he sees voucher icon on the main page:
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He can click it and enter voucher number. He can try up to X times where X is set in configuration menu. If he fails to enter correct voucher number all these times he will loose possibility to enter voucher number again for some time. The values how many times he can try and how long to disable attempts to enter are configurable in Configuration menu.
When user enters correct voucher number – his balance will increase, he will get new payment record and this voucher will be marked as used.
ERROR: Vouchers are not accepted
Make sure Active Till is tomorrow or more distant in the future.
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PayPal payments User can pay for the services or increase his balance using PayPal. MOR is fully integrated to PayPal and does not need any manual intervention from administrator.
All PayPal options can be set in Configuration menu PayPal section. PayPal payment option can be reached pressing on the PayPal picture in the main entry window:
When user clicks on this image he will get possibility to enter desired amount he wishes to pay:
Note the minimum value which can be set in Configuration menu. If user enters lower value – he will not be able to proceed.
After pressing on Confirm we get another window which shows us how much we will pay:
After pressing on Pay we go to PayPal webpage where we enter our account details and confirm the transfer.
After PayPal completes the transfer user gets back to the MOR interface. Now under his payments he will see PayPal payment with some Status about his payment. PayPal payment can have many reasons why it is not completed. For better understanding all this info is presented in Payments page: *Possible reasons of not completed PayPal payment: echeck: The payment is pending because it was made by an eCheck that has not yet cleared.
intl: The payment is pending because seller holds a non-U.S. account and do not have a withdrawal mechanism. Seller must manually accept or deny this payment. multi-currency: Seller do 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 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.
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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 – user's balance will increase by the value (without VAT) he paid by PayPal.
This integration let's have fully autonomous system where user can register online, pay using PayPal, configure his device and dial instantly WITHOUT any intervention from systems administrator. ATTENTION – in order PayPal to work – your server should have real IP and should be reachable from the internet.
WebMoney payments
This payment gateway is popular in Russia and neighboring countries. Now MOR users can use http://www.webmoney.ru/ to pay for services.
Same logic applies as in PayPal payments. Settings are in SETTINGS - Setup - Payments - Webmoney:
Enabled - enable or disable it (hides logo on front page)
Purse - purse ID from WebMoney
Currency - purse's currency
Default Amount - which amount is default for user who wants to pay by WM (he can change this amount)
Minimal Amount - minimal amount which user can pay over WM
Test - should it be test or real payment
Test mode - describes how many payments are successful in test
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Reporting MOR has many reporting screens. Using them you can check various information. Some reports are very CPU intensive – a lot of data is gathered from database and counted. It is not recommended to
run these reports in operational hours if MySQL DB is on the same server as Asterisk.
Provider's calls
In main Providers window press on to open Provider's calls page:
Here you can see call for selected provider in selected time period. You can change typr of the call. Also you can
export selected calls to PDF or CSV format.
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Provider's stats You can reach this report in STATISTICS – Various – Providers. This report shows all provider's stats at once:
In selected period you can check various stats about all providers in your system. If you click on number in Answered/No Answer/Busy/Failed column – you will open window where these calls will be listed. ASR - Answer Seizure Ratio
Its calculated by taking the number of successfully answered calls and dividing by the total number of calls attempted
(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) ACR – Average Call Duration
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User's calls This report shows all calls user made is selected period. It is possible to export this list to PDF or CSV format. It can be found in STATISTICS – Calls – Calls:
In search section you can select direction of the calls:
Outgoing
Incoming
status of calls:
Answered
No Answer
Busy
Failed
Missed – No Answer + Busy + Failed
and device for user.
User call's stats
These stats can be found in STATISTICS – Calls – Call Stats. They show a lot of data for selected user. All data is self-explanatory.
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New call's stats New calls per day mean that such numbers have NEVER been dialed before. This function is useful in Call Center environment.
New calls can be accessed in STATISTICS – Calls – New calls. Main window shows how much new calls each user made in selected day:
Icon shows each call in the list.
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Loss making calls Sometimes you import/put rates for provider lower then for the user. In such cases when the call is made you are taking loss. To find such calls you should use this function. Just select desired period and all Loss making calls will be visible. You can found this report in STATISTICS – Calls – Loss making calls:
This window shows you loss making calls in selected period. If you click on user or provider rate – you will go directly
to rate's edit window and edit it.
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Hangupcause codes
MOR application saves hangupcause codes to DB so it is easy to check what the error was and why call failed.
These codes can be seen in 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 207 Not clear who should receive call 210 Balance > 0, but not enough to make call 1s in length 211 Low balance for user 212 Too low balance for more simultaneous calls 213 Low balance for DID owner 214 Too low balance for DID owner for more simultaneous calls 215 Low balance for reseller 216 Too low balance for reseller for more simultaneous calls
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Payment stats This report shows all payments made. It is good to review all you payment history. Search is possible over some properties. You can found this report in SETTINGS – Accounting – Payments:
The payments can be various type:
Manual – entered through GUI
Invoice – when invoice is marked paid
PayPal – when PayPal payment received
Voucher – when voucher is used
Only manual payments can be deleted from this window. Other payment types can't be deleted directly. Invoice type payment is deleted when invoice is marked non paid or deleted, Voucher type payment is deleted when voucher is marked as non active or deleted. PayPal type payments can not be deleted.
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Finances Finances window lets view all user's financial situation in one window. It can be reached in STATISTICS – Finances:
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Profit stats It is important to know in any moment the profit your system generates you. It can be done in STATISTICS – Finances – Profit:
All fields are self-explanatory. You can check profit in some period and for selected user. PDF report shows same info.
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Country stats This report shows call by countries. It can be foud in STATISTICS – Various – Country Stats:
You can select time period and user.
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Fax2Email stats This report show all info about received (or not) faxes. It can be found in STATISTICS – Various – Faxes:
All users are visible here. Fax can have 3 states:
Received – everything ok with this fax – it's received and sent to appropriate emails
Corrupted – fax is received but it's corrupted – e.g. no good to send to email
Mistaken – that was simple call to fax number not a fax
You can press on number for each state to get list of these faxes:
Pressing on you can get this fax in PDF format.
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Realtime user stats Realtime is used in Call Center in big screen on the wall to show agents their current performance during work-day. You can find Realtime in STATISTICS – Various – Realtime.
Press on Most called users.
Realtime window updates every 60 seconds. In it you see all users with their stats:
How to view user's stats in Realtime Stats
Add user to some Group
Go to: MANAGEMENT - <Group Name> - Member Stats
Check Show in Realtime Stats
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Realtime Stats Usage
This window can be shown on big LCD screen for Call Center agents to check their performance as seen in photo on one Call Center in Lithuania:
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Active Calls
On main window in Quick Stats it is possible to see Active Calls - these are the calls, which are in the system at this moment.
It is possible to press on it and new window opens - with detailed Active Calls.
You can get to this window in STATISTICS - Various - Realtime - Active Calls.
In this window all active calls are presented with more details, such as: Start Date, Duration, Src, Dst.
In order to see correct duration on multi-server systems, all time on servers should be synchronized.
In order to see Active Calls in GUI, in file /etc/asterisk/mor.conf you should have:
active_calls=1
NOTE: local calls are not visible on MOR GUI, only calls to/from outside
User login stats
This report shows when user logged into the GUI. Note – not to the system with device, but to the GUI with his username and password. It can be found in STATISTICS – Calls – Logins:
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In selected period you can see when user logged in and logged out. Also representation in graphical form.
Quick stats This is not quite a report but a quick look over some stats. It can be find in main entry window. You can reach it at PERSONAL MENU – Start Page:
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Addons Addons to the MOR are not included into standard MOR package. They should be purchased and installed separately.
Calling Cards 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
Calling Card module lets manage cards printed on plastic/paper or sold over internet for international calling.
The calling card platform is dedicated to calling cards - e.g. the plastic/paper cards with scratch surface which can be
sold in shops or in streets. Also these cards (or their number/pin) can be sold over internet. When user buys a card, he gets number/pin of the card. Then he dials some access number and IVR asks him to enter this number/pin. Then user hears credit remaining and time remaining and can enter destination. If destination is ok - he gets connected and can talk. After call is finished – the call's price is deducted from card's balance.
Differences between Users and Cards in MOR
There are 2 main ways how person can use MOR system to dial out:
1. Using his User account 2. Using Calling Card
The main difference is that 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 at all. That means:
3. User (as entity in MOR) can't have Cards. 4. Card (as entity in MOR) can't belong to some User.
System Setup
1. First Card Group should be created with correct details 2. Cards for Card Group should be generated
3. Calling Card Dial-Plan should be created 4. DID should be assigned to this Dial-Plan
All these steps are described in details further.
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Work-flow
User buys Calling Card:
a. Over internet
b. Printed in the store/street/etc User finds NUMBER and PIN of the Card – it can be on the plastic/paper Card or given to user in Online Calling Card Store
User dials access number – this number is given to user same way as NUMBER/PIN. User can dial this number from anywhere in the world. System asks him (user) to enter NUMBER and PIN (or only PIN) – When user dials to access number – IVR will ask him to enter card NUMBER and/or PIN.
NUMBER and/or PIN are/is good – in this step system checks database to see if Card with provided NUMBER/PIN exists. If Card is not find – user is asked to enter new NUMBER/PIN. It is possible to change how many times system
will let to reenter NUMBER/PIN. Default is 3 tries. System tells remaining balance on the card – this step is optional it can be disabled.
System asks to enter Destination – IVR will ask user to enter destination he wish to dial. System tells remaining time – based on entered destination and remaining balance system will tell user remaining time he can talk with this destination. This step is optional. It is influenced by Ghost Minute Percent. System dials Destination – using LCR system chooses cheapest provider and dials entered Destination/
After the call it's price is deducted from Card's balance – self explanatory.
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Card Groups
Card Group is set of Calling Cards with same properties as:
Image Description Starting price (Price)
Setup fee Ghost minute percent Daily charge VAT percent
Least Cost Routing Tariff Location
Number length PIN length
Image – just for informational purposes and eye pleasure to view on the GUI. Could be (and should be) same as real printed card for consistency. But that's not the rule.
Description – short memo for admin to view some more info for the cards. It is useful when there are a lot of Card Groups and Card Group name is not enough. Starting price (Price) – the price which all Cards in this Group will have before first use. It does not include VAT. It's in default currency (!).
Setup Fee – the fee which will be charged when call is connected EVERY time when the card is used. This is similar to
Connection Fee used in Tariffs but is widely used by Calling Card providers. Ghost minute percent – the value to trick the user to think he has more minutes when he really has less. Example: user has enough balance in his card to call for 100 minutes. Ghost minute percent = 1.3. So when user enters his Number/PIN system tells: "You have 130 minutes remaining". Note: this is not increase in percent, this is the value, by which actual minutes should be multiplied to tell them to user.
Daily Charge – the amount which will be deducted EACH day from the card starting from the date when card is used for the first time. That means, if user makes first call on Monday and second call on Friday, his card will be deducted for 5 days using this value (Mo, Tu, We, Th, Fr). VAT percent – actual value which will be used when presenting Card's price to the User. E.g. when telling user his
remaining balance of the Card or showing same info on the screen.
Least Cost Routing (LCR) – which LCR will be used for all Cards in this Card Group. Tariff – price list which will be used for this Card Group. Location – this is useful when Cards will be sold in some region with specific dialing rules. Rules for location can vary for different Calling Card Groups.
Number Length – the length of Card's number. You can select any value here. It is recommended to select different values for different Card Groups to minimizing risk of guessing other card and using it unauthorized. Number for Card is generated in sequence order. PIN Length – number of Card's PIN. Same recommendation applies as for Card's Number. PIN is generated
randomly.
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Card Group Management
Card Groups as many other entities in MOR can be created/edited/deleted/viewed.
If Calling Card module is enabled you can find it (as Administrator) in: SETTINGS – Addons – Calling Cards:
Here you can review your active Calling Card Groups.
(..) – Shows Cards window where you can manage Group's Cards
- All of the details of Calling Card Group
- Exports Cards into CSV format
- Edit Group's details
- Delete Group
Cards in CSV file
When you want to print Cards on plastic/paper you need to provide necessary information for Card Printing service.
This could be done using CSV file. Pressing on icon let's you download all Card info in one file. You can process
this file later in MS Excel or other compatible software – make correct format for your Card manufacturer and send it to him.
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.
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Calling Card Groups Image
Unfortunately there is no way to upload Cards' images over web. You need manually add JPG or PNG images to /home/mor/public/images/cards. Later when creating or editing Group, you need to enter image's name into the appropriate field.
Cards
In this window you can manage all Cards which belongs to the Calling card group. All fields are self-explanatory.
Adding new Cards
Add cards – brings the window where you can add additional Cards to the Card Group:
In this window you should enter Card's start and end numbers. They will form interval which will be created. If any
Card of same number exists – the old Card will remain and new one will not be created. Card PIN number is created automatically – it's random and unique. It is wise to keep it's length not small to avoid cheating from users which can guess short PIN number.
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Create Dial Plan
When we have Calling Card Group and Cards generated we can create Dial-Plan for this Group.
Dial Plan can be treated as set of rules how to handle particular Calling Card Group. You can set different rules for same Calling Card Group. This entity (Dial Plan) let's you to have more control over Calling card Groups. Let's go to SETTINGS – Billing – DIDs – Dial Plans:
In this image we can see already created Dial Plan for Calling Card Group.
Let's review each field: Name – just for informational purposes Number length – tells IVR how much digits to wait for NUMBER which is asked to be entered
PIN length – how many digits for PIN to wait Ask only PIN – as PIN is unique it is sometimes wise to ask only PIN because users can make mistakes and get annoyed entering too much digits
Tell time – should IVR tell time after Destination is entered?
Tell balance – should IVR tell balance after user enters NUMBER/PIN? Ask Number/PIN (times) – how many times user can try to enter Number/PIN Ask destination (times) – how many times user can try to enter Destination
Assigned DID(s) – just for informational purposes to see which DID(s) are assigned to this Dial Plan When creating new Dial Plan you will need to enter correct Number and PIN length values – e.g. of existing Calling Card Group's. Otherwise you will not be able to create Dial Plan.
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Assign DID to Calling Card Group Dial Plan
When we have created Dial Plan for some Calling Card Group – we can assign one or more DIDs to this Dial Plan.
DID or in other words – Access Number is vital for using Calling Cards. User will use this number to reach IVR. You can assign unlimited DIDs to same Dial Plan. DID can be assigned to one Dial Plan only. It is useful when you are selling Calling Cards in different regions – e.g. countries. So in country A user will dial one number (which is cheaper for him or even free) and in country B user will dial another number. They will reach same IVR and will use same service but they will save on telephone fees.
SETTINGS – Billing – DIDs:
We will press on icon for the first DID:
We select our Dial Plan and press confirmation icon to assign our Dial Plan to this DID:
Now user can dial number 37037210220 (in example) and he will be able to reach IVR and use his Calling Card.
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Online Calling Card Shop
MOR has simple Calling Card shop where you can sell Calling Cards. It can be reached by same IP as MOR, just using 'ccshop' instead of 'callc'. E.g. http://<domain/ip>/callc will lead to MOR when http://<domain/ip>/ccshop will open Calling Card Online Shop:
All Card Groups are shown here with their images. If we want to buy one card we enter amount and press Add to cart button for that Card:
Shopping cart show what we selected to purchase. We can continue shopping, empty cart or checkout.
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Let's go directly to Checkout because all other functions are very simple:
Pressing on PayPal button you will be directed to PayPal site where you will be able to pay for the Card.
Calling Card Owner portal
In Calling Card Online Shop it is possible for user to enter his NUMBER/PIN and get his own portal:
If in environment.rb you will make CC_Single_Login = 1, you will get:
That's up to you which one to use.
Let's enter Card's details and login to Card's Owner Portal:
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Here you can see Card's info and calls made using this card.
Step-by-Step example
In order to show all the Calling Card set-up process we will go through small step-by-step tutorial. Here we will
configure Calling Card service from the start. Let's say we have a 2EUR Card:
We need to create Calling Card Group to handle all such Cards.
First of all let's upload our Card's image to /home/mor/public/images/cards:
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Now we go to SETTINGS – Addons – Calling Cards and press on New card group. In new window we enter
our Card's details:
Here we should mention Price with VAT field. It should be in default currency which can be found bellow and in our case it's USD. Because of that – we have to convert Card Group's price to USD. After creating a Card Group we will see all it‟s details:
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Now when we have Calling Card Group we need to generate some Cards. Let's press on Add cards:
Here we will create 10 Cards starting from 2000000001:
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Now when we have Calling Card Group and Cards we need to create rules how to handle all these cards. For that we
need to create Dial Plan. Let's go to SETTINGS – Billing – DIDs – Dial Plans and press on New Dial Plan. In
new window we enter following details:
Name is just for informational purposes, NUMBER and PIN length should match our just created Calling Card Group's settings. We want IVR to ask for number AND pin, so we leave Ask only PIN unchecked. IVR will tell us remaining time and balance. And will ask for number/pin and destination 3 times most:
In last window we see that we do not have DID(s) assigned to our Dial Plan, so we go to SETTINGS – Billing –
DIDs and select some of available DIDs and press on it's icon:
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Here we select our Calling Card Test DP and press on confirmation mark . New window shows status about our action:
That's it. Calling Card engine is ready to serve calls. In our example number 370523456 is Access number to be dialed to reach IVR.
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Auto-Dialer Auto-Dialer is system which automatically dials set of numbers and if number is answered – plays some prerecorded message. The minimal infrastructure for the Auto-Dialer is shown bellow:
User interface
When Auto-Dialer (AD) is installed – every user can use it.
User can find AD menu under PERSONAL MENU – Auto Dialer. Main window shows user's campaigns:
User can Activate or Stop the Campaign by pressing or button near Campaign‟s status. Campaign can only be Activated if it has free (not-called) numbers AND some Actions. So please before activating
Campaign import some Numbers and create Actions.
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Add new campaign – let's you add new campaign:
Important fields here are:
Start/Stop Times – when calls will be made
Max Retries – how many times to retry unsuccessful call
Retry Time – after how many seconds should system retry the call?
Wait Time – how long to wait till the call will be answered
Device – each Campaign should be assigned to Device in order for MOR to bill the call
Numbers
Press on in Numbers column to check campaigns numbers:
Here you can see all numbers and status for these numbers. User can import phone numbers from TXT file. Numbers should be in correct format.
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Press on Import numbers from file and you will have to select file to upload:
After uploading file you will see status of imported numbers:
You can delete all number by pressing Delete all numbers.
If number is Completed, it‟s possible to reactivate it (tell system to call it again pressing icon near number.
Actions
Press in Actions column to access Campaign's actions:
In this window user programs the Actions for Campaign:
Types of Actions:
WAIT – how many second to wait
PLAY – play some file
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When Action is created it‟s possible to edit its settings by pressing on or delete action by pressing on .
For various types of Actions you edit different settings:
WAIT – enter new wait time in seconds
PLAY – upload new file to play
Admin interface
When Auto-Dialer (AD) is installed Admin can find Auto-Dialer under menu SETTINGS – Addons – Auto Dialer. Here he can get view of all users‟ campaigns:
The important info for each Campaign:
Status – is Campaign Active or Stopped
Run time – tells when calls will be made
All other fields are self-explanatory
It‟s possible for admin to see the Actions for users‟ Campaigns. Just press on Actions button.
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Reseller Addon
Description
This addon incorporates basic reseller support into MOR PRO
Currently it lets to create user with type 'reseller'
When this user logs into the system he is able to:
1. Create tariffs 2. Create users 3. Assign tariffs to users 4. View calls and other stats for his own users
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
Currently DIDs can be assigned only by admin. E.g. reseller can't assign DID to his user. Example:
ADMIN - RESELLER - USER
ADMIN can assign DID to USER directly bypassing RESELLER.
NOTE: This functionality will be adjusted in the future to let resellers add DID directly.
PayPal
PayPal is disabled for reseller's users. Only resellers and users which belong to system admin can use PayPal to pay for services. It will be available in future versions.
Manual
Manual for Reseller Addon made by TuT Telecom can be found here.
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Click2Call Addon
Concepts
Client - interested person who enters his number in the banner
Device - Operator - person (worker in company) who answers calls
Campaign - virtual entity which describes one marketing campaign with several properties such as who have
to answer the call, etc
User - Company - one separate organization which can have many Campaigns
Campaign
It is virtual entity which describes 1 marketing campaign.
Relations:
User can have many campaigns
Campaign belongs to only 1 user
Campaign has 1 device(operator) which answers to this campaign's calls.
Campaign can specify comment type. E.g. Campaign's comments can have specific fields.
Campaign can be (by user):
Created
Edited
Deleted
To create C2C Campaign (Campaign) user has to login using his username and password.
Then PERSONAL MENU - Various - Click2Call and press on Add new campaign
Here user can enter Campaign Name and Description.
Operator describes which device should be used to answer the call from the client. Usually it is Virtual device from which Forward is set to some real Mobile or other Companies real phone.
First Dial sets to which party call should go first. Default is Client. E.g. - when client enters his number in some banner and call is initiated - client gets call first. When he answers - call goes to company.
Comment
Each Click2Call Call can have comment.
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Comment can have different fields. It is possible to setup comment fields for each Campaign separately.
To setup comment fields for each C2C Campaign press on Comment Fields icon in Campaigns window.
Here you can add Comment Fields. Each of them can have Name and Description.
Type can be text, textarea or checkbox. They describe the data you can enter for each field.
text - for short text (name, city...)
textarea - for big text (description, problem, note...)
checkbox - can be checked or unchecked (underage, sale...)
Press on Change comment field order to reorder Comment Fields. Just Drag&Drop them to new place.
Greeting
When Call is initiated and and Client answers he hears some Greeting. It is assigned to C2C Campaign when this campaign is created.
You can upload your custom Greeting. It should be in WAV or MP3 format and it can't be bigger then 5mb in size.
System setup
Create User/Device (Company/Operator)
Create some Virtual device for the Company and forward it to your phone which should be used to serve the calls.
NOTE: you can use SIP/IAX2/etc devices if you want to use VoIP phone to serve calls.
Allow this User to have Click2Call (C2C) Campaigns:
Go to SETTINGS - Addons - Click2Call and select user which should be allowed to have C2C Campaigns. Subscribe this user and he will be able to manage his C2C Campaigns in his login window.
Create Campaign
o Configure Comments for campaign o Upload custom Greeting
Click2Call initiation
On webpage user sees banner where he enters his phone number.
o By this action he agrees that he will receive call from company which is advertising in this banner.
Banner send HTTP POST request to MOR server. o c2c_callback - Initiates Click2Call specific callback
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Call destination control
Destinations which can be dialed have rates in tariff which is assigned to users/companies.
Destinations without rates will not be dialed. o API will return error code that this destination is not supported.
Example: user enters number starting with 900…
We do not have rate for such call in our tariff so call will not be placed.
Call
System dials user first by default (if not set otherwise). o It is possible to choose where to call first: user or company.
When user pickups the phone:
o System plays some prerecorded message to user (asking to wait, thanking for the calls, etc) o System dials to compamy
When company pickups - calls are bridged between user and company.
Failed call handling
If call can't be placed to user:
o It can be redialed X times doing Y length pauses between redials
If call can't be placed to user X times call is visible in companie's GUI as FAILED call o Company can press on this call and callback will be initiated
First company will be dialed
When company answers - user will be dialed When (IF) user answers - calls will be bridged
Statistics
Each company is able to see initiated calls with their status: COMPLETED/FAILED CALLING TO USER/FAILED
CALLING TO COMPANY
Each call can have comment
o Comment can have separate fields which can be set differently for each comment type o Call lists with comments can be exported to CSV
Call has field Direction which is based on user's number - tells from which country, city user is (or which
mobile priovider)
Each Click2Call call consists of 2 separate real calls - to user and to company. Their stats are shown.
Call summary stats - how many calls in selected period, completed/failed user-company, etc +realtime
Diff. time interval comparison.
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API MOR API lets other software interact to MOR without knowing internal structure of DB and other logic. MOR accepts requests by HTML POST and returns result in XML.
Login
login
Logs in user into MOR Request
HTML POST: http://<SERVER_IP>/mor/api/login?u=username&p=password Result
<?xml version="1.0" encoding="utf-8" standalone="yes"?>
<action>
<name>login</name>
<status>ok</status>
</action>
Everything is ok. User is logged in to the system.
<?xml version="1.0" encoding="utf-8" standalone="yes"?>
<action>
<name>login</name>
<status>failed</status>
</action>
This means that wrong username and/or password and user can't be authenticated.
Logout
Logs out user from MOR
Request
HTML POST: http://<SERVER_IP>/mor/api/logout?u=username&p=password Result
<?xml version="1.0" encoding="utf-8" standalone="yes"?>
<action>
<name>logout</name>
<status>ok</status>
</action>
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Everything is ok. User is logged out from the system.
<?xml version="1.0" encoding="utf-8" standalone="yes"?>
<action>
<name>logout</name>
<status>failed</status>
</action>
This means that wrong username and/or password and user can't be authenticated and logged out from
the system.
Callback
callback
Initiates callback Request
http://<YOUR_SERVER_IP>/mor/api/callback?u=<USERNAME>&p=<PASSWORD>device=<DEVICE_ID>
&src=<SOURCE>&dst=<DESTINATION>
USERNAME/PASSWORD - for user, not device!
DEVICE_ID - which device should be used to initiate callback and which CALLERID will be set on the call
SOURCE - which number to dial first
DESTINATION - will be dialed when SOURCE will answer
You can omit <DESTINATION>. SOURCE will be asked to enter it if you do so.
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>
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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 FAILED call and reason why it failed
Check Asterisk CLI for error
Use Call Tracing to locate error and fix it
c2c_callback
Initiates Click2Call specific callback Request
http://<YOUR_SERVER_IP>/mor/api/c2c_callback?u=<USERNAME>&p=<PASSWORD>campaign_id=<CAMPAIGN_ID
>&client=<CLIENT>
USERNAME/PASSWORD - for user, not device!
CAMPAIGN_ID - for this campaign this c2c callback is
CLIENT- cient's number
All fields are necessary to fill. 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 campaign <Status>Bad campaign</Status>
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Campaign does not exist or not belong to this user No client number <Status>No client</Status>
No cient number in request - nowhere to call Call is not received
Check your CDR for FAILED call and reason why it failed
Check Asterisk CLI for error
Use Call Tracing to locate error and fix it
Accounting
invoices
Retrieves list of invoices in selected time period.
Request If user has 'admin' rights - he will get all invoices from selected period, if user has 'user' rights - he will get only his invoices in selected period. HTML POST:
http://<SERVER_IP>/mor/api/invoices?u=username&p=password&from=1188604800&till=1191196799&lang=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 in which language do 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> </Product>
</Invoice>
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<Invoice agreementnumber="0000000000" user_id="0" clientid="240887380"> <Product> <Name>Calls</Name> <Quantity>65</Quantity> <Price>112.3324</Price> </Product>
</Invoice> </Invoices> Errors
<Error>user not found</Error> <Error>no invoices found</Error>
Other
Branding How to change logo is described in Configuration section.
If you want to change CSS properties – main CSS file is in /home/public/stylesheets/mor.css
Stylesheet for Calling Card shop is: /home/public/stylesheets/ccshop.css
Security
After MOR installation you need to change MySQL passwords for root (which is empty) and for mor (which is 'mor').
For user mor you need to change password in following places:
MySQL itself
/home/mor/config/database.yml
/etc/asterisk/mor.conf
/etc/asterisk/res_mysql.conf
/var/lib/asterisk/agi-bin/mor-recordings.php
/home/mor/public/c2c/c2c.php
How to change admin password
Login as admin
Go to SETTINGS - Users
Press on Edit for System Administrator
Enter new password
Press Change
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NOTE: Also change password for user 101 (default password is 101).
Change DB name
If you want to change DB name from mor to another one, make sure to edit /etc/asterisk/extconfig.conf and change mor to your new DB name. (Also change database in all other mentioned places).
Various
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 right way?
First execute this command:
asterisk -vvvvvvrx 'stop gracefully'
This command will not let any new calls to begin. Then execute this:
asterisk -vvvvvvrx 'show channels'
You should see:
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
That means Asterisk is turned off. That is what you want.
If you see other values here - that means you have some active calls going:
6 active channels
3 active calls
Wait some time and then execute same command again. Repeat it till you get correct message.
Then you can restart/shutdown your server.
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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.
App mor.so version
On command prompt issue:
asterisk -vvrx 'module show like mor'
You should get something like:
app_mor.so MOR Billing Solution v0.6 0
That means you have 0.6 version of app_mor.so
Ports which should be opened
For server some ports should be opened:
TCP 80 - for GUI/HTTP
TCP 22 - for SSH connection
UDP 4569 - for IAX2 connection
UDP 5060, 10000-20000 - for SIP connection
Asterisk under NAT
If your Asterisk box is under NAT, you should edit /etc/asterisk/sip.conf file:
[general]
externip = <YOUR REAL IP>
localnet = <LOCAL NET IP/MASK>
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Example:
[general]
externip = 23.45.67.89
localnet = 10.10.1.0/255.255.255.0
Forward SIP ports (UDP 5060, 10000-20000) on your firewall to this Asterisk machine.
Restart Asterisk.
Attention! Do not put these settings in wrong place! They should be in [general] section.
Useful links
VoIP Bandwidth Calculator
http://telconassociates.com/voip-bandwidth-calculator.html
http://www.voipcalculator.com/calculator/lipb/
http://www.asteriskguru.com/tools/bandwidth_calculator.php
Lost MySQL root password?
http://wiki.kolmisoft.com/index.php/Lost_MySQL_root_password%3F
Linksys SPA942 Configuration
http://wiki.kolmisoft.com/index.php/Linksys_SPA942_Configuration
Video call with CounterPath eyeBeam
http://wiki.kolmisoft.com/index.php/Video_call_with_CounterPath_eyeBeam
How to connect to Huawei devices
http://wiki.kolmisoft.com/index.php/How_to_connect_to_Huawei_devices
How to translate MOR to other language
http://wiki.kolmisoft.com/index.php/MOR_GUI_Translation
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Guides
PBX connection to MOR
Common connection diagram
Any IP/PSTN capable PBX can be connected to MOR:
To simplify diagram - we can show the common way how any PBX can be connected to MOR:
Here main principles will be explained. For detailed example please see Trixbox section.
In this diagram we see that we have 1 PBX connected to MOR Billing system. 2 phones (Phone 1 and Phone 2) are connected to PBX. Now both phones can dial out to the PSTN through PBX and MOR.
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Dialing from PBX
Our task is to bill each Phone separately.
In MOR we need to create 1 User/Device for PBX
We need to create 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 for 1 virtual device for each real device connected to PBX
In device's CLI menu we need to add CallerID's of these devices as described in PBX - using CallerID of device MOR recognizes who is calling and assigns call to correct device
Dialing to PBX
DID should be forwarded to PBX for real device to get call.
In order to assign DID to virtual device you need to do following steps:
1. Mark virtual device as Trunk
2. Assign DID to virtual device 3. Go to device's Call Flow 4. In Before Call state Forward call to PBX device
That way when call goes to virtual device through some DID it should go through PBX Trunk.
If you do not forward call to PBX device – call will fail.
And this let‟s user login to GUI and see all incoming calls to his device.
ATTENTION! In order for virtual device to receive call from the DID, PBX should route call to correct device. MOR does not know about real location/connection type of virtual device, it just sends call to PBX
and PBX should send call to correct device. MOR CAN'T influence PBX's configuration so PBX should be configured separately to handle incoming call from the DIDs.
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MOR and Trixbox
As MOR and Trixbox (FreePBX) both use Asterisk Realtime and custom extensions - they can't work nicely on the same server.
In order to integrate Trixbox with MOR you need to connect in following way:
Connection diagram
As an example we will configure one phone dialing. Phone is connected to Trixbox which is connected to MOR:
Here for simplicity all devices are in 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 following settings:
username: mor_trixbox
password: mor_trixbox_psw
And phone to Trixbox is connected with:
username: 201
password: 201psw
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Dialing from phone to MOR
Here device makes call to example destination 9999999999.
When call reaches Trixbox correct CallerID number for this phone is applied. In our example it is 1111111111.
Then call is forwarded to MOR.
When MOR gets a call it sees that call is sent by Trixbox.
Also it sees that Trixbox is Trunk with ANI - that means MOR needs to chech CallerID of the call to find real
caller.
MOR checks for CallerID number (1111111111) and sees that this CallerID is assigned to phone device.
The MOR retrieves all info necessary to send call further (LCR/Tariff/Rates/Providers/etc).
When call is finished - it is logged in MOR not for Trixbox, but for phone's user/device. That is what we wanted from our configuration.
Trixbox configuration for phone
Lets connect phone to Trixbox. In FreePBX create Extension for 201 phone:
Most important details:
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Here we set CallerID number which should be applied to the call from this device. Password and dial string also here.
Trixbox configuration for MOR
In Trixbox we need to define MOR as a Trunk:
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Define some route in Trixbox to use this Trunk (here is just an example):
TrixBox Trunk Troubleshooting
When passing multiple CallerIDs from multiple extensions in TrixBox through a single ANI-enabled trunk to MOR, make 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' CallerID in MOR to provide them with separate call cost tracking.
Also, be sure NOT to define the inbound USER CalleID on the same TrixBox trunk with the same empty entry. If you do, the CallerID will be stripped from all incoming calls
MOR configuration for Trixbox
We need to create connection between MOR and Trixbox. In this example we will configure MOR-Trixbox connection using SIP.
We need to create user for Trixbox - pbx_user, put any details for it - it does not matter:
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Now we need to configure device for Trixbox:
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 port is 5060 (for SIP)
MOR configuration for phone
To correctly bill phone calls you need to create separate user and device for it on MOR.
First let's create user for phone (phone_user):
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Make sure you configure this user correctly - e.g. enter password/LCR/Tariff and other necessary details - this user will be used for real dialing out.
Then we need to create virtual device for our phone:
All settings for this device are not important at all except:
1. Make sure you marked this device as Trunk - important for receiving calls 2. Make sure you have good password for security 3. Make sure this devices host = dynamic (for security reasons to do not let you entr Trixbox IP here, because
201 is not Trixbox!)
CallerID for phone in MOR
In order MOR to recognize device/phone we need to assign CalledID to our phone's device:
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NOTE: currently it is not possible to enter same CallerID as device's name. That way call will fail. It is due to the way Asterisk handles authentication.
Dialing to phone using DID
Here call is coming to DID: 2222222222
When it reaches MOR - MOR knows that 2222222222 should be forwarded to device 201
Device 201 forwards 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 call in MOR log call is in device's 201 logs (CDR) - as it should be
MOR configuration
You need to create DID 2222222222 in MOR and assign phones user/device to this DID:
And in this device's Call Flow you need to forward call to Trixbox:
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Trixbox configuration
Now when call arrives in Trixbox, it should know how to route call to phone 201.
We can configure this in 2 ways:
Using Direct DID
Using Inbound Route
Direct DID configuration
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Inbound Route configuration
Final notes
Here as an example we used DID 2222222222 and CallerID 1111111111. In most cases in real installations these
values are the same. That means - if user is dialing out and his CallerID (for example is) 3333333333 then he can receive calls by DID: 3333333333. You can configure this in same way using this manual. Different values for DID and CallerID here are chosen to make example more understandable.
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Configuring more phones
In order to configure more phones:
1. Connect phone to Trixbox 2. Configure user/device in MOR for phone 3. Make sure CallerID for phone in Trixbox is the same as CallerID (CID) for phone's device in MOR 4. Create DID for phone on MOR, assign it to phone's device, forward call from device to Trixbox in MOR 5. Make sure Trixbox knows how to route call from DID to phone
Limitations
Using this configuration to connect Trixbox (FreePBX actually) to MOR has some limitations:
1. MOR device's CallerID can't be the same as 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 actually making a call - so these devices with identical CallerIDs will be billed together using one account (user/device) in the MOR
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Troubleshooting Before starting to work you need to do Integrity Check. When some problem happens making outgoing call – first thing you should do is to use Call Tracing.
When you have troubles after Call Tracing or with non outgoing calls – you should login to Asterisk CLI and make a call. After call is made – you should look for some error message. Most common ones are described below. If you need technical assistance and support – KolmiSoft has different support plans which you could use. Just contact us.
Integrity Check This window test various properties of billing system and tells where potential problems can be.
It can be reached in SETTINGS - Billing - Functions - 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 and others. Such Destinations appear after tariff rate import. It is necessary to assign these Destinations to correct Destination Groups just after new rates are imported.
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Call Tracing This function is for checking the setup of MOR in order to be sure user will be able to dial out. To setup MOR is not so trivial (we are trying to change this with this manual and GUI improvements). It often happens for users to miss
some important detail trying to make everything work. Now Call Tracing will save a lot of head-scratching and we hope will help for a lot of beginners.
Important! This function is for testing Dialing OUT. DID testing is not
implemented yet. This function can be found in SETTINGS – Billing – Function – Call Tracing. It can be quickly accessed in Users
menu (SETTINGS – Users) also. Just press icon for some user.
In the first window we have to select which user we want to check:
User testing
Next window shows us did we configure everything properly for selected user. Let's look at each section separately. The first one shows info special for user:
Here we see 4 errors which does not let user to operate. Let's say we want to make user to make calls. So we will go through each one of errors and fix them.
1. This error tells us that user is Blocked and it will not be able to make calls. We should go to User settings
and unblock him.
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2. Balance and Credit for this user are empty. And user type is Prepaid. That means he has no funds to cover
the call fees. There're several ways how to go about this:
a. If user is postpaid – he should use some payment method to increase his balance. PayPal, Manual
Payment and other ways are good to do this.
b. We can change user type to postpaid and then we should adjust his credit. Maybe it's unlimited,
maybe some limit exists. But that will allow user to dial out.
3. We see that user has assigned some tariff and type of tariff. Type of tariff is just for informational purposes.
Here 'Explanation' tells us that tariff is empty. That is the reason why user will not be able to dial out. User
can dial out only to these destinations which have rates. Currently none of them has. You need to put some
rates to the existing tariff or assign non-empty tariff to this user.
4. User has no devices. Device is necessary to dial out. Just create one for this user.
After solving all these issues we should see:
We can proceed with next section which is about LCR:
It tells us that LCR we assigned to the user has no Providers. We should fix it. 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 1 active provider in order for user to be able to dial-out:
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Now none of our active providers has any rates. We should add rates to the provider's tariff. We can press directly on the Tariff to go to its rates and add some of them. After that we should see:
To summarize all this section you should check for and read error descriptions. After fixing them you should see
. That means 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 issues are solved 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 some rates for some destinations (so only these destinations are available to dial), he has some devices to use for dialing. Also his LCR has active providers with not empty tariffs.
You can proceed.
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Device testing
Following section let's us choose device, call date and enter destination which we want to test. For example:
This data means that we are testing virtual call from device IAX2/104 to number 863042438 on 2007 October 20th 00:00. When we press Proceed we will go to the next page and we will see something like:
Details section shows us just informational info. We knew this info before so it is not important.
If you see direction as Unknown that means your number is not in E.164 compatible format. Edit Localization rules to transform your number to E.164 number format. Localization section is much more important.
Here we see device's Location, Applied rule and result after localization.
IMPORTANT! Software can't read user's mind so it can't know where user is dialing. This can be illustrated by this example:
User dials 863042438. After Localization we see that number is still: 863042438. Which means it's somewhere in China. But it's wrong because user dialed Lithuanian number in local format! What does it mean? It means that Localization is not configured properly. MOR can't show you error here because there's no way to tell if it's ok or not. You should decide for yourself. What MOR can do – just to show you all info it can get from the call and it does that.
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So in order to proceed with example we have to set correct Localization rule for this number format. After doing this we will see:
Now everything is ok. Localized destination is E.164 compatible. Now MOR understands that it's Lithuania as it should be.
If we get following window:
That means we do not have rates or custom rates for dialed destination. After adding them we should see:
This window also shows us that we do not have custom rate for this destination. But it's enough to have simple rate for 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 rate for this destination. After fixing it we should see:
That's it. Now real call should go out without problems. To summarize Call Tracing for Device – 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. When all data are in place – there's no obstacle for call to be send to provider.
IMPORTANT! Green message at the end of Call Tracing does not mean call will be successful! It only means that MOR will send call to provider. What string it will send to provider you can see in last table. Applied Provider Rule parameters (Cut and Add) are also visible in the table. MOR can't know if provider accepts your dial string or not. That depends how you configured Provider Rules. Call Tracing shows you formatted number which is sent to provider but you should decide is it correct or not.
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Asterisk related errors By SIP error codes:
SIP/2.0 401 Unauthorized
Check your username/password
SIP/2.0 403 Forbidden
Same error: Received response: "Forbidden" from '"Unknown" If you get this message on your MOR CLI, then in device settings check Insecure Port and Invite fields.
If you get this message on your non-MOR CLI, for peer settings add: insecure = port,invite
SIP/2.0 404 Not Found
Sent from the proxy when either the intended party does not exist (wrong number dialed) or the user is not registered on that proxy. This message might be sent from the gateway if it's caused by dialing a wrong number.
This is also sent from the proxy in response to INVITE request if device is not registered to that proxy. You can tell if this is the case because the proxy does not authenticate the Invite request with a "407 Authentication Required".
Solutions
1. Make sure proxy/PBX has such number you are sending him 2. Make sure you send number in expected format (configured properly as your Provider expects) 3. Make sure you are registered to the proxy if it is necessary to do so (ask your Provider)
SIP/2.0 407 Proxy Authentication Required
The 407 Proxy Authentication Required response indicates that the phone must first authenticate itself with the proxy
server. If received by the phone, the phone may repeat the INVITE request with a suitable Proxy-Authorization field. This field should contain the authentication information of the user agent for the next outbound proxy or gateway.
Solutions
Make sure username/password are correct
Make sure you used username instead of user on far server (e.g. other Asterisk server) describing peer
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SIP/2.0 486 Busy Here
This response indicates that the callee was contacted but that their system is unable to take additional calls.
The SIP gateway generates this response when the called party is busy.
Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.
Solution
In Provider's details enter username into field Fromuser.
SIP/2.0 503 Service Unavailable
Got SIP response 503 "Service Unavailable." back from 23.45.67.89 Check codecs for your server and your provider. Often there are no matching and call is not possible.
SIP/2.0 603 Declined
This response indicates that the callee was contacted but cannot or does not want to participate in the call.
The SIP IP phone can generate this response if the user is using call screening. Upon receiving this response, the phone notifies the user with fast-busy signal and disconnects the call.
The SIP gateway does not generate this response. Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call.
Solution
In Provider's details enter provider's hostname or ip into field Fromdomain.
In field fromuser enter username
Check Trustrpid
Check both settings for Insecure
Call rejected by IP: No authority found
Call gets rejected
The 'No authority found' error is generated when an iax entry [example] matching the username sent can not be found. It will also occur if the secret is wrong.
Check password of your device
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CallerID not passed
Problem
CallerID is empty or wrong on receiving end from MOR
Troubleshooting
Make sure there're empty fields in Device's settings window CallerID section (Name/Number)
TrixBox Trunk Troubleshooting
When passing multiple CallerIDs from multiple extensions in TrixBox through a single ANI-enabled trunk to MOR, make 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' CallerID in MOR to provide them with separate call cost tracking.
Also, be sure NOT to define the inbound USER CalleID on the same TrixBox trunk with the same empty entry. If you do, the CallerID will be stripped from all incoming calls
DID is not working
Do you need to register to your Provider in order to use DID from him? (Usual answer is YES). Did you registered to your provider?
Check Asterisk CLI for error message.
If you can't see any output on Asterisk CLI when making call to DID, use sip debug or iax2 set debug commands
Check for error message
's' extension
If 's' extension is reached means your provider does not send destination information.
Usually you will hear "SOMETHING IS TERRIBLY WRONG" when call goes to s extension.
Solution:
1. In register string add /SOME_DID at the end 2. Create SOME_DID in DIDs window 3. Assign this SOME_DID to some device
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Example:
register => username:[email protected]/1234567890
Now every call which comes from sip.provider.com will go to DID 1234567890.
We need to create such DID and assign some device to it to receive all calls coming from this provider.
This solution is workaround when provider does not send correct destination information (most of providers do this).
One bad thing about this is that only 1 device can receive calls from this provider.
In order to receive calls to several DIDs->Devices you need to tell your provider to send correct destination information.
Fax is not received
After trying to use Fax2Email functionality in Asterisk CLI you see:
Check permissions on /var/spool/asterisk/faxes
mor_fax2email: MOR Fax2Email v0.1.3 AGI script started. mor_fax2email: Failed to connect database server. Check debug for more info. mor_fax2email: ERROR! Not connected to database.
Solution:
In /var/lib/asterisk/agi-bin/mor.conf correct DB connection settings.
MOR can't determine who is calling. Make sure accountcode is set for caller
(Provider or Device)
Provider/Device is not authenticated.
Check settings for Provider/Device
For provider - make sure hostname and IP address have correct values. o Both these fields should be filled o If hostname is not assigned, then it should have IP address - same value as IP address field
Check port setting.
o Maybe call is coming not through default SIP port 5060, but from 5061? Check it! o Port is SOURCE port, that means FROM which port call is coming, NOT TO which port is coming.
If you are using H323 in file h323.conf add UserByAlias=no
If you are using ZAP/H323 provider - you could forget about accountcode=X setting.
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Retrieved 0 adv.rates, max adv.rate: 0.000000, total event price: 0.000000
That means no Advanced Rates found for that destination and calling is impossible.
Use Call Tracing (read about it in the manual) to troubleshoot this problem.
One-way audio or not audio at all
Make sure NAT settings are correct
Make sure you are using same RTP range as your provider
While connecting with H.323 client you get no audio or garbled audio and messages like this on Asterisk
console: rtp.c:489 ast_rtp_read: Unknown RTP codec 107 received. Try to disable Speex or some other codec on Asterisk or/and client side.
Make canreinvite = no in device settings
No sound on Voicemail or Playback
Calling works, MOH works, but VM and Playbac - does not.
Make sure Zaptel is configured properly. If you are not using Zaptel - remove it from loading.
Dropping extra frame of G.729 since we already have a VAD frame at the end
VAD = Voice Activity Detection or Silence Suppression
Asterisk does not support VAD. It should be disabled on other (client's) end to do not receive such messages.
These messages are harmless.
Same as: Comfort noise support incomplete in Asterisk (RFC 3389)
Comfort noise support incomplete in Asterisk (RFC 3389)
Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 23.45.67.89
Same as: Dropping extra frame of G.729 since we already have a VAD frame at the end
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mysql_reconnect: mor: Unknown connection error: (2006) MySQL server has gone away.
This error message appears when connection between MySQL database and MOR application app_mor.so timeouts. Default timeout is 8h. Timeout happens when in 8h there are no requests to MySQL database.
app_mor.so automatically reconnects to MySQL database so there's no need to take any additional actions.
Call dropped upon connection
Looks like this:
Executing [106@mor:1] Dial("SIP/dlink-081bb860", "SIP/spa1|60|Ttr") in new stack
-- Called spa1
-- SIP/spa1-081f6b98 is ringing
-- SIP/spa1-081f6b98 answered SIP/dlink-081bb860
== Spawn extension (mor, 106, 1) exited non-zero on 'SIP/dlink-081bb860'
Make sure you are using same codecs on server and your devices.
Starting asterisk: Cannot find your TTY
The fix for this is simple: use your favourite editor, and edit /usr/sbin/safe_asterisk, comment out the TTY line, and viola, you should now be able to automatically start, restart, or stop asterisk from the init script until your hearts content.
Naturally, as always, backup files that you edit. You may find yourself stuck later.
http://forums.whirlpool.net.au/forum-replies-archive.cfm/607081.html
Same error, different message:
/usr/sbin/safe_asterisk: line 123: /dev/tty9: Permission denied
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False ANSWER
If you are using GREY routes (not legal) be ready to experience false Answer.
That means that i some way between caller and callee some "grey" provider answers the call but still plays ringing sounds to fake the call.
The call will be marked as ANSWERED but in reality nobody will answer.
To avoid this - talk with your provider or use "white" routes.
Betamax
This is very often happens with Betamax company group providers: http://backsla.sh/betamax
Voice Trading
http://www.voicetrading.com/
This provider supports 3 quality routes:
Dialing number in E164 format: GREY
Prefix 00: STANDART
Prefix 000: PREMIUM
GREY plan has FALSE ANSWER problem.
Unable to allocate RTCP socket: Too many open files in system
[Apr 3 00:09:52] WARNING[15997]: rtp.c:1863 ast_rtcp_new: Unable to allocate RTCP
socket: Too many open files in system
[Apr 3 00:09:52] WARNING[15997]: udptl.c:782 ast_udptl_new_with_bindaddr: Unable to
allocate socket: Too many open files in system
[Apr 3 00:09:52] WARNING[15997]: channel.c:778 ast_channel_alloc: Channel allocation
failed: Can't create alert pipe!
[Apr 3 00:09:52] WARNING[15997]: chan_sip.c:3966 sip_new: Unable to allocate AST channel
structure for SIP channel
[Apr 3 00:09:52] WARNING[15997]: app_dial.c:1191 dial_exec_full: Unable to create
channel of type 'SIP' (cause 0 - Unknown)
Under linux, by executing
cat /proc/sys/fs/file-nr
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you'll get 3 numbers on one line, which mean:
(max number of files simultaneously open since boot)
(how many files are open right now)
(max number of files allowed to be simultaneously open)
and you can change the last number (the max limit) by doing e.g.
echo 8192 > /proc/sys/fs/file-max
which can be done automatically on every boot by installing that command in an initialization file like /etc/rc.d/rc.local
http://www.dbforums.com/archive/index.php/t-341298.html
http://confluence.atlassian.com/display/DOC/Fix+'Too+many+open+files'+error+on+Linux+by+increasing+filehandles
Echo
This is too big to discuss all possible causes here. Please follow link below.
http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/EA_ISD.html
Asterisk stops working with Internet loss
If Internet is gone an Asterisk is unable to serve local/zap calls, then make sure it does not try to register to external providers.
Comment out register lines in sip.conf/iax.conf. Uncomment them when internet comes alive.
Can't register device
Make sure settings are correct (server IP/username/password)
Make sure UDP ports 4569, 5060, 10000-20000 are open.
Check Asterisk CLI when trying to register with command sip debug or iax2 debug.
o If no output at all - ports are closed
o If some output - follow error messages, e.g. SIP 2.0/404 Not found or similar.
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Registration from 'SOME DEVICE' failed for 'IP' - Not a local domain
Do not use domain setting in sip.conf file.
Unable to find key 'XXX' in family 'SIP/Registry'
Make sure you have rtcachefriends = no in sip.conf
In MOR GUI in device's settings make sure nat = yes
Registration from 'sip:[email protected]' failed for '34.56.67.89' - No matching
peer found
Check username/password for device
Device can't dial out
Use Call Tracing if you are trying to dial to PSTN
Check callee settings if you are dialing locally
Make sure you have selected correct codecs for device!
No path to translate from XXX to YYY
Make sure you have correct codecs enabled between devices and providers.
Call rejected by IP: No such context/extension
Device/Extension is not configured in client (Asterisk server)
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Mor authorize: Too low balance for more simultaneus calls!!!
The reason can be that your user is PREPAID and he has not enough balance to cover more simultaneous calls. MOR uses algorithm to reserve balance when call is made to be safe if user is prepaid. That way when second, third, etc calls are made - MOR checks free - not reserved balance and decides should it allow to dial or not.
To lower security level and allow users with low balance to dial out several times, go to /etc/asterisk/mor.conf and change to:
min_frozen = 1
That way you are taking risk that you will take loss if user talks for more then frozen_time (default 90s). E.g. if user make a call, his balance is frozen for 90s with the rate to current destination. If user talks more - you can take loss.
Device can't be reached
Check nat option for device. Enable or disable it upon situation.
Check port info - adjust it depending on protocol. Default: SIP: 5060, IAX2: 4569
Check codecs - enable necessary codecs
Unable to create channel of type 'XXX' (cause 3 - No route to destination)
In iax2.conf set
rtignoreregexpire = yes
and in sip.conf set
ignoreregexpire = yes
Reload Asterisk
Device is not receiving video
Make sure you have videosupport=yes in device's settings
Make sure you selected appropriate video codecs in device's settings
Make sure in sip.conf you have:
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[general]
videosupport=yes
More info
Zaptel YELLOW alarm
When doing zap show status we get:
Description Alarms IRQ bpviol CRC4
Wildcard TE12xP Card 0 YEL 53 0 0
Possible solutions
http://www.asteriskguru.com/tutorials/yellow_alarm.html
The problem got solved when the PRI line was put down and restarted by the service provider, at their end.
Unable to open master device '/dev/zap/ctl'
Often happens just after Zaptel installation. Rebooting the server solves it.
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GUI problems
We're sorry, but something went wrong
We're sorry, but something went wrong.
We've been notified about this issue and we'll take a look at it shortly.
Troubleshooting
Check for error msg when trying to open GUI
Check GUI log just after window fails to open.
Check GUI log
When you get error on the GUI:
We're sorry, but something went wrong.
We've been notified about this issue and we'll take a look at it shortly.
or something else is not working (e.g Graphs), you need to check RoR log. It has info what is wrong. . It can be done accessing file /home/mor/log/production.log.
Starting from MOR v0.5.0.5 it is possible to do that online using address http://<YOUR IP>/billing/debug/show_log.php.
It shows last 100 lines (usually it is enough) to find the cause of the problem.
Errors
You can get:
Fatal error: Allowed memory size of 16777216 bytes exhausted (tried to allocate 48372783
bytes) in /home/mor/public/debug/show_log.php on line 8
That means file is too huge. Login to server over SSH and do:
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cp /home/mor/log/test.log /home/mor/log/production.log
Forbidden You don't have permission to access /mor/ on this server.
Possible reason: Apache is misconfigured and does not describe MOR page.
LoadError (Expected /home/mor/app/controllers/callc_controller.rb to define
CallcController)
Gem install pdf-writer
Debian
apt-get install libopenssl-ruby
/etc/init.d/apache2 restart
Centos
In file /home/mor/controllers/callc_controller.rb
Comment line:
#require 'smtp_tls'
Restart Apache. If MOR GUI starts to work - that means your ruby does not support SMTP. Either ruby should be recompiled or appropriate package should be installed.
LoadError (Expected /home/mor/app/controllers/stats_controller.rb to define StatsController)
Execute:
gem install pdf-writer -y --no-rdoc --no-r
rrno::EACCES (Permission denied - /tmp/mor_debug.txt)
Login to bash and execute:
chmod 777 tmp
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NoMethodError (undefined method `in_place_edit_for' for CallcController:Class)
To fix this you need to execute: gem install rails -v=1.2.6
then go to /home/mor/config/environment.rb
and add line:
RAILS_GEM_VERSION = '1.2.6' save and restart apache – it should work. MOR code will be fixed to be Rails 2.0 compatible.
FCGI info is shown in text format
If you see something like this:
#!/usr/bin/ruby
#
# You may specify the path to the FastCGI crash log (a log of unhandled
# exceptions which forced the FastCGI instance to exit, great for debugging)
# and the number of requests to process before running garbage collection.
#
# By default, the FastCGI crash log is RAILS_ROOT/log/fastcgi.crash.log
# and the GC period is nil (turned off). A reasonable number of requests
# could range from 10-100 depending on the memory footprint of your app.
#
# Example:
# # Default log path, normal GC behavior.
# RailsFCGIHandler.process!
#
# # Default log path, 50 requests between GC.
# RailsFCGIHandler.process! nil, 50
#
# # Custom log path, normal GC behavior.
# RailsFCGIHandler.process! '/var/log/myapp_fcgi_crash.log'
#
require File.dirname(__FILE__) + "/../config/environment"
require 'fcgi_handler'
RailsFCGIHandler.process!
That means mod_fcgid is not installed for Apache.
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Errno::EACCES (Permission denied - /tmp/mor_cf_xxxx or /var/spool/asterisk/outgoing/mor_cf_xxxx)
Bad permissions on /var/spool/asterisk/outgoing/
Fix:
chmod 777 /var/spool/asterisk/outgoing
ActiveRecord::RecordNotFound (Couldn't find User with ID=0)
Execute to DB:
UPDATE users SET id = 0 WHERE username = 'admin';
Interlaced PNG images are not currently supported
This happens when generating PDF Invoice. Make sure your PNG image is not interlaced and does not have transparency. Also you can save your logo as JPG.
No route matches "/" with {:method=>:get}
Reason: MOR GUI files are not copied to /home/mor
Solution:
cp -r /usr/src/mor/gui/* /home/mor/
uninitialized constant Gem::GemRunner
In file /usr/bin/gem add line:
require 'rubygems/gem_runner'
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Folder content is shown
Disable SELinux Also make sure apache configuration file describes mor folder and AllowOverwrite=All where necessary.
No such file or directory - /tmp/mysql.sock
Make symlink from original mysql.sock (found in /etc/mysql/my.cnf or /etc/my.cnf) to /tmp/mysql.sock ln -s /var/lib/mysql/mysql.sock /tmp/mysql.sock
NoMethodError (undefined method `exchange_rate' for nil:NilClass)
Make sure you have all currencies for tariffs. Example: if you have tariff in USD, but do not have USD in Currencies window - you will get this error. Create USD in
Currencies window.
TypeError (PNG alpha channel transparency is not supported; only palette
transparency is supported.)
Save your logo without transparency.
ActiveRecord::StatementInvalid (Mysql::Error: Got a packet bigger than 'max_allowed_packet' bytes:
In /etc/mysql/my.cnf enter bigger value for max_allowed_packet e.g.
max_allowed_packet = 64Mb
Reference: http://dev.mysql.com/doc/refman/5.0/en/packet-too-large.html
Do not forget to restart MySQL service:
/etc/init.d/mysqld restart
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Can't open phpMyAdmin
It can be reached in: http://<IP>/phpmyadmin
Errors
The configuration file now needs a secret passphrase (blowfish_secret)
Login to your server over SSH
If you have Centos/Fedora go to:
/var/www/html/phpmyadmin
If you have Debian go to:
/var/www/phpmyadmin
Open file config.inc.php, find line:
$cfg['blowfish_secret'] = ; /* YOU MUST FILL IN THIS FOR COOKIE AUTH! */
And enter any characters between single commas. Example:
$cfg['blowfish_secret'] = 'mun9b8y7tv6cr5ex4'; /* YOU MUST FILL IN THIS FOR COOKIE AUTH!
*/
Save the file, go to webrowser, press CTRL+F5 and access phpMyAdmin.
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Email problems This error can appear when trying to send email from Email section or when user registers: Try with Default settings
Default settings
SMTP Server: smtp.gmail.com
Domain: localhost.localdomain
Login: kolmitest
Password: kolmitest99
Batch Size: 50
Net::SMTPAuthenticationError (535 auth failure) or
Net::SMTPAuthenticationError (535 invalid user ID or password)
On both of these errors:
1. Check your SMTP server details
2. Try to use another SMTP server
3. Try default settings:
NameError (undefined local variable or method `xxxxxx' for #)
Do not add variables which are not from this list:
<%= server_ip %>
<%= device_type %>
<%= device_username %>
<%= device_password %>
<%= login_url %>
<%= login_username %>
<%= login_password %>
<%= username %>
<%= first_name %>
<%= last_name %>
<%= full_name %>
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Timeout::Error (execution expired) /usr/lib/ruby/1.8/timeout.rb:54:in `open'
Open port 25 to allow SMTP connection.
SocketError (getaddrinfo: Name or service not known)
Execute (on Debian):
apt-get install mime-construct
mime-construct --to "[email protected]" --subject 'hi there' --string "body"
Go to /var/log/exim4/mainlog
If you see:
2008-04-14 08:35:06 1JlOMs-0001fU-I4 <= [email protected] U=root P=local S=404
2008-04-14 08:35:12 1JlOMs-0001fU-I4 ** [email protected] R=dnslookup T=remote_smtp: SMTP
error from remote mail server after end of data: host gmail-smtp-in.l.google.com
[64.233.185.27]: 550-5.7.1 [190.37.134
.234] The IP you're using to send mail is not authorized\n550-5.7.1 to send email
directly to our servers. Please use the SMTP\n550-5.7.1 relay at your service provider
instead. Learn more at\n550 5.7.1 http:
//mail.google.com/support/bin/answer.py?answer=10336 7si13749186wrl.12
That means you should use your providers SMTP server to send emails.
501 Syntactically invalid HELO argument(s)
Domain is empty. Enter: localhost.localdomain
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Can't initiate web callback
Errno::EHOSTUNREACH (No route to host - connect(2))
/lib/rami.rb:380:in `initialize'
/lib/rami.rb:380:in `new'
You can't access Asterisk AMI interface.
Check values AMI_Host, AMI_Username, AMI_Secret
Check /etc/asterisk/manager.conf
o enabled=yes o settings in [mor] section
Execute
telnet <host IP> 5038
from GUI server, you should see:
Asterisk Call Manager/1.0
Check firewall settings, port 5038 (TCP) should be open on Asterisk server
Installation errors
Rails: command not found
$PATH can't find rails
It is usually under /var/lib/gems/1.8/bin
Do:
cd /home
/var/lib/gems/1.8/bin/rails mor
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Yaml.rb:133: BUG Segmentation fault
/usr/lib/ruby/1.8/yaml.rb:133: [BUG] Segmentation fault
This means that your server does not have enough memory (RAM) for installation (GEM) to work.
Solution: Increase RAM on your server or change server.
This often happens on cheap VPS (Virtual Private Servers).
The problem is that ruby's decompresser uses over 100MB of RAM when extracting the rails and other gems. Since VPS servers have limited RAM and no virtual memory, the process fails.
Use top command to check it. If you have something like:
Mem: 131072k total, 131072k used, 0k free, 0k buffers
Swap: 0k total, 0k used, 0k free, 0k cached
That means real trouble.
Use cat /proc/user_beancounters command. If you have something like (note failcnt at privvmpages):
uid resource held maxheld barrier limit failcnt
privvmpages 36796 65564 65536 81920 36
The failures shown in privvmpages is most likely the reason for the segmentation fault you've encountered.
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Various
Asterisk crashing
If you suspect asterisk crashes because of mor, go to /var/lib/asterisk/modules move app_mor.so to safe place and
restart Asterisk. If it still crashes - it is not problem of MOR. If it stopped crashing - report a problem to [email protected] Asterisk 1.4.11 sometimes crashes when SIP incoming call comes. None other version has this strange bug. Upgrade it to newest one if you experience this strange behavior.
THANK YOU FOR USING MOR!
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2008 05 16 Vilnius Lithuania KolmiSoft
