Kolmisoft Wiki - User contributions [en]

Reseller Addon

2016-03-18T08:39:54Z

Rita:

==Description==

This addon incorporates basic reseller support into MOR

Currently it lets to create user with type 'reseller'

When this user logs into the system he is able to:

# Create tariffs
# Create users
# Setup their own tariffs and assign them to their users
# View calls and other stats for his own users
# Add own logo to GUI
# Customize invoice with their details (logo is not supported)
# Can calculate own profit

When his users are making calls - reseller's balance decreases along with user's balance. If reseller is prepaid and he is out of balance or postpaid and credit limit is reached - his users will not be able to make calls even if they have money in their balance.

===DIDs===

DIDs can be created only by admin. Reseller can assign DIDs available to him to his users.

ADMIN can assign DID to USER directly bypassing RESELLER.
 

=== Calling Cards ===

[[Questions_about_Calling_Cards#Q._Can_Resellers_have_own_Calling_Cards.3F | Can Resellers have own Calling Cards?]]

 
=== Complete Calls ===

To get all Calls in selected period for Reseller and his User use following link:

[[Image:reseller_calls_complete.png]]

 

===How to create a Reseller?===

Got to '''SETTINGS –> Users'''
 
[[File:Users_path.png]]
 
Press the [[image:add.png]] icon to add new User:
 
[[File:Users_list.png]]
 
Select the "reseller" user type from the marked drop down box:
 
[[File:Users_change_usertype.png]]
 

==Reseller billing==

Example:

You (admin) pay your provider 0.3$ per minute;

Reseller pays you 1.2$ per minute;

Reseller's client pays reseller 2$ per minute.

Your profit is 0.9$;

Reseller's profit is 0.8$.

If reseller uses his own provider then he does not pay you.

NOTE: only [[Reseller_Pro_Addon|Reseller PRO]] can have his own providers.
 

=Manual for Resellers=

This manual is dedicated for resellers to work on MOR

* [[Manual for Resellers]]
 

=Web access=

Each Reseller has it's own Homepage. URL can be found in '''MAIN MENU -> Settings''' -> '''Global''' tab.

It is possible to have different domain name for each Reseller. Domain name should be pointed to Reseller's URL in such case.

NOTE: Kolmisoft does not provide domain names, so you need to check with certain domain provider for such option.

 

=See also=
* [[Reseller Pro Addon]]
* [[Reseller Permissions]]
* [[Reseller Settings]]
* [[MOR Addons]]
* [[How to show rates for resellers users]]
* [[Multi-level Reseller support]]

Limit the amount of calls per route

2016-03-01T06:59:35Z

Rita:

If you want to limit the amount of calls sent to each route for a User, the User has to send the calls to each route with a prefix. This way you can configure different providers for each prefix and add a [[Simultaneous_call_limitation#Provider | call limit]] to each provider. Use localization rules to route calls with prefix to a certain provider.
 
'''Example:''' The User uses two routes: ''Provider_premium'' and ''Provider_standard'', and you want to allow only 2 simultaneous calls to the ''Provider_premium'' and 5 to the ''Provider_standard''.
 
'''Configuration:'''
 
1) Create two new providers: ''Provider_premium'' and ''Provider_standard''.
 
2) Add call limit to each provider: 2 for ''Provider_premium'', and 5 for ''Provider_standard''.
 
3) Create two new LCRs: ''LCR_Premium'' and ''LCR_Standard''. Add ''Provider_premium'' to ''LCR_Premium'', and ''Provider_standard'' to ''LCR_Standard''.
 
4) Ask User to send calls to ''Provider_premium'' with prefix 1111, and to ''Provider_standard'' with 2222.
 
5) Create a new Location with these Localization rules:
Name: Premium, Cut: 1111, Add: , Change User LCR to this LCR: LCR_Premium
Name: Standard, Cut: 2222, Add: , Change User LCR to this LCR: LCR_Standard

6) Assign Location to User's devices.
 
=See also=
* [[Simultaneous call limitation]]
* [[Localization]]

Limit the amount of calls per route

2016-03-01T06:54:45Z

Rita:

Limit the amount of calls per route

2016-03-01T06:54:08Z

Rita: Created page with 'If you want to limit the amount of calls sent to each route for a User, the User has to sent the calls to each route with a prefix. This way you can configure different providers…'

If you want to limit the amount of calls sent to each route for a User, the User has to sent the calls to each route with a prefix. This way you can configure different providers for each prefix and add a [[Simultaneous_call_limitation#Provider | call limit]] to each provider. Use localization rules to route calls with prefix to a certain provider.
 
'''Example:''' The User uses two routes: ''Provider_premium'' and ''Provider_standard'', and you want to allow only 2 simultaneous calls to the ''Provider_premium'' and 5 to the ''Provider_standard''.
 
'''Configuration:'''
 
1) Create two new providers: ''Provider_premium'' and ''Provider_standard''.
 
2) Add call limit to each provider: 2 for ''Provider_premium'', and 5 for ''Provider_standard''.
 
3) Create two new LCRs: ''LCR_Premium'' and ''LCR_Standard''. Add ''Provider_premium'' to ''LCR_Premium'', and ''Provider_standard'' to ''LCR_Standard''.
 
4) Ask User to send calls to ''Provider_premium'' with prefix 1111, and to ''Provider_standard'' with 2222.
 
5) Create a new Location with these Localization rules:
Name: Premium, Cut: 1111, Add: , Change User LCR to this LCR: LCR_Premium
Name: Standard, Cut: 2222, Add: , Change User LCR to this LCR: LCR_Standard

6) Assign Location to User's devices.

MOR API rate get

2016-02-29T09:17:42Z

Rita: /* Parametres */

=Description=
[[MOR API]] rate_get feature enables you to check rate over http.
 

Admin able to check Rates of his Users.
Accountant with Permissions able to check Rates of Admins Users.
Reseller able to check Rates of his Users.
 
Note if Admin has checked Allow Resellers to use Admin Tariffs in [[API Configuration]] then using this API Reseller can get Rates which belongs to Admin.

=Configuration=

Go to '''SETTINGS –> Setup –> Settings'''
 
[[File:settings_path.png]]
 
In API tab ensure that MOR API is enabled and check '''Allow devices to check rate over HTTP''' option :
 
[[File:Settings_api.png]]
 

=Usage=

* From '''MOR X4''' call: '''/api/rate_get'''
* For Backwards-compatibility old name 'rate' is also usable. Call: '''/api/rate'''
* Methods: POST, GET(if allowed, not recomended)
 
===Parametres===

* u - username for authentication. '''From X5 - Required'''.
* username – username of user whose rates you want to see.
* prefix – prefix number of destination you want to see.
* [[MOR API hash construction | hash]] - SHA1 hash constructed using '''username''' and '''API_Secret_Key''' ([[MOR API hash construction | More described in Constructing hash]]). '''Required''' (if setting '''Disable hash checking''' is not checked).
'''NOTE:''' Don't use full number in API request. Use correct prefix only, because this method don't do prefix search, its only shows actual price for the exact prefix.
 

===Request===

HTML POST: http://<SERVER_IP>/billing/api/rate_get?u=admin&username=USERNAME&prefix=PREFIX&hash=HASH

Example:

http://127.0.0.1/billing/api/rate?u=admin&username=admin&prefix=370&hash=47fc73b6d374bd0a85e3984c1e1623451681569c
 

===Returns===

'''Success'''

Result is returned in format: rate#destination#prefix

<?xml version="1.0" encoding="UTF-8"?>
<page>
<rate>0.987#Lithuania#370</rate>
</page>

'''NOTE.''' MOR API rate method shows rate, which is applied to User. If rate by prefix is not found in [[Custom Rates]], it will be taken from Tariff assigned to User.
 
----
'''Errors'''

* User do not have rates for selected destination:

<?xml version="1.0" encoding="UTF-8"?>
<page>
<error>Rate was not found</error>
</page>

* User with selected username does not exist:

<?xml version="1.0" encoding="UTF-8"?>
<page>
<error>User was not found</error>
</page>

* '''Allow devices to check rate over HTTP''' option in [[MOR_API_rate_get#Configuration | API Settings]] is not set:

<?xml version="1.0" encoding="UTF-8"?>
<page>
<error>Feature disabled</error>
</page>

* API is not allowed:

<?xml version="1.0" encoding="UTF-8"?>
<status>
<error>API Requests are disabled</error>
</status>

* User does not belong to the user who is trying to check rate or it was not found in the system:

<?xml version="1.0" encoding="UTF-8"?>
<status>
<error>User was not found</error>
</status>

* Accountant must have See Financial Data read or write [[Accountant_permissions | permission]] and Manage Tariffs read or write permission enabled:

<?xml version="1.0" encoding="UTF-8"?>
<status>
<error>You are not authorized to view this page</error>
</status>

 

=See Also=

* [[MOR API]]

Other

2015-12-28T12:34:11Z

Rita:

[[What is SSH?]]

[[yum]]

[[TOP usage|top - monitor resources used by applications]]

[[M2 Call Trace Does Not Work]]

CDR Rerating

2015-10-23T09:32:00Z

Rita: /* Tariff from Localization */

= Description =

This function is used when you change Tariff 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. It affects balance of User and statistics will show information with new Prices.
 
=Usage=

Go to '''SETTINGS –> Billing –> CDR –> Rerating''':
 
[[Image:cdrrerating_image1.png]]
 
Select date interval and User whose Calls you want to rerate. You can select All Users and include calls of Reseller Users.



Also you are able su set '''Ghost Time''' percent, this value will increase user_billsec and/or reseller_billsec and price which depends on changed user_billsec and/or reseller_billsec. Ghost Time will be calculated just from initial billsec, and if you will rerate two timesr with same percentage value, on second time nothing will change. 0 leaves or returns initial user_billsec and/or reseller price.
 
Table below will show which data of Calls will be affected according each selection scenario.
 
{| class="wikitable" border="1" style="border-collapse: separate; border-spacing: 0; border-width: 1px; border-style: solid; border-color: #C0C0C0; padding: 0"
|-
!
! One of Admin Users
! One of Admin Resellers
!                 All Users                
! One of particular Reseller Users
! All Users (Reseller Users Included)
|-
| user_billsec,
user_rate,
user_price
|                 +
|                 -
|                 +
   (just for Admin Users Calls)
|                 +
|                 +
|-
| reseller_billsec,
reseller_rate,
reseller_price
|                 -
|                 +
 (for Calls of All Reseller Users)
|                 +
|                 -
|                 +
       (just for Reseller User Calls)
|}
'''NOTE''' If in Call of Reseller User will be used one of [[Common Use Providers]] Tariff to rerate Reseller Price will be taken form ADDONS -> Resellers -> Common use Providers. If in Call of Reseller User will be used one of Provider which belongs just to Reseller, Provider Price will be rerated by that Provider tariff.
 
* Press '''Rerate in Background(FAST)''' if you want to send Rerating task to [[Background Tasks]]. This way rerating of huge amount of Calls goes faster than SLOW method and also you do not need to wait in current page till rerating will be finished. '''Also Provider Billing details will be rerated'''

* Press '''Rerate in GUI(SLOW)'''. 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'''. '''Provider Billing details will NOT be rerated.''' If you choose this way for Rerating you will be redirected to second step:
 
[[Image:cdrrerating2.png]]
 
Here you see call stats for selected date interval. When you are sure everything is correct – press '''Rerate''':
 
[[Image:cdrrerating3.png]]
 
Here you will see the Difference after Rerating.
 
== Rerate all users ==


Select 'All Users' from User selection to rerate all Users at once in selected period.
 
== Testing mode ==

Select '''Tariff for User''' below '''TESTING MODE''' header:
 
[[Image:cdrtestrerating.png]]
 
...and press '''Test with selected Tariff (SLOW)''' to check differences of Rates for selected User. This way Rerating will not save changes to Calls/Users.

Testing mode is useful for testing "What if User had this Tariff" scenarios.
 
'''Selected Tariff is used just in [[Image:icon_test.png]]Testing mode.''' Rerating functionality rerates old Calls by Tariff set in User Settings, if you did not change Tariff for User, after rerating Calls Prices will be same as were before.
 
== Tariff from Localization ==

If call's prefix is in Localization Rule which has some Tariff, this Tariff is used to count call's price instead of User's Tariff.

 

=Other notes=

* Re-rate does not take into account Flat Rates and User will be charged for such Calls. Do not do rerating for such User.

* Re-rate does not work when call's DID, LCR or Owner Device is changed by Localization rules.

* Re-rate will take Active Rate when Rates with [[Rate Effective From | Effective From]] are re-rated.

CDR Rerating

2015-10-15T09:56:17Z

Rita: /* Tariff from Localization */

= Description =

This function is used when you change Tariff 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. It affects balance of User and statistics will show information with new Prices.
 
=Usage=

Go to '''SETTINGS –> Billing –> CDR –> Rerating''':
 
[[Image:cdrrerating_image1.png]]
 
Select date interval and User whose Calls you want to rerate. You can select All Users and include calls of Reseller Users.



Also you are able su set '''Ghost Time''' percent, this value will increase user_billsec and/or reseller_billsec and price which depends on changed user_billsec and/or reseller_billsec. Ghost Time will be calculated just from initial billsec, and if you will rerate two timesr with same percentage value, on second time nothing will change. 0 leaves or returns initial user_billsec and/or reseller price.
 
Table below will show which data of Calls will be affected according each selection scenario.
 
{| class="wikitable" border="1" style="border-collapse: separate; border-spacing: 0; border-width: 1px; border-style: solid; border-color: #C0C0C0; padding: 0"
|-
!
! One of Admin Users
! One of Admin Resellers
!                 All Users                
! One of particular Reseller Users
! All Users (Reseller Users Included)
|-
| user_billsec,
user_rate,
user_price
|                 +
|                 -
|                 +
   (just for Admin Users Calls)
|                 +
|                 +
|-
| reseller_billsec,
reseller_rate,
reseller_price
|                 -
|                 +
 (for Calls of All Reseller Users)
|                 +
|                 -
|                 +
       (just for Reseller User Calls)
|}
'''NOTE''' If in Call of Reseller User will be used one of [[Common Use Providers]] Tariff to rerate Reseller Price will be taken form ADDONS -> Resellers -> Common use Providers. If in Call of Reseller User will be used one of Provider which belongs just to Reseller, Provider Price will be rerated by that Provider tariff.
 
* Press '''Rerate in Background(FAST)''' if you want to send Rerating task to [[Background Tasks]]. This way rerating of huge amount of Calls goes faster than SLOW method and also you do not need to wait in current page till rerating will be finished. '''Also Provider Billing details will be rerated'''

* Press '''Rerate in GUI(SLOW)'''. 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'''. '''Provider Billing details will NOT be rerated.''' If you choose this way for Rerating you will be redirected to second step:
 
[[Image:cdrrerating2.png]]
 
Here you see call stats for selected date interval. When you are sure everything is correct – press '''Rerate''':
 
[[Image:cdrrerating3.png]]
 
Here you will see the Difference after Rerating.
 
== Rerate all users ==


Select 'All Users' from User selection to rerate all Users at once in selected period.
 
== Testing mode ==

Select '''Tariff for User''' below '''TESTING MODE''' header:
 
[[Image:cdrtestrerating.png]]
 
...and press '''Test with selected Tariff (SLOW)''' to check differences of Rates for selected User. This way Rerating will not save changes to Calls/Users.

Testing mode is useful for testing "What if User had this Tariff" scenarios.
 
'''Selected Tariff is used just in [[Image:icon_test.png]]Testing mode.''' Rerating functionality rerates old Calls by Tariff set in User Settings, if you did not change Tariff for User, after rerating Calls Prices will be same as were before.
 
== Tariff from Localization ==

If call's prefix is in Localization Rule which has some Tariff, this Tariff is used to count call's price instead of User's Tariff.
 
However, '''localized''' destinations are used for the Rerating.
 '''Example:'''
 
1) User dialled 001137067012345. Localization rule was applied:

'''Cut:''' 0011, '''Add:''' , '''Change Tariff to:''' Tariff B.

So this call to 001137067012345 was billed according to Tariff B rates. Localized destination number 37067012345 was saved to the database.

Now when you are rerating, destination number 37067012345 is taken. The rule to cut 0011 and change tariff cannot be applied here.

2) User dials 37067012345. Localization rule was applied:

'''Cut:''' 3706, '''Add:''' 3706, '''Change Tariff to:''' Tariff B.
This call to 37067012345 was billed according to Tariff B rates. Since the destination number is not changed, Localization rules will be applied to number 37067012345 and Tariff B will be used when rerating.

 

=Other notes=

* Re-rate does not take into account Flat Rates and User will be charged for such Calls. Do not do rerating for such User.

* Re-rate does not work when call's DID, LCR or Owner Device is changed by Localization rules.

* Re-rate will take Active Rate when Rates with [[Rate Effective From | Effective From]] are re-rated.

CDR Rerating

2015-10-07T08:29:18Z

Rita: /* Tariff from Localization */

= Description =

This function is used when you change Tariff 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. It affects balance of User and statistics will show information with new Prices.
 
=Usage=

Go to '''SETTINGS –> Billing –> CDR –> Rerating''':
 
[[Image:cdrrerating_image1.png]]
 
Select date interval and User whose Calls you want to rerate. You can select All Users and include calls of Reseller Users.



Also you are able su set '''Ghost Time''' percent, this value will increase user_billsec and/or reseller_billsec and price which depends on changed user_billsec and/or reseller_billsec. Ghost Time will be calculated just from initial billsec, and if you will rerate two timesr with same percentage value, on second time nothing will change. 0 leaves or returns initial user_billsec and/or reseller price.
 
Table below will show which data of Calls will be affected according each selection scenario.
 
{| class="wikitable" border="1" style="border-collapse: separate; border-spacing: 0; border-width: 1px; border-style: solid; border-color: #C0C0C0; padding: 0"
|-
!
! One of Admin Users
! One of Admin Resellers
!                 All Users                
! One of particular Reseller Users
! All Users (Reseller Users Included)
|-
| user_billsec,
user_rate,
user_price
|                 +
|                 -
|                 +
   (just for Admin Users Calls)
|                 +
|                 +
|-
| reseller_billsec,
reseller_rate,
reseller_price
|                 -
|                 +
 (for Calls of All Reseller Users)
|                 +
|                 -
|                 +
       (just for Reseller User Calls)
|}
'''NOTE''' If in Call of Reseller User will be used one of [[Common Use Providers]] Tariff to rerate Reseller Price will be taken form ADDONS -> Resellers -> Common use Providers. If in Call of Reseller User will be used one of Provider which belongs just to Reseller, Provider Price will be rerated by that Provider tariff.
 
* Press '''Rerate in Background(FAST)''' if you want to send Rerating task to [[Background Tasks]]. This way rerating of huge amount of Calls goes faster than SLOW method and also you do not need to wait in current page till rerating will be finished. '''Also Provider Billing details will be rerated'''

* Press '''Rerate in GUI(SLOW)'''. 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'''. '''Provider Billing details will NOT be rerated.''' If you choose this way for Rerating you will be redirected to second step:
 
[[Image:cdrrerating2.png]]
 
Here you see call stats for selected date interval. When you are sure everything is correct – press '''Rerate''':
 
[[Image:cdrrerating3.png]]
 
Here you will see the Difference after Rerating.
 
== Rerate all users ==


Select 'All Users' from User selection to rerate all Users at once in selected period.
 
== Testing mode ==

Select '''Tariff for User''' below '''TESTING MODE''' header:
 
[[Image:cdrtestrerating.png]]
 
...and press '''Test with selected Tariff (SLOW)''' to check differences of Rates for selected User. This way Rerating will not save changes to Calls/Users.

Testing mode is useful for testing "What if User had this Tariff" scenarios.
 
'''Selected Tariff is used just in [[Image:icon_test.png]]Testing mode.''' Rerating functionality rerates old Calls by Tariff set in User Settings, if you did not change Tariff for User, after rerating Calls Prices will be same as were before.
 
== Tariff from Localization ==

If call's prefix is in Localization Rule which has some Tariff, this Tariff is used to count call's price instead of User's Tariff.
 
However, '''localized''' destinations are used for the Rerating.
 '''Example:'''
 
1) User dialled 001137067012345. Localization rule was applied:

'''Cut:''' 0011, '''Add:''' , '''Change Tariff to:''' Tariff B.

So this call to 001137067012345 was billed according to Tariff B rates. Localized destination number 37067012345 was saved to the database.

Now when you are rerating, destination number 37067012345 is taken. The rule to cut 0011 and change tariff cannot be applied here.

2) User dials 37067012345. Localization rule was applied:

'''Cut:''' 3706, '''3706''' , '''Change Tariff to:''' Tariff B.
This call to 37067012345 was billed according to Tariff B rates. Since the destination number is not changed, Localization rules will be applied to number 37067012345 and Tariff B will be used when rerating.

 

=Other notes=

* Re-rate does not take into account Flat Rates and User will be charged for such Calls. Do not do rerating for such User.

* Re-rate does not work when call's DID, LCR or Owner Device is changed by Localization rules.

* Re-rate will take Active Rate when Rates with [[Rate Effective From | Effective From]] are re-rated.

CDR Rerating

2015-10-07T08:29:07Z

Rita: /* Tariff from Localization */

= Description =

This function is used when you change Tariff 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. It affects balance of User and statistics will show information with new Prices.
 
=Usage=

Go to '''SETTINGS –> Billing –> CDR –> Rerating''':
 
[[Image:cdrrerating_image1.png]]
 
Select date interval and User whose Calls you want to rerate. You can select All Users and include calls of Reseller Users.



Also you are able su set '''Ghost Time''' percent, this value will increase user_billsec and/or reseller_billsec and price which depends on changed user_billsec and/or reseller_billsec. Ghost Time will be calculated just from initial billsec, and if you will rerate two timesr with same percentage value, on second time nothing will change. 0 leaves or returns initial user_billsec and/or reseller price.
 
Table below will show which data of Calls will be affected according each selection scenario.
 
{| class="wikitable" border="1" style="border-collapse: separate; border-spacing: 0; border-width: 1px; border-style: solid; border-color: #C0C0C0; padding: 0"
|-
!
! One of Admin Users
! One of Admin Resellers
!                 All Users                
! One of particular Reseller Users
! All Users (Reseller Users Included)
|-
| user_billsec,
user_rate,
user_price
|                 +
|                 -
|                 +
   (just for Admin Users Calls)
|                 +
|                 +
|-
| reseller_billsec,
reseller_rate,
reseller_price
|                 -
|                 +
 (for Calls of All Reseller Users)
|                 +
|                 -
|                 +
       (just for Reseller User Calls)
|}
'''NOTE''' If in Call of Reseller User will be used one of [[Common Use Providers]] Tariff to rerate Reseller Price will be taken form ADDONS -> Resellers -> Common use Providers. If in Call of Reseller User will be used one of Provider which belongs just to Reseller, Provider Price will be rerated by that Provider tariff.
 
* Press '''Rerate in Background(FAST)''' if you want to send Rerating task to [[Background Tasks]]. This way rerating of huge amount of Calls goes faster than SLOW method and also you do not need to wait in current page till rerating will be finished. '''Also Provider Billing details will be rerated'''

* Press '''Rerate in GUI(SLOW)'''. 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'''. '''Provider Billing details will NOT be rerated.''' If you choose this way for Rerating you will be redirected to second step:
 
[[Image:cdrrerating2.png]]
 
Here you see call stats for selected date interval. When you are sure everything is correct – press '''Rerate''':
 
[[Image:cdrrerating3.png]]
 
Here you will see the Difference after Rerating.
 
== Rerate all users ==


Select 'All Users' from User selection to rerate all Users at once in selected period.
 
== Testing mode ==

Select '''Tariff for User''' below '''TESTING MODE''' header:
 
[[Image:cdrtestrerating.png]]
 
...and press '''Test with selected Tariff (SLOW)''' to check differences of Rates for selected User. This way Rerating will not save changes to Calls/Users.

Testing mode is useful for testing "What if User had this Tariff" scenarios.
 
'''Selected Tariff is used just in [[Image:icon_test.png]]Testing mode.''' Rerating functionality rerates old Calls by Tariff set in User Settings, if you did not change Tariff for User, after rerating Calls Prices will be same as were before.
 
== Tariff from Localization ==

If call's prefix is in Localization Rule which has some Tariff, this Tariff is used to count call's price instead of User's Tariff.
 
However, '''localized''' destinations are used for the Rerating.
 '''Example:'''
 
1) User dialled 001137067012345. Localization rule was applied:

'''Cut:''' 0011, '''Add:''' , '''Change Tariff to:''' Tariff B.

So this call to 001137067012345 was billed according to Tariff B rates. Localized destination number 37067012345 was saved to the database.

Now when you are rerating, destination number 37067012345 is taken. The rule to cut 0011 and change tariff cannot be applied here.

2) User dials 37067012345. Localization rule was applied:

'''Cut:''' 3706, '''3706''' , '''Change Tariff to:''' Tariff B.
This call to 37067012345 was billed according to Tariff B rates. Since the destination number is not changed, Localization rules will be applied to number 37067012345 and Tariff B will be used when rerating.

 

=Other notes=

* Re-rate does not take into account Flat Rates and User will be charged for such Calls. Do not do rerating for such User.

* Re-rate does not work when call's DID, LCR or Owner Device is changed by Localization rules.

* Re-rate will take Active Rate when Rates with [[Rate Effective From | Effective From]] are re-rated.

CDR Rerating

2015-10-07T08:28:17Z

Rita: /* Tariff from Localization */

= Description =

This function is used when you change Tariff 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. It affects balance of User and statistics will show information with new Prices.
 
=Usage=

Go to '''SETTINGS –> Billing –> CDR –> Rerating''':
 
[[Image:cdrrerating_image1.png]]
 
Select date interval and User whose Calls you want to rerate. You can select All Users and include calls of Reseller Users.



Also you are able su set '''Ghost Time''' percent, this value will increase user_billsec and/or reseller_billsec and price which depends on changed user_billsec and/or reseller_billsec. Ghost Time will be calculated just from initial billsec, and if you will rerate two timesr with same percentage value, on second time nothing will change. 0 leaves or returns initial user_billsec and/or reseller price.
 
Table below will show which data of Calls will be affected according each selection scenario.
 
{| class="wikitable" border="1" style="border-collapse: separate; border-spacing: 0; border-width: 1px; border-style: solid; border-color: #C0C0C0; padding: 0"
|-
!
! One of Admin Users
! One of Admin Resellers
!                 All Users                
! One of particular Reseller Users
! All Users (Reseller Users Included)
|-
| user_billsec,
user_rate,
user_price
|                 +
|                 -
|                 +
   (just for Admin Users Calls)
|                 +
|                 +
|-
| reseller_billsec,
reseller_rate,
reseller_price
|                 -
|                 +
 (for Calls of All Reseller Users)
|                 +
|                 -
|                 +
       (just for Reseller User Calls)
|}
'''NOTE''' If in Call of Reseller User will be used one of [[Common Use Providers]] Tariff to rerate Reseller Price will be taken form ADDONS -> Resellers -> Common use Providers. If in Call of Reseller User will be used one of Provider which belongs just to Reseller, Provider Price will be rerated by that Provider tariff.
 
* Press '''Rerate in Background(FAST)''' if you want to send Rerating task to [[Background Tasks]]. This way rerating of huge amount of Calls goes faster than SLOW method and also you do not need to wait in current page till rerating will be finished. '''Also Provider Billing details will be rerated'''

* Press '''Rerate in GUI(SLOW)'''. 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'''. '''Provider Billing details will NOT be rerated.''' If you choose this way for Rerating you will be redirected to second step:
 
[[Image:cdrrerating2.png]]
 
Here you see call stats for selected date interval. When you are sure everything is correct – press '''Rerate''':
 
[[Image:cdrrerating3.png]]
 
Here you will see the Difference after Rerating.
 
== Rerate all users ==


Select 'All Users' from User selection to rerate all Users at once in selected period.
 
== Testing mode ==

Select '''Tariff for User''' below '''TESTING MODE''' header:
 
[[Image:cdrtestrerating.png]]
 
...and press '''Test with selected Tariff (SLOW)''' to check differences of Rates for selected User. This way Rerating will not save changes to Calls/Users.

Testing mode is useful for testing "What if User had this Tariff" scenarios.
 
'''Selected Tariff is used just in [[Image:icon_test.png]]Testing mode.''' Rerating functionality rerates old Calls by Tariff set in User Settings, if you did not change Tariff for User, after rerating Calls Prices will be same as were before.
 
== Tariff from Localization ==

If call's prefix is in Localization Rule which has some Tariff, this Tariff is used to count call's price instead of User's Tariff.
 
However, localized destinations are used for the Rerating.
 '''Example:'''
1) User dialled 001137067012345. Localization rule was applied:

'''Cut:''' 0011, '''Add:''' , '''Change Tariff to:''' Tariff B.

So this call to 001137067012345 was billed according to Tariff B rates. Localized destination number 37067012345 was saved to the database.

Now when you are re-rating, destination number 37067012345 is taken. The rule to cut 0011 and change tariff cannot be applied here.

2) User dials 37067012345. Localization rule was applied:

'''Cut:''' 3706, '''3706''' , '''Change Tariff to:''' Tariff B.
This call to 37067012345 was billed according to Tariff B rates. Since the destination number is not changed, Localization rules will be applied to number 37067012345 and Tariff B will be used when re-rating.

 

=Other notes=

* Re-rate does not take into account Flat Rates and User will be charged for such Calls. Do not do rerating for such User.

* Re-rate does not work when call's DID, LCR or Owner Device is changed by Localization rules.

* Re-rate will take Active Rate when Rates with [[Rate Effective From | Effective From]] are re-rated.

MOR - Blocked IPs

2015-10-05T07:53:00Z

Rita:

Go to '''ADDONS -> Monitorings -> Blocked IPs''' to see all blocked IPs with blocking reasons. Blocking reasons give a hint what needs to be fixed.
 
[[File:blocked_ips.png]]
 
Explanation of the values given in the '''Reason''' column can be found [[How_to_be_secure_using_MOR#Additional_software_to_increase_MOR_system_security|here]].
=See Also=

* [[How to block someone's IP]]
* [[How to add Fail2ban exception for my IP]]

Invoices

2015-10-05T05:50:48Z

Rita: /* See also */

= Description =

An invoice, bill or tab is a commercial document issued by a seller to a buyer, relating to a sale transaction and indicating the products, quantities, and agreed prices for products or services the seller had provided the buyer.
 
= Configuration =

You can manage Invoices settings, details, Address information and functionality in the Invoices section of the [[Configuration_from_GUI#Invoices | SETTINGS -> Setup -> Settings]] window. 
If you want show user_billsec instead of billsec in Users invoices and last calls, go to [[Configuration_from_GUI#Visual | Settings –> Various]] and check ''Show user_billsec for Users instead of billsec''
 

Go to User's edit page to enable or disable types of invoices he can see in his invoices details. And also these selected types will be the ones that are sent to user via email.

[[File:Invoices_types.png]]

=Invoices list=

Invoices can be viewed, generated, recalculated, sent, also  list of Invoices can be exported to CSV file (Invoice number is included in the output data) in '''SETTINGS –> Accounting –> Invoices'''. In the main window, you can see list of all Invoices:
 
[[Image:invoices_list_x6.png]]
 
The icons [[Image:icon_check.png]] and [[Image:icon_cross.png]] show whether this invoice is Paid or Unpaid, Sent or Not Sent.
 
'''NOTE''' Each Invoice generated in particular Time Zone and on particular Currency. That means that in each line of Invoice Price and Price with TAX will be shown in currency which is written in Currency column. The Currency column tells which currency Invoice was generated with. Also the exchange rate at the invoice generation moment is saved in the database so it may differ from the up to date exchange rate.
 
Total Price and Price with TAX are calculated into viewing Currency, which you can choose in upper corner of page. In Total price calculation there are used saved exchange Rate of each Invoice.
 
'''Example'''
 
Default system Currency is USD
 
Invoice A is generated on price = 100 in currency EUR with exchange rate 0.5 
Invoice B is generated on price = 100 in currency DZD with exchange rate 0.25 
Invoice C is generated on price = 100 in currency ALL with exchange rate 0.75 
 
But now exchange rate are different: EUR - 0.75, DZD - 0.5 and ALL - 1.0. 
 
If you want to see Total price(sum of these 3 Invoices) in EUR. For Invoice B and C will be set exchange rate of today, but for Invoice A exchange rate will be from that day when Invoice was generated (0.5). 

Same calculations with other currencies: 
If you want to see Total price in USD. For Invoice A, B and C the exchange rates will be set from the day when invoice was generated (EUR - 0.5, DZD - 0.25 and ALL - 0.75). 
If you want to see Total price in DZD. For Invoice A and C will be set exchange rate of today, but for Invoice B exchange rate will be from the day when Invoice was generated (0.25). 
If you want to see Total price in ALL. For Invoice A and B will be set exchange rate of today, but for Invoice C exchange rate will be from the day when Invoice was generated (0.75). 
 
Total price will be 733.3333 USD, or 500 EUR, or 266.6667 DZD, or 700 ALL 
 

=Invoice details=

Click on the [[Image:icon_details.png]] icon to get details of the Invoice:
 
[[Image:invoice_details_x6.png]]
 
Invoice details are shown in Currency which was used to generate Invoice.
 
==General Details==

'''Payment date''' - An option which will set the period of payment on current month. By default ''Years'' and ''Month'' is current, it is only possible to change ''Day of Month'' which is indicating the last date of ''Invoice'' issue. 
'''Currecy of Invoice''' -  Invoices are exported in Invoice currency.

'''Language of Invoice'''
 
Default language in MOR is English. However, you can send invoices in different languages (click on the flag icon on the top of the page).
 

==User Details==

User Details which will be used in xlsx file of invoice. These Details are taken form [[User Details]] and it can be changed manually, but after recalculate it will be reset to same information as it is in [[User Details]].
 
==Files to send or download==

Here you can view the details of all invoices and click on the [[Image:icon_pdf.png]] or [[Image:icon_csv.png]] icons to generate invoice files in PDF or CSV formats accordingly. First icon [[Image:icon_csv.png]] will download a xlsx file.

*'''Invoice XLSX''' xlsx file shows information which are set by you. Check [[Invoices_(MOR_X6)#XLSX_Invoices | XLSX Invoice]]
*'''Invoice''' PDF shows separate services Prices and Total Invoice Prices, CSV shows just Total Invoice Prices.
*'''Invoice detailed''' shows same information like simple Invoice and includes Calls list grouped by Destination.
*'''Invoice by CallerID''' shows Calls list grouped by CallerID.
*'''Invoice Destinations''' shows Calls list grouped by Destination.

You can manage what kind of Invoice should be showed for User and/or sent to User as email attachment in [[User_Details#Invoices |User Details]]. Here you can check ''Show zero Calls'' option if you need to calculate 0 price Calls into Invoices.
 
=== XLSX Invoices ===

XLSX file to export an Invoice is made from your own XLSX format template and information from Invoices main window.
Example of template can be downloaded for [http://www.kolmisoft.com/packets/x6_invoice_template.xlsx here].
 
'''Default XLSX template'''

Create a xlsx file. Template you can prepare as you wish, you can fill it with information of company, draw tables with column names, describe information taken from Invoice main window and so on. Template '''have to be uploaded in SETTINGS -> Setup -> Settings''', Invoices - XLSX tab setting '''XSLX Template'''
* Check '''New XLSX template apply for old Invoices''' if you want to change XLSX template for Invoices which is already generated.
* Check '''Group regular Calls by Destinations''' if you want XLSX file us as detailed Invoice.
 
'''Balance Line settings'''

''Outstanding balance (Debt)'', ''Outstanding balance (Debt) with TAX'', ''Total amount due'' options are present when ''Show balance line'' ('''SETTINGS -> Setup -> Settings''', Invoices - Postpaid - Settings) setting is enabled.
 
'''Configuration for Dynamic information'''

When you already have a template for example with all data of your company, with drawn tables and so on, you need to define where should be shown the dynamic information of invoice. How and what information is displayed can be set same '''SETTINGS -> Setup -> Settings''', Invoices - XLSX tab. Just need to define cell address in correct format.

For example "A2", "B15", "AA9"
 
[[File:callc.png]]
 

'''NOTE''': If you do not want to use particular dynamic value just leave empty input for it.

=== PDF invoices ===

The layout of PDF invoices is not customizable.

Logos are not supported in PDF invoices.

You can print your invoices on paper with a pre-printed logo if you want to send paper invoices.
 
=== CSV invoices ===

Detailed CSV invoice does not include list of all calls. Please use [[How_to_export_CDRs|CDR export]] if you need list of calls in CSV format.
 

==Comment==
 

Sometimes it is useful to add a comment to the invoice. It can be used in xlsx file also.
 
==Invoice Details==
Click on the Invoice Details icon in the Invoice details page
 
[[Image:Service_details.png]]
 
to get Services Details of the Invoice:
 
[[Image:invoice_services_x6.png]]
 
'''Services:'''
 
* '''SMS''' – quantity and price of SMS sent in selected period. Invoice for Reseller counts all SMS that was sent by his users using price for Reseller. Note, that SMS are included to Invoice only when [[SMS Addon]] is used
* '''DID Owner Cost''' – price for having and using own DID(s)
* '''Subscriptions''' – price of subscription applied to selected period. Period considered as month is from first till last day of month and from concrete day of month till next month one day before selected last month. Eg. from 2013-02-26 till 2013-03-25
* '''Calls''' – quantity and price of outgoing calls
* '''Calls from Users''' – quantity and price of resellers users calls
* '''Calls to DIDs''' – price for calling to DID(s)
 

= Invoice generation =

In the main Invoices window, click [[Image:icon_generate_invoices.png]] '''Generate Invoices''' to generate additional invoices:
 
[[Image:generate_invoices_x6.png]]
 
Select for which users to generate invoices. Prepaid and Postpaid users can have separate details. Also, you can check to generate an invoice for only one particular user. Invoice will be generated if selected User existed at least one day of all Invoice period. When Invoice Generation Task is created, admin can check the progress in [[Background Tasks]].

'''NOTE:''' For Prepaid User after every executed Manual Payment will be generated Invoice automatically. 

'''NOTE:''' If Currency of User is different than default System Currency. All his Invoices will be generated with [[Currencies| Currency]] Exchange Rate of current day. 
'''NOTE:''' When generating invoice, current user details will be used. 
 

= Invoice recalculation =

An invoice cannot be modified after being sent. But if you want to allow recalculations after sending invoices, go to: '''Settings -> Setup -> Settings'''. Then click on the tabs: '''Invoices -> Functionality'''. Select ''Invoice allow recalculate after send'':
 
[[Image:ivoicesnewnew.png]]
 
===Recalculate a group of Invoices===

To recalculate invoices in the main Invoices window click [[Image:icon_generate_invoices.png]] '''Recalculate invoices''':
 
[[Image:recalc.png]]
 
Select which type of Invoices or whose Invoices to recalculate.
 
=== Recalculate a single Invoice ===

If an invoice is not marked as paid - it can be recalculated for the same user, same period. To achieve this - press the marked icon:
 
[[File:Recalculate_single_invoice_x6.png]]
 
= Pay unpaid Invoices =

In list of Invoices click [[Image:icon_edit.png]]'''Bulk management''' and button '''Pay unpaid Invoices''' will appear.
 
[[File:pay_unpaid_invoices.png]]
 
Click '''Pay unpaid Invoices''' and '''All''' unpaid Invoices will be marked as paid. For each marked Invoice Payment will be created.
 

= Send invoices by email (manual) =

Click [[Image:icon_email_send.png]] to open more options to enter the period in which you want to send invoices. Click '''Send''' to send invoices to their Users in this period:
 
[[Image:invoices_send_x6.png]]
 
You can use these settings to choose which users should get an email with the invoice. Using these settings you can send emails in parts.
 
* '''Email sending''' option must be enabled in Email Settings to use this feature
* Tip: you can '''send invoices in parts''' by using text fields to send invoices for users which '''Username, First name, Last name''' or '''Number''' begins with some letters. For example: you have users with usernames john, albert and alex. You want to send invoices only for users whose usernames begins with '''"a"''' so you just have to enter '''"a"''' in a '''Username text field''' to send invoices only to albert and alex.

 
= Generate and send invoices by email (automatically) =

Use [http://wiki.kolmisoft.com/index.php/Cron_actions#Generate_Invoices_and_send_by_Email Cron Actions] to Generate and send Emails automatically.

 

=See also=

* [[Accounting]]
* [[Invoice configuration]]
* [[Pay Invoice with balance deduction]]
* [[Outstanding balance(Debt)]]
* [[Invoice by CallerID]]
* [[MOR API invoices_get]]
* [[Cannot generate Invoice]]
* [[Duplicate value in Cell Address field]]

Invoices

2015-10-05T05:50:28Z

Rita:

Duplicate value in Cell Address field

2015-10-05T05:50:09Z

Rita:

If you see error '''Duplicate value in Cell Address field''' when editing any settings at ''SETTINGS -> Setup -> Settings'', it means that you have duplicate values at ''SETTINGS -> Setup -> Settings -> Invoices -> XLSX''.
 
To solve the issue, please go to ''SETTINGS -> Setup -> Settings -> Invoices -> XLSX'' and check for duplicate values, as shown in the picture below.
[[File:Duplicate lines.png]]
 
Please change the fields, so no duplicated would be present.

=See also=

* [[Invoices (MOR X6)]]
* [[Invoice configuration]]

I have a problem with Invoices

2015-10-05T05:49:14Z

Rita:

What problem do you have?

* [[Cannot generate Invoice]]
* [[Detailed Invoice does not match simple Invoice]]
* [[Detailed Invoice does not include CDRs]]
* [[Rerating does not work]]
* [[Invoice_limit_reached|Invoice reach limit: 20]]
* [[Invoice by CallerID does not include Incoming calls]]
* [[Cannot generate PDF]]
* [[Balance, Calls and Invoices are zero]]
* [[There is no Invoice PDF button from User login page]]
* [[Duplicate value in Cell Address field]]
 
= See also =
* [[MOR_Manual#Accounting | Accounting - Invoices]]

Duplicate value in Cell Address field

2015-10-05T05:48:20Z

Rita:

File:Duplicate lines.png

2015-10-05T05:46:47Z

Rita:

Duplicate value in Cell Address field

2015-10-05T05:42:16Z

Rita: Created page with 'If you see error '''Duplicate value in Cell Address field''' when editing any settings at ''SETTINGS -> Setup -> Settings'', it means that you have duplicate values at ''SETTINGS…'

If you see error '''Duplicate value in Cell Address field''' when editing any settings at ''SETTINGS -> Setup -> Settings'', it means that you have duplicate values at ''SETTINGS -> Setup -> Settings -> Invoices -> XLSX''.
 
To solve the issue, please go to ''SETTINGS -> Setup -> Settings -> Invoices -> XLSX'' and check for duplicate values, as shown in the picture below.
 
Please change the fields, so no duplicated would be present.

How to configure 911 service

2015-09-30T05:35:15Z

Rita:

= Method #1 =

NOTE: This guide is for the case when Provider maps 911 or similar number to some external number, like 12125551212
If Provider does not map 911 to external number and requests you to send 911 to Provider directly, check Method #3.

This guide will show how to configure 911, 411 or any other similar service.

You can read more about it at: [http://www.myvoipprovider.com/VoIP_911_Service/ VoIP 911 Services].

First of all, you need a provider that will allow you to use such a service. Some such providers may be found [http://www.voip-info.org/wiki/view/VOIP+911+Service+Providers here].

When you buy a service from such a provider, he will give you an access number to dial.

As an example, let's take the number 12125551212, which should be dialed to access the service.

= Short guide =

In order to configure number 911, we will need to:

* Create a User/Virtual Device for the 911 service.
* Forward a Virtual Device to 12125551212.
* Create a DID 911.
* Assign a Virtual Device to DID 911.

= Detailed guide =

Create User/ Virtual Device for 911 service:

[[Image:settingup911_1.png]]

Forward Virtual Device to 12125551212:

[[Image:settingup911_2.png]]

Create a DID 911 and assign a Virtual Device to it:

[[Image:settingup911_3.png]]

= Method #2 =

* Create User 911.
* Create Virtual Device with Extension 911 for this user.
* Make correct LCR for this user (with correct Provider which can handle 911 calls).
* In Devices Call Flow, Forward the call to the External number given by your Provider.

 
= Method #3 =

This method describes configuration when Provider requires direct number, like 911

Let's configure number 311 which Provider expects.

So User dials number 311. In order it to work we need Localization Rule which will add 00 to it to make it 00311 (min/max=3, cut 3 add 003).
Make sure proper Location is added to the User (or use Global Location for this Rule).

Let's create new Destination with prefix 00311.

In the user tariff we set price for this Destination (00311).

Now on the Provider which receives this call on his Location Rules we create Rule for Destination to remove leading 00 (min/max 5, cut 003, add 3)

Also set some price on Provider's Tariff for this Destination.

That is all - now User can call 311 and such call will go to the Provider.

NOTE: User will see 00311 on his invoices. 00 means this is short technical number.

All this is necessary to avoid conflict with normal Destination Prefixes.

After configuration is it wise to check it with [[Call Tracing]]
 

= See also =
* [[MOR_Manual#DIDs | DIDs]]
* [[Localization | Localization rules]]

Blocked Rates

2015-09-28T06:36:42Z

Rita:

= Description =

Blocked Rate blocks a route to particular Destinations. In Wholesale or Provider type Tariffs you are able to block a prefix (Destination) by setting "-1" or text "Blocked" instead of Price for chosen Destination.

Blocked Rates are useful when Provider does not support some [[Directions and Destinations| Directions or Destinations]]. Some of Providers send a Rates Sheet with a word "Blocked" instead of Price, which means, that this Destination is blocked. MOR [[Rate import from CSV]] recognizes Blocked Rates by "-1" or text "Blocked" instead of Price.

Also it can help you if you want to forbid some [[Directions and Destinations| Directions or Destinations]] for End Users by yourself.
 
'''NOTE''' Retail type Tariff does not support such functionality and Blocked Rates do not affect [[Flat-Rates]].
 
=== Example ===

Let's say we have Destination with Prefix 37062 and Price "blocked" (or -1). All calls to 37062******* will be blocked.
 

= See also =

* [[Block_some_Destination_to_User#Block_Destination_to_all_Users_by_Tariff |Block Destination to all Users by Tariff]]
* [[Block_some_Destination_to_Provider#Block_Destination_to_Provider_by_Tariff| Block some Destination to Provider by Tariff]]
* [[Rate import from CSV]]
* [[Directions and Destinations]]

Blocked Rates

2015-09-28T06:35:38Z

Rita:

Available from MOR X6
= Description =

Blocked Rate blocks a route to particular Destinations. In Wholesale or Provider type Tariffs you are able to block a prefix (Destination) by setting "-1" or text "Blocked" instead of Price for chosen Destination.

Blocked Rates are useful when Provider does not support some [[Directions and Destinations| Directions or Destinations]]. Some of Providers send a Rates Sheet with a word "Blocked" instead of Price, which means, that this Destination is blocked. MOR [[Rate import from CSV]] recognizes Blocked Rates by "-1" or text "Blocked" instead of Price.

Also it can help you if you want to forbid some [[Directions and Destinations| Directions or Destinations]] for End Users by yourself.
 
'''NOTE''' Retail type Tariff does not support such functionality and Blocked Rates do not affect [[Flat-Rates]].
 
=== Example ===

Let's say we have Destination with Prefix 37062 and Price "blocked" (or -1). All calls to 37062******* will be blocked.
 

= See also =

* [[Block_some_Destination_to_User#Block_Destination_to_all_Users_by_Tariff |Block Destination to all Users by Tariff]]
* [[Block_some_Destination_to_Provider#Block_Destination_to_Provider_by_Tariff| Block some Destination to Provider by Tariff]]
* [[Rate import from CSV]]
* [[Directions and Destinations]]

Data import

2015-09-11T09:02:08Z

Rita:

= User/Device/DID/CLI import =

You must import the data in the following order '''from CSV files''':

1. Users data 
2. Devices data 
3. DIDs data 

'''Import Devices from CSV''' link is disabled while Users data is not imported. '''Import DIDs from CSV''' link is disabled while Devices data is not imported.

CLIs data can be imported for already existing Devices by Device ID or for just imported Devices by Temporary Device ID (which is in '''SETTINGS -> Users -> Import''').

== CSV file format ==
 
'''Please note that such details like tariff, LCR, Localizations, IDs should be specified in csv instead names.'''

Example you have LCR called "My first LCR" this LCR ID is 2. You should specify in csv 2, but not My first LCR.

Other way if you will not specify ID system will not find it.

You can upload your data from a CSV file that has the following structure: 
1. Each record is started in a new line. 
2. Each piece of record information is separated by commas (see the example). 
3. Text is enclosed within inverted commas (for example: "name"). 
4. Numbers are written without inverted commas (for example: 4). 
5. The first line can be used for your own needs, such as for naming each field of information. '''If you use this line, do not select the "include first line?" option when asked.
'''

'''For example:''' 
We have a CSV file named '''example.csv''' with the contents:

"id", "username", "password", "e-mail" 
1, "username1", "verySecretPassword", "example@example.com" 
2, "username2", "34r34r4232", "example2@example.com" 
3, "username3", "23er29923wjwe", "example3@example.com" 

'''Explanation:''' 
The '''first line''' is used in this case for naming the fields, so '''when asked, do not select the "include first line?"''' option. (If you won't use the first line for '''naming the fields''', please do select '''"include first line?")''' 
The other three lines (starting 1, 2, 3) are for data records. Each record (the first one is: 1, "username1", "verySecretPassword", "example@example.com") represents: 

1 - user id. 
"username1" - user's username. 
"verySecretPassword" - user's password. 
"example@example.com" - user's e-mail. 

Your CSV file can have any number of lines with any number of fields ('''each line must have the same number of fields''').

P.S. The file should not have any other symbols than those mentioned above. Be careful, as some editors can add their own formatting symbols. In Windows we recommend using Notepad, and in Linux: gedit, kedit, nano.

'''Here are some examples:'''

'''Users CSV file example:''' 

38,2,0,"NULL","NULL",TRUE,"11420471-2-03","Example ZRt","Example ZRt","example example. 11","76/356-650","example@example.hu",04/06/07 01:48 PM," ","en",97,20,0,FALSE,TRUE,5,0,04/06/07,"NULL","NULL",0,FALSE,04/06/07,"Lajosmi",6050,1,0,1,1,TRUE,"NULL","NULL",3 
45,2,0,"95757bfa00bf85b17921d709b5af0234","NULL",TRUE,"12969247-2-03","Example Ákos","Example Example, Kft","Example's 38","70/235-45-89","example@example2.hu",04/06/09 11:28 AM," ","en",97,20,0,FALSE,TRUE,5,0,04/07/09,"NULL","NULL",0,FALSE,04/06/09,"Kecexample",6000,1,0,1,1,TRUE,"NULL","NULL","NULL" 
2138,2,0,"cc1b3ae8986559b10786d6ea73912bd5","NULL",TRUE,"11871633-2-03","Kos Tas","Ks Kft","Alsó ast","20/229-36-78","pli@example.hu",12/13/07 09:42 AM," ","en",97,20,0,FALSE,TRUE,5,0,12/13/07,"NULL","NULL",0,FALSE,12/13/07,"Sirly",6031,1,0,1,1,TRUE,"NULL","NULL","NULL"

In the first column is a Temporary User's ID; the other columns represent other data. As mentioned above, it does not matter in which order your data is represented in the CSV file. It only matters that each column (the data between commas) represents the same data value (id, email, password or other).

'''Devices CSV file example:'''

38,38,76824435,"3676586421","asd dsa u.12","jamaica","7ewfw@sip.example.com",TRUE,"dynamic","sip",TRUE,"NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL" 

45,45,76846301,"3676886421","fgh hgf u.","ergerSiP","76886421@sip.example.com",TRUE,"dynamic","sip",TRUE,"NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL","NULL" 

2138,2138,76505921,"3676505921","jkl lkj u.12","treK","6654@sip.example.com",FALSE,"dynamic","sip",TRUE,"NULL",FALSE,FALSE,10,FALSE,FALSE,FALSE,FALSE,FALSE,FALSE,FALSE,FALSE,,FALSE

The first column here represents "id", the second "user_id", and the others represent other data. YOUR ORDER CAN BE DIFFERENT. The main thing is that each type of data (email, password, id...) must be in the same column. When you are asked to provide data when importing, be sure to assign the correct data from CSV for each column and MOR will know which column represents which data.

'''Dids CSV file example:'''

37067531061,38,38 
37065431061,45,22 
37062431031,2138,2138 

In this example I have used the first column for DID, the second for Temporary User ID, and the third for Temporary Device ID. Your data order can be completely different. You just need to select the correct values when asked.

== Importing data ==

Firstly, from the MOR Graphical User Interface (GUI), navigate to: '''Settings -> Users -> Import''', as in the picture:
 
[[Image:Imports_menu.png|alt import_menu]]
 
=== Temporary User ID & Temporary Device ID ===

These fields are necessary to map DIDs to Devices and Devices to Users.

Temporary IDs can be integer digits only.

Example:

You have some other system with some user_ids, device_ids, DID_ids. You export them into CSV files. The relations between them are kept by these IDs, so importing them to MOR will keep these relations.

First you need to import a User with some temporary ID, then make sure this ID is near each Device you are trying to import. Select this ID as Temporary User ID when importing Devices. That way these devices will be mapped to this User.

== Importing users ==

To import users, click on the "Import Users from CSV" link:
 
[[Image:Imports_users_selection.png|alt Imports_users_selection]]
 
Now locate the CSV file in your local system by clicking the "Browse" button. As indicated above, don't click the "Include first line?" checkbox if you have used the first line for commenting on the fields.
 
[[Image:Upload_csv_file.png|alt upload_csv_file]]
 

'''Click on the "Upload file" button.'''

In next menu, please fill in the requested data (you should select the correct field from your CSV file). '''You must select a field for each selection box marked with an asterisk (*) symbol'''. Having finished, press the "Assign columns" button.
 
[[Image:Select_fields.png|alt select_fields]]
 
On the next page, check whether all the data was assigned correctly. If not, click the "Back" button and correct the mistakes. If all the data was assigned correctly, click the "Confirm columns" button. After the "Columns assigned" notification informs you that data successfully imported.

'''NOTE:''' When you are going to import CSV file, which includes Tariff and/or LCR, the '''IDs''' of Tariff and LCR should be specified in CSV file.

Where to find ID of Tariff or LCR? Just go to some of them and press on edit icon, where Browser navigator bar will show full link with location, last number is ID.

Having completed all these steps, you should see a page similar to this:
 
[[Image:Users_imported.png|alt Users_imported]]
 

== Importing devices ==

Click on the "Import Devices from CSV" link and follow the same procedure for uploading a CSV file as in the "Importing users" section. That is, locate the CSV file in your local system by clicking the "Browse" button. As stated before, don't click the "Include first line?" checkbox if you have used the first line for commenting on the fields. Click on the "Upload file" button. Now you should see a menu similar to this:
 
[[Image:Device_selection.png|alt device_selection]]
 
In this menu, once again select a matching field from your CSV file for each selection box. '''It is very important to correctly select Temporary User ID and Temporary Device ID, because these fields are necessary to map DIDs to Devices and Devices to Users.'''

'''NOTE:''' When you are going to import CSV file, which includes Location, the '''name''' of Location should be specified in CSV file.

Having finished selecting the fields, press the "Assign columns" button.

Once the columns are assigned, check whether all the data is correct. If not, press the "Back" button and correct the mistakes. Now, if you haven't made any mistakes, you should see two tables similar to these:
 
[[Image:Users_devices_tables.png|alt users_devices]]
 

== Importing DIDs ==

Click on the "Import Devices from CSV" link and follow the same steps for uploading a CSV file as in the "Importing users" section. That is,
locate the CSV file in your local system by clicking the "Browse" button. As stated before, don't click on the "Include first line?" checkbox if you have used the first line for commenting on the fields. Click on the "Upload file" button. Now you should see a menu similar to this:
 
[[Image:Dids select.png|alt dids_selection]]
 
Select all matching fields from your CSV here. It is very IMPORTANT to select the CORRECT Temporary User ID and Temporary Device ID! Having selected everything correctly, click the "Assign columns" button. In the next page, if all imported data is assigned correctly, click the "Confirm columns" button. If you haven't made any mistakes, you should now see a "DIDs successfully imported" message.

== Importing CLIs ==



CLIs data can be imported for already existing Devices by Device ID or for just imported Devices by Temporary Device ID.

Click on the "Import CLIs from CSV" link and follow the same steps for uploading a CSV file as in the "Importing users" section. That is, locate the CSV file in your local system by clicking the "Browse" button. As stated before, don't click on the "Include first line?" checkbox if you have used the first line for commenting on the fields. Click on the "Upload file" button. Now you should see a menu similar to this:
 
[[Image:CLIs select.png|alt clis_selection]]
 
Select all matching fields from your CSV here. It is very IMPORTANT to select the CORRECT Device ID (if it is selected "Use Device ID") or Temporary Device ID(if it is selected "Use Temporary Device ID")! Having selected everything correctly, click the "Assign columns" button. In the next page, if all imported data is assigned correctly, click the "Confirm columns" button. If you haven't made any mistakes, you should now see a "CLIs successfully imported" message.

You can now check whether all your data has been imported.

To check users, go to: '''SETTINGS –> Users'''.

To check devices, go to: '''SETTINGS –> Users –> Devices'''.

To check DIDs, go to: '''SETTINGS –> Billing –> DIDs'''.

To check CLIs, go to: '''SETTINGS –> Billing –> Devices –> CLIs'''.
 
[[Image:Dids devices users.png|alt dids_devices_users]]
 

= Explanation of IDs =

Many users find it difficult to understand what User_ID or Device_ID mean in Data Import.

These are temporary IDs created only for import purposes. '''These IDs have NO relation to User ID/Device ID in the Users/Devices windows of the MOR system.'''

The workflow of Import is:

1. We have Users/Devices/DIDs.

2. We create a separate file with Users and add a unique ID to every User. The ID is NOT related to User ID in MOR DB/GUI.

3. We create a separate file with Devices and add a unique ID to every Device. The ID is NOT related to Device ID in MOR DB/GUI. Near each device we add a temporary User ID from previous file, which shows which Device belongs to which User.

4. We create a separate file with DIDs and near each DID add a User_ID/Device_ID that shows which DID belongs to which User/Device.

5. We import the User file, and MOR remembers the temporary User_ID.

6. We import the Device file, and MOR remembers the temporary Device_ID and checks the User_ID for such Devices which have a User_ID near them in the file. It assigns this Device to the appropriate User.

7. We import the DID file. MOR checks the Device_ID/User_ID in the DIDs file and assigns a DID to this Device/User.

8. We import the CLI file. If during import there is selected "Use Temporary Device ID " MOR checks the Device_ID in the CLIs file and assigns a CLI to this Device. If it is checked "Use Device ID" MOR checks the ID of Device in the CLIs file and assigns a CLI to this Device

NOTE: in all processes, only a TEMPORARY User_ID/Device_ID is used. In the database, Users' ID/Devices' ID are not used directly. They have no relation to the User_ID/Device_ID from import files.

= See also =

* [[Export Users and Devices]]

Export Users and Devices

2015-09-11T09:00:49Z

Rita: Created page with 'It is not possible to export Users and Devices to a CSV file in MOR. However, you can create a backup of all MOR configurations in GUI and download it. T…'

It is not possible to export Users and Devices to a CSV file in MOR.
 
However, you can create a [[Backup_system | backup]] of all MOR configurations in GUI and download it. The backup includes all the Users and their Devices.

= See also =

* [[Backup system]]
* [[Data import]]

Additional Asterisk Server

2015-08-25T06:26:05Z

Rita:

In order to add an additional Asterisk server to your system:
 
'''1) Create a new MySQL User in Database server for the Asterisk:'''
CREATE USER 'mor'@'IP_OF_ADDITIONAL_ASTERISK_SERVER' IDENTIFIED BY 'mor';

GRANT ALL PRIVILEGES ON `mor`.* TO 'mor'@'IP_OF_ADDITIONAL_ASTERISK_SERVER' WITH GRANT OPTION ;
Change the IP_OF_ADDITIONAL_ASTERISK_SERVER to your Asterisk server IP.
 
'''2) Change Asterisk configuration files on Asterisk server:'''
mcedit /etc/asterisk/mor.conf
''hostname=localhost'' change to ''hostname=IP_OF_DATABASE_SERVER''
 
''server_id=1'' change to ''server_id=2''

mcedit /etc/asterisk/res_mysql.conf (do not need to edit in fresh X5 installs)
''dbhost = 127.0.0.1'' change to ''dbhost = IP_OF_DATABASE_SERVER''

mcedit /var/lib/asterisk/agi-bin/mor.conf
''host = localhost'' change to ''host = IP_OF_DATABASE_SERVER''
 
''server_id = 1'' change to ''server_id = 2''

mcedit /etc/asterisk/res_config_mysql.conf
''dbhost = 127.0.0.1'' change to ''dbhost = IP_OF_DATABASE_SERVER''

mcedit /etc/mor/system.conf
''server_id = 1'' change to ''server_id = 2''

mcedit /etc/asterisk/manager.conf
after line "permit=127.0.0.1/255.255.255.0" add line ''permit=IP_OF_GUI_SERVER/255.255.255.0''
 
'''3) Add the Asterisk server to GUI:'''

Open MOR GUI and go to SETTINGS > Billing > Servers. Add the Asterisk server here. All values can be left the same as MAIN server, but ID, Hostname and Server IP should be changed.

Various Guides

2015-08-25T06:25:33Z

Rita: /* Asterisk */

==Guides==

===Testing===

* [[Performance Testing with Sipp]]

===Installation===
* [[G723/G729 Codec installation]]
* [[MySQL Replication]]
* [[Sangoma Wanpipe installation]]
* [[Centos installation]]
* [[Setting up https secure connection | Setting up https secure connection (SSL)]]
* [[Install Eluna Stats]]
* [[Configure SSH connection between servers]]
* [[Fail2Ban installation]]
* [[OVH server installation]]
** [[OVH server Rescue Mode]]

===MySQL===
* [[Live Backups]]
* [[How to access MySQL database via WEB]]
* [[Lost MySQL root password?]]
* [[Backup old Calls]]
* [[How to recover corrupted MySQL data]]
* [[How to restore system from Backup which are made from GUI]]

===Asterisk===

* [[Ports which should be opened]]
* [[Asterisk under NAT]]
* [[Asterisk CLI debug]]
* [[Which codecs should I choose for devices]]
* [[How to restart Asterisk server]]
* [[How to change RTP port range for Asterisk]]
* [[Two SIP listening ports for single Asterisk]]
* [[sip.conf]]
* [[RTPAUDIOQOS Demystified]]
* [[Explanation about stucked calls]]
* [[SIP Domains in Asterisk]]
* [[Additional Asterisk Server]]

===Other===

* [[Voice quality]]
** [[How to increase call volume]]
* [[Latency]]
* [[Softphones]]
* [[How to change server's IP]]
* [[Server working very slowly]]
* [[Timezone in RoR]]
* [[Sound files]]
** [[Convert WAV file to Asterisk playable format]]
** [[TTS Engines]]
* [[Fax over VoIP]]
** [[Send_Receive_Fax-T38 | Send Receive Fax over T38]]
* [[NTP: Server Time Sync]]
* [[Selinux]]
* [[Fail2Ban]]
* [[How to block someone's IP]]
* [[High availability (Heartbeat clustering)]]
* [[Migration to other server]]
* [[What is direct URL]]
* [[How to join Webex meeting]]
* [[How to capture packets using wireshark]]
* [[SIP Capture with TCPDump]]
* [[SIP Call Flow]]
* [[Load-balancing using DNS SRV]]

=== Softphones ===

* [[Idefisk configuration| Idefisk 1.37 Configuration]]
* [[X-PRO Configuration]]
* [[Video call with CounterPath eyeBeam]]

=== Hardware ===

* [[Cisco Gateway Settings]]
* [[How to connect to Huawei devices]]
* [[TELES configuration]]
* [[VoiceBlue Ring problem ]]
* [[Linksys SPA942 Configuration]]
* [[Linksys PAP2]]
* [[How to get pause on mobile phone keyboard]]
* [[CDMA support]]

 

==Documentation==

* [[VoIP Business Glossary]]
* [[VoIP Advantages]]

* [[Why we do not suggest to use IAX2]]
* [[Various documentation| FAQ]]

* '''[[Ticket System | Support System]]'''
** [[Support System explanation]]
** [[Create new Ticket in Ticket System]]
** [[Write comment to the Ticket]]
** [[In what order tickets are solved in Support System]]
** [[Support]]
** [[Amazon S3 Backups]]

* [[Forum|Forum]]
* [[Pricing structure]]
* [[Training]]

Additional Asterisk Server

2015-08-25T06:22:42Z

Rita:

In order to add an additional Asterisk server to your system:
 
'''1) Create a new MySQL User in Database server for the Asterisk:'''
CREATE USER 'mor'@'IP_OF_ADDITIONAL_ASTERISK_SERVER' IDENTIFIED BY 'mor';

GRANT ALL PRIVILEGES ON `mor`.* TO 'mor'@'IP_OF_ADDITIONAL_ASTERISK_SERVER' WITH GRANT OPTION ;
Change the IP_OF_ADDITIONAL_ASTERISK_SERVER to your Asterisk server IP.

'''2) Change Asterisk configuration files on Asterisk server:'''
mcedit /etc/asterisk/mor.conf
''hostname=localhost'' change to ''hostname=IP_OF_DATABASE_SERVER''
 
''server_id=1'' change to ''server_id=2''

mcedit /etc/asterisk/res_mysql.conf (do not need to edit in fresh X5 installs)
''dbhost = 127.0.0.1'' change to ''dbhost = IP_OF_DATABASE_SERVER''

mcedit /var/lib/asterisk/agi-bin/mor.conf
''host = localhost'' change to ''host = IP_OF_DATABASE_SERVER''
 
''server_id = 1'' change to ''server_id = 2''

mcedit /etc/asterisk/res_config_mysql.conf
''dbhost = 127.0.0.1'' change to ''dbhost = IP_OF_DATABASE_SERVER''

mcedit /etc/mor/system.conf
''server_id = 1'' change to ''server_id = 2''

mcedit /etc/asterisk/manager.conf
after line "permit=127.0.0.1/255.255.255.0" add line ''permit=IP_OF_GUI_SERVER/255.255.255.0''

'''3) Add the Asterisk server to GUI:'''

Open MOR GUI and go to SETTINGS > Billing > Servers. Add the Asterisk server here. All values can be left the same as MAIN server, but ID, Hostname and Server IP should be changed.

Additional Asterisk Server

2015-08-25T06:17:20Z

Rita: Created page with 'In order to add an additional Asterisk server to your system: '''1) Create a new MySQL User in Database server for the Asterisk:''' CREATE USER 'mor'@'IP_OF_ADDITIONAL_…'

In order to add an additional Asterisk server to your system:
 
'''1) Create a new MySQL User in Database server for the Asterisk:'''
CREATE USER 'mor'@'IP_OF_ADDITIONAL_ASTERISK_SERVER' IDENTIFIED BY 'mor';

GRANT ALL PRIVILEGES ON `mor`.* TO 'mor'@'IP_OF_ADDITIONAL_ASTERISK_SERVER' WITH GRANT OPTION ;
Change the IP_OF_ADDITIONAL_ASTERISK_SERVER to your Asterisk server IP.

'''2) Change Asterisk configuration files on Asterisk server:'''
mcedit /etc/asterisk/mor.conf
''hostname=localhost'' change to ''hostname=IP_OF_DATABASE_SERVER''
''server_id=1'' change to ''server_id=2''

mcedit /etc/asterisk/res_mysql.conf (do not need to edit in fresh X5 installs)
''dbhost = 127.0.0.1'' change to ''dbhost = IP_OF_DATABASE_SERVER''

mcedit /var/lib/asterisk/agi-bin/mor.conf
''host = localhost'' change to ''host = IP_OF_DATABASE_SERVER''
''server_id = 1'' change to ''server_id = 2''

mcedit /etc/asterisk/res_config_mysql.conf
''dbhost = 127.0.0.1'' change to ''dbhost = IP_OF_DATABASE_SERVER''

mcedit /etc/mor/system.conf
''server_id = 1'' change to ''server_id = 2''

mcedit /etc/asterisk/manager.conf
after line "permit=127.0.0.1/255.255.255.0" add line ''permit=IP_OF_GUI_SERVER/255.255.255.0''

'''3) Add the Asterisk server to GUI:'''

Open MOR GUI and go to SETTINGS > Billing > Servers. Add the Asterisk server here. All values can be left the same as MAIN server, but ID, Hostname and Server IP should be changed.

End IVR Type

2015-08-20T08:56:35Z

Rita: /* End IVR #5 */

=Description=

This option allows to select which End IVR to use for Calling Card Dial-Plan.

[[File:End_ivr_type2.png]]

 
== End IVR #1 ==

This is default IVR.

When call ends or caller presses * to terminate call, he comes into this IVR.

Here he hears message: "Press 1 to dial new destination, press 2 to dial same destination, press 3 to check your balance, press 4 to use new card"

When user presses some button [1,2,3,4,5] he will be able to do such actions:

1 - Can enter new destination
2 - Can dial same old destination (redial)
3 - Can hear his balance
4 - Can enter new PIN to use new Card
5 - Topup a cards balance with another card balance

More information about card topup functionality can be found [[Card TopUp | here]]

'''NOTE:''' This End IVR is dedicated only to Calling Card Dial plan.

In Authorization by PIN/ANI End IVR#1 options are these:

1 - Can enter new destination
2 - Can dial same old destination (redial)
3 - Can hear his balance

 

== End IVR #2 ==

This type of IVR allows to enter new Destination or redial old one by pressing 1#.

User can enter destination and # to dial to it, or he can enter 1# and then system will redial old destination.

 

== End IVR #3 ==

This IVR is just a hangup. E.g. call will end at once using this IVR and no further actions will be possible.

'''NOTE:''' This End IVR is dedicated only to Calling Card Dial plan.

In Authorization by PIN/ANI the End IVR#3 forwards Call to End IVR#1.

 

== End IVR #4 ==

When user presses some button [1,2,3,4] he will be able to do such actions:

1 - Can enter new destination
2 - Can dial same old destination (redial)
3 - Can enter new PIN to use new Card
4 - Topup a cards balance with another card balance

 
== End IVR #5 ==

Menu for Resellers mostly which is used to Topup cards for their clients.

When user presses some button [1,2,3,4,5,6,7] he will be able to do such actions:

1 - Can enter new destination
2 - Can dial same old destination (redial)
3 - Can enter new PIN to use new Card
4 - Topup a cards balance with another card balance
5 - Topup a cards balance with another card balance with PIN
6 - Topup a cards balance with another card balance with CLI
7 - Topup a cards balance with another card balance with NUM + PIN

These options are used to increase Calling Card A balance with Calling Card B money:

''5 - Topup a cards balance with another card balance with PIN .''This option asks you to enter PIN number of card B.

''6 - Topup a cards balance with another card balance with CLI.'' When cards is authenticated by caller ID, this option asks you to enter phone number of card B.

''7 - Topup a cards balance with another card balance with NUM + PIN.'' This option asks you to enter card's number and PIN of the card B.

 

== End IVR #6 ==

This End IVR allows the user to topup a card's balance with another card('''Action #5''') by pressing '*' instead of dialing a destination. After toping up the card, the user will be prompted to dial a destination again.
 
Once the call is terminated, the user will hear the same menu options as with End IVR #1:

When the user presses some button [1,2,3,4,5] he will be able to do such actions:

1 - Can enter new destination
2 - Can dial same old destination (redial)
3 - Can hear his balance
4 - Can enter new PIN to use new Card
5 - Topup a cards balance with another card

 

== End IVR #7 ==

This End IVR allows the user to control whether his calling card is authorized by pin/number or automatically by CallerID number. This is useful when users want to temporarily disable automatic authorization for security reasons.

Note: Pinless dialing should be enabled in dial plan.

When the user presses some button [1,2,3,4] he will be able to do such actions:

1 - Save CallerID number
2 - Clear saved CallerID number
3 - Can dial same old destination (redial)
4 - Can enter new destination

 

== End IVR #8 ==

This End IVR allows to clear saved CallerID number and change card.

When the user presses some button [1,2,3,4] he will be able to do such actions:

1 - Can enter new destination
2 - Can dial same old destination (redial)
3 - Clear saved CallerID number
4 - Can enter new PIN to use new Card

 

== End IVR #9 ==

End IVR #9 is very similar to End IVR #5. The difference between End IVR #9 and End IVR #5 is that End IVR #9 allows to play custom sound files in card topup.

Custom files are these:

cc_enter_pin_to_topup.wav - played when user is asked to enter pin of a card to which balance should be added.
cc_enter_pin_to_take_topup.wav - played when user is asked to enter pin of a card from which balance should be deducted to primary card.

these files should be prerecorded and uploaded to /var/lib/asterisk/sounds/mor/ivr_voices/en (if language is not english, then instead of "en" there should be directory "ru" or any other language).

Also, End IVR #9 skips topup intro message.

When user presses some button [1,2,3,4,5,6,7] he will be able to do such actions:

1 - Can enter new destination
2 - Can dial same old destination (redial)
3 - Can enter new PIN to use new Card
4 - Topup a cards balance with another card balance
5 - Topup a cards balance with another card balance with PIN
6 - Topup a cards balance with another card balance with CLI
7 - Topup a cards balance with another card balance with NUM + PIN

 

== End IVR #10 ==

End IVR #10 is the same as IVR #1 except it does not allow to topup calling cards.

When user presses some button [1,2,3,4] he will be able to do such actions:

1 - Can enter new destination
2 - Can dial same old destination (redial)
3 - Can hear his balance
4 - Can enter new PIN to use new Card

 

= See also =

* [[Authorization_by_ANI#End_IVR|End IVRs for Authorization by ANI dial plan]]

PBX Functions

2015-08-19T06:28:27Z

Rita: /* Hide CallerID */

=Usage=

Go to '''ADDONS –> PBX functions'''
 
[[image:pbx_finctions_path.png]]
 
Here you can see list of PBX functions which are available on MOR.
 
[[image:Pbx_functions.png]]
 

'''NOTE''' that Reseller cannot manage PBX functions by him self, Admin have to allow/disallow PBX functions for Reseller So if you want to disable some PBX function for resellers, just click on [[Image:icon_check.png]] icon to disable it, as it is marked in picture above.
==Functions available on MOR==

===Tell balance===

This function allows one to hear the balance of the account. The account holder can be a user or card.

It is possible to hear the balance in different currencies and languages.

First MOR tries to find the user by accountcode, then by CallerID using ANI. Later, if these methods fail, it asks for a PIN.

MOR asks for the Card PIN if the card balance should be told.
 
===Milliwatt===

Generate a Constant 1000Hz tone at 0dbm (mu-law) - good for testing line quality for loss-packets.

[http://www.voip-info.org/wiki/index.php?page=Asterisk%20cmd%20Milliwatt More info about Milliwatt]
 
===Use Voucher===

This function allows one to enter a voucher number over the phone to top up the user's balance.

* [[How to add Use Voucher function to DID]]
 
=== Dial Local ===

More info: [[Dial Local]]
 
=== DTMF Test ===

System will ask you to enter a destination and press #. After that, it will tell you what you entered.

If the system asks you again to enter the destination WITHOUT first telling you what you entered, you have problems with DTMF.
 
=== Check VoiceMail ===

[[Voicemail#Retrieve_VoiceMail_from_outside_over_DID | Allows to check VoiceMail over DID]].
 
===Card TopUp===

This functionality allows to increase Calling Card A balance with Calling Card B money. More information: [[Card TopUp]].
 
=== Current Time ===

Says a date and time to the caller.
 
=== Hide CallerID ===

Allows to hide CallerID number. 
If Client calls to this PBX Function, his CallerID presentation will be set to "unavailable". CallerID presentation is set on Device level. All Calls from this Device will be anonymous. To disable anonymous CallerID, Client has to call to this PBX function again. Each Call to this PBX function toggles CallerID anonymity. CallerID anonymity can be changed manually as well. To do so, just change "Incoming Call CallerID Presentation" to "None" in [[Device settings]].
 

=See also=

* [[PBX Functions Addon]]
* [[PBX connection to MOR]]
* [[I have a problem with PBX Functions]]

I want to change billing directory in URL

2015-08-12T06:16:31Z

Rita:

If you want to change billing directory in URL:
 
[[File:billing.png]]
 
Go to /home/mor/config/environment.rb file and

Web_Dir = Rails.env.to_s == 'production' ? "/billing" : ''

change to

Web_Dir = Rails.env.to_s == 'production' ? "/new_dir_name" : ''

'''NOTE''' new_dir_name has to be your chosen name
 
Then

mv /var/www/html/billing /var/www/html/new_dir_name
 
And edit /etc/httpd/conf.d/m2.conf file by changing

RailsBaseURI /billing

to

RailsBaseURI /new_dir_name
 
Make

service httpd restart

and your new_dir_name will appear in URL.

I HAVE A PROBLEM

2015-07-17T11:50:29Z

Rita:

Before proceeding, you need to update MOR GUI to the newest version because many problems can be fixed just by updating GUI.

* [[Update MOR GUI]]

After the update, check your problem. If it still exists - continue.

 
----
 

With '''WHAT''' do you have a problem?

 

I have a problem with:
* [[I have a problem with Auto Dialer| Auto Dialer]]
* [[I have a problem with Accountant| Accountant]]
* [[I have a problem with API| API]]
* [[I have a problem with Asterisk | Asterisk]]
* [[I have a problem with Authorization by PIN | Authorization by PIN]]
* [[I have a problem with Backups | Backups]]
* [[I have a problem with Balance | Balance]]
* [[I have a problem with Callback | Callback]]
* [[I have a problem with Calling Cards | Calling Cards]]
* [[I have a problem with CallerID | CallerID]]
* <big><big><big>[[I have a problem with Calls | Calls]]</big></big></big>
* [[I have a problem with CallFlow | Callflow]]
* [[I have a problem with Codecs | Codecs]]
* [[I have a problem with Currencies | Currencies]]
* [[I have a problem with CSV | CSV]]
* [[I have a problem with Devices | Devices]]
* [[I have a problem with DIDs | DIDs]]
* [[I have a problem with DTMF | DTMF]]
* [[I have a problem with Email | Email]]
* [[I have a problem with Fax | Fax]]
* [[I have a problem with Google Maps| Google Maps]]
* [[I have a problem with GUI | GUI]]
* [[I have a problem with HGC Codes| HGC Codes]]
* [[I have a problem with Invoices | Invoices]]
* [[I have a problem with IVR | IVR]]
* [[I have a problem with Languages | Languages]]
* [[I have a problem with MOR FREE| MOR FREE]]
* [[I have a problem with MySQL| MySQL]]
* [[I have a problem with Network| Network]]
* [[I have a problem with Payment Gateway | Payment Gateway]] (PayPal, WebMoney and similar)
* [[I have a problem with Payments | Payments]]
* [[I have a problem with PAI | PAI]]
* [[I have a problem with PBX Functions | PBX functions]]
* [[I have a problem with Provider| Provider]]
* [[I_have_a_problem_with_Queues | Queues ]]
* [[I have a problem with Recordings | Recordings]]
* [[I have a problem with Registration | Registration]]
* [[I have a problem with Ring Group | Ring Group]]
* [[I have a problem with Security| Security]]
* [[I have a problem with Server| Server]]
** [[I have a problem with Hard Drive | Hard Drive (HDD/SSD)]]
* [[I have a problem with SIP TLS | SIP TLS]]
* [[I have a problem with SMS | SMS]]
* [[I have a problem with Statistics| Statistics]]
* [[I have a problem with Subscriptions| Subscriptions]]
* [[I have a problem with SVN | SVN]]
* [[I have a problem with Rates | Tariffs/Rates]]
* [[I have a problem with Taxes | Taxes]]
* [[I have a problem with Time | Time]]
* [[I have a problem with Transfer | Transfers]]
* [[I have a problem with Voicemail| Voicemail]]
* [[I have a problem with Voucher | Vouchers]]
* [[I have a problem with RFC | RFC]]
* [[I have a problem with MNP | MNP]]
* [[I have a problem with SIP Proxy | SIP Proxy/Balancer]]
* [[I have a problem with Call Flood | Active calls flood (too many connecting calls)]]
* [[Other|Other]]

Transfers do not work

2015-07-17T11:48:45Z

Rita: Created page with 'If transfers do not work, please make sure you have enabled the option in sip.conf. Instructions on how to enable transfers in MOR can be found here.'

If transfers do not work, please make sure you have enabled the option in sip.conf.
 
Instructions on how to enable transfers in MOR can be found [[Transfers_with_MOR |here]].

Configuration from GUI

2015-07-15T11:30:58Z

Rita: /* Various */

=Where I can find this settings menu?=

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

[[File:Settings_path.png]]
 
== Global ==

'''SETTINGS –> Setup –> Settings –> [[Global_Settings | Global]]'''

* '''Company''' – name of the company.
* '''Company email''' – email address of the company.
* '''Version''' – version number.
* '''Copyright Title''' – copyright title supports html text format.
* '''Browser Title''' – browser title for MOR GUI.
* '''Front Page Text''' – some text to place on front page.
* '''Login Page Text''' – some text to place on login page
 

== Registration ==

'''SETTINGS –> Setup –> Settings –> Registration'''

* '''Enabled?''' – do we want to let users [[Online registration | register online]] automatically?
* '''Hide registration link in Login page?''' – tick this option if you do not want users to be able to register online.
* '''Default country'''
* '''Asterisk server IP''' – what IP should user connect his softphone/server to?
* '''Send email to user after registration''' – just yes or no.
* '''Send email to admin after registration''' – just yes or no.
* '''[[ReCAPTCHA]] configuration'''
* '''[[User_Agreement|Terms and conditions]]'''
* '''Allow to create user and device with matching username and password''' – just yes or no. If this setting is not enabled - username and password from MOR GUI will have to be different than device's username and password.
* '''Enable VAT checking for EU companies''' – you can choose whether to check '''TAX Registration Number:''' during the [[Online registration]] or not. Done over [http://ec.europa.eu/taxation_customs/vies/?locale=lt EU page].
* '''Allow blank VAT for EU companies''' – allows to leave blank '''TAX Registration Number:''' field during the [[Online registration]]
 
===See also===

* [[Default user]]
 

== Invoices ==

'''SETTINGS –> Setup –> Settings –> Invoices'''

Invoice configuration is divided into '''Prepaid''' and '''Postpaid'''.
 
=== Settings ===

* '''Number start''' – how the invoice number should start.
* '''Number length''' – the length in digits AFTER Number Start – e.g. Number Start is not counted into this length. This value is active only for first number type.
* '''Number type''' – select more appropriate number format.
* '''Invoice period start''' – when invoices are issued, default is first day of the month.
* '''Show calls in detailed invoice''' – yes or no.
* '''[[Outstanding balance(Debt) | Show balance line]]''' – shows balance line in Invoice PDF files when Invoice is generated from first day of month till last one.
* '''Round finals to 2 decimals''' – round or not total in Invoices to 2 decimals.
* '''Use short file name'''
* '''Invoice Page Limit:''' – invoice will be generated till this page limit. Additional page is added with an explanation that page limit was reached.
 

=== Details ===

* '''Address format''' – changes field positioning in the invoice for the address. You must select it in order to show address in the invoice.
* '''Address line 1-4''' – address info.
* '''Bank details line 1-5''' – bank details info.
* '''Invoice balance line''' – line for balance (e.g. "Your current balance is")
* '''Invoice to pay Line'''
* '''End title''' – a title for the end of your invoice.

* '''Add Average rate in detailed invoice''' – adds average rate in detailed invoice.
* '''Show Time in Minute''' – shows time in mm:ss instead of hh:mm:ss, for example, not 02:13:42 but 133:42.
 

=== Additional ===

* '''Show additional details on separate page''' – enables additional custom page at the end of the invoice.
* '''Additional details''' – the information to be displayed in a custom page at the end of the invoice.
 
=== Functionality ===

* '''Invoice allow recalculate after send''' – allows recalculating invoices after being sent.
 
===See also===

* [[Invoices]]
* [[Invoice configuration]]
* [[Invoices (MOR X6)]]
 

== Emails ==

'''SETTINGS –> Setup –> Settings –> Emails'''

* ''' Sending globally enabled''' – if this is checked, you will be able to use Mass Mailing and the system will send emails upon user registration.
* '''SMTP server''' – hostname of the server.
* '''Port''' – port of SMTP server.
* '''Login''' – username to your SMTP server.
* '''Password''' – password to your SMTP server.
* '''Batch size''' – value for Mass Mailing, or how many emails to send at once.
* '''From''' – email address to put into From: field in email. It is required to fill in(correctly) this field if email sending is enabled.
* '''Test email sending''' – save settings and press this to test email sending. Do not forget to save settings first!

'''P.S. Remember to set admin's e-mail in admin's personal settings. Without this test email sending will not work!'''

 
===See also===

* [[Emails|Mass Emails]]
* [[Email variables]]
 
== Various ==

'''SETTINGS –> Setup –> Settings –> Various'''

* '''Days for DID to close''' – how long to keep DID closed before making it free.
* '''Agreement number length''' – number length in digits.
* '''Change dahdi''' – Sometimes this is used to change "Zap" to something else in GUI, e.g. to "PSTN", mainly because not everybody knows what Zap stands for. PSTN is more common.
* '''Device PIN length''' – length for automatically generated PIN codes for devices.
* '''Device range MIN''' – from which extension automatic extension assigning starts.
* '''Device range MAX''' – till which extension automatic extension assigning goes.
** For example, from this interval: [Device range MIN, Device range MAX] MOR will select extensions for newly created Devices.
* '''Enable "Allowed addresses" option for Devices:''' – allows [[Device_settings#Allowed_Addresses | Allowed Addresses]] functionality in Devices settings.
* '''Enable "Allowed addresses" option for Providers:''' – allows Allowed Addresses functionality in Providers settings.
* '''User wholesale tariff enabled''' – enables user wholesale tariffs.
* '''Play IVR for unauthenticated call'''
* '''Banned CLIs default IVR''' – default IVR to route banned numbers.
* '''Do not allow duplicate device user names'''
* '''Do not allow users to change their email addresses'''
* '''Allow short passwords in devices'''
* '''Logout link''' – a user will be redirected to this link after pressing "Logout"
* '''Change ANSWER to FAILED if HGC not equal to 16 for Users''' – changes the Hangupcause for users in call reports from Answered to Failed if Hangupcause is not 16.
* '''Inform when Devices are created or updated''' – check to get email about every newly created or updated Device.
* '''Archive Calls older than''' – how old Calls should be archived?
* '''Archive only ANSWERED calls''' – check to archive just answered Calls.
* '''Archive Calls every day at''' – when Calls should be archive process started?
* '''Stop Archiving Calls at''' – when Calls should be archive process stopped ?
* '''Show Usernames on PDF/CSV Export Files in Last Calls''' – if checked, when particular User in Last Calls page search is selected, Last Calls PDF/CSV file name will be composed also from Name of selected User. If you are looking for All Users, no User Name will be used. (Name of User will be shown in Name of CSV/PDF file).
* '''Do not allow PREPAID User balance drop below zero (block them instead)'''

* '''Do not block POSTPAID User when balance goes bellow zero on subscription count'''
* '''Allow identical email addresses to different Users''' – allows to create Users or Register users with identical email addresses. After checking this setting you will be not able to check '''Show forgot password''', because this functionality require unique email in all system. Setting can be unchecked if all emails are unique in system.
* '''Allow same IP and Port combinations in multiple Devices''' – allows to create Devices with the same IP and Port combination.
* '''Server free space limit''' – indicator when system should inform admin about low [[Multi_Server_support|server]] space. By default 20%. Notice will appear in [[Integrity Check]] and every hour will be send an email "Server low free space" template which can be found and edited in [[Emails]] page. Default email template for low disk space: "Some of your Servers exceeded free space limit, please investigate."
 

== Tax ==

'''SETTINGS –> Setup –> Settings –> Tax'''

Settings are explained here: [[Multi-Tax system]]
 
== Backups ==

'''SETTINGS –> Setup –> Settings –> Backups'''

Configuration is explained here: [[Backup system]]
 
== API ==

'''SETTINGS –> Setup –> Settings –> API'''

* '''Allow API''' – enable/disable API.
* '''Allow GET API''' – enable/disable GET in API - use only for testing!
* '''API Secret Key''' – secret key for API. This will be used as a password for signing queries to MOR API, check [[MOR_API | MOR API documentation]] for more information
* '''XML API Extension''' – .NET does not follow standards (surprise!); enable this setting if you are using it to get XML starting with HTTP tags.
* '''Allow API Login Redirect''' – enable/disable
* '''API allow registration''' – enable/disable registration through API
* '''API allow Payments''' – enable/disable payments through API
* '''API Payment confirmation''' – enable/disable payments confirmation through API
* '''[[Balance_in_phone|Allow devices to check balance over HTTP]]''' – enable/disable device balance checking
* '''[[MOR_API_rate_get|Allow devices to check rate over HTTP]]''' – enable/disable device rate checking
* '''Disable hash checking''' – lets you skip hash parameter using API
* '''Uniquehash''' – shows your uniquehash
 
===See also===

[[MOR API]]
 

== Functionality ==

'''SETTINGS –> Setup –> Settings –> Functionality'''
 
===FAX===

* '''Fax device enabled''' – enable/disable Fax device in MOR. When admin enables FAX, Functionality tab appears in Reseller's GUI configuration. He can then enable or disabled fax for his own users himself.
* '''Fax2Email From Sender''' – there is a field "From:" in an email form. When MOR sends Fax to Email, "Fax2Email From Sender" value is displayed in "From:" field.
 

===CSV===

* '''CSV Column Separator''' – how columns are separated in imported/exported CSV files.
* '''CSV Decimal Separator''' – how decimal parts in numbers are separated in imported/exported CSV files.
* ''' Archive CSV file when size reaches''' – if CSV file reaches a certain size, the system will archive it to save time to download it (0 means to archive always)
 
===Active Calls===

* '''How many maximum [[Active_Calls|Active Calls]] to show in GUI''' – how many [[Active_Calls| Active Calls]] to show in GUI
* '''[[Active_Calls| Active Calls]] refresh interval''' – [[Active_Calls|Active Calls]] refresh interval in seconds
* '''Show [[Active_Calls|Active Calls]] for Users''' – tick if you want to allow users to see [[Active_Calls|Active Calls]]
* '''Show Servers in [[Active_Calls|Active Calls]]'''
* '''Show DIDs in [[Active_Calls|Active Calls]]'''
 

===Tariffs/Rates===

* '''Show rates for users''' – allows users to see rates
* '''Show rates without Tax''' – shows rates without TAX
* '''Show [[Advanced_Rates|Advanced Rates]] for users''' – allows users to see [[Advanced_Rates|Advanced Rates]]
* '''Show zero rates in LCR tariff export:''' – shows zero rates in LCR tariff exported file.
NOTE: if [[Custom Rates]] and/or Tariff Rates are set for the user, only the custom rates will be shown in the details of destination rates.
 

==Privacy==

'''SETTINGS –> Setup –> Settings –> Privacy'''

Settings are explained here: [[Privacy_to_hide_Destination_ends|Privacy to hide Destination ends]]
 
== Visual ==

'''SETTINGS –> Setup –> Settings –> Visual'''

* '''Items per page''' – the number of items to display per one page
* '''[[Global_date_format| Default date format]]''' – System default date format
* '''Default Time Format''' – System default time format
* '''Number digits''' – how many digits to display after a comma
* '''Hide Quick Stats in Start page''' – hide Quick stats from start page
* '''Hide device passwords for users''' – do not display device passwords for users
* '''Hide HELP banner''' – hide HELP banner in GUI
* '''Hide "I want to" help links'''
* '''[[Remove_Manual_link_from_admin_GUI|Hide "Manual" link]]''' – hides "Manual" link
* '''Show logo on first page''' - untick to hide logo on the start page
* '''Show Full Source (CallerID Name and Number)'''
* '''Hide payment options for postpaid users''' - will hide the payment option for all postpaid users (including Resellers if they are postpaid).
* '''Hide non-completed payments for user''' – hides non-completed payments (the ones which are started in MOR, but are not completed in a payment gateway website)
* '''Show only main page''' – this is a very advanced feature for those who want to hide default MOR menu tree on the left so that they could build their own menu using advanced techniques without modifying MOR GUI sources.
* '''Show forgot password''' – turns on '''[[MOR_forgot_user_password | Forgot password]]''' functionality.
* '''Hide Recordings for all users''' – hides recordings for all system users from menu and device settings.
* '''Show recordings with zero billsec''' – shows recordings with zero billsec
* '''[[Last_Calls#Faster_Page_Loading|Count Last Calls total values only after click]]''' – select to load Total Values lines separately in order to dramatically improve Last calls and Old calls loading time.
* '''Default Number Decimal''' – with this option you can set default number decimal for all GUI
* '''Show user_billsec for Users instead of billsec''' – shows user_billsec instead of billsec in Users invoices and last calls.

* '''Show Calls statistics to User for last''' – number of Calls statistics to show in days. For example, if you enter 5 days, users will only see 5 days old calls, they will not be able to check older calls.
* '''Show Device and it's CallerID in Last Calls''' – shows additional column in [[Last Calls]] page with Device and its CallerID information. Click to reach that Device settings page.
 

== Server load ==

'''SETTINGS –> Setup –> Settings –> Server load'''
 
Settings for GUI Server and Database Server

* '''HDD utilisation higher than''' – Percentage of CPU time during which I/O requests were issued to the device (bandwidth utilization for the device).
* '''CPU General Load higher than''' – CPU General load higher than – Percentage of the CPU for user processes.
* '''CPU Load Stats 1 higher than''' – average load per one minute on the system
* '''CPU MySQL process higher than''' – Percentage of the CPU for MySQL process
* '''CPU Ruby process higher than''' – Percentage of the CPU for Ruby process
* '''CPU Asterisk process (if present) higher than''' – Percentage of the CPU for Asterisk process
 
* '''Delete Server Load Stats older than''' – deletes internal servers load statistics older than defined number of days.
 
=See also=

* [[Server is overloaded]]

Providers

2015-07-07T15:01:29Z

Rita: /* Advanced */

= Description =

In the old PSTN world, Providers were called Trunks. Providers are your route out of your VoIP network to the outside world.

Providers can be one of several types: ZAP, SIP, IAX2, and H323. The type determines which technology is used to connect to the Provider.

At the very start, you need to create a Provider through which you will dial out to PSTN or other VoIP networks.

Every Provider will charge you for calling a destination through his service (although the rate might be zero). The rate to every destination is different. Because of this, we need to have a Tariff (or Price List) for each Provider. This Tariff will tell our billing what price we will pay to a destination when using this Provider's services. In other words, this will be our Selfcost.
 
=Usage=

Go to '''SETTINGS –> Billing –> Providers'''.
 
[[File:providers_path.png]]
 
Here you can see a list of all Providers:
 
[[File:providers_list.png]]
 
* Click [[image:add.png]] '''New Provider''' to create new Provider, enter the Name for a new Provider, select its Technology, Tariff, Asterisk server which should be used for this provider and click Create.
* Click [[image:icon_edit.png]] icon to change settings for old Provider.

* Click [[Image:icon_check.png]] icon to disable Provider and it will not be used while it is cross in Active column. It affects only outgoing calls. Incoming calls from this provider will be accepted.
* Click [[Image:icon_cross.png]] icon to enable Provider if you want to use it again.
NOTE:
* You can't create a new Provider if there are no Tariffs available for Providers.
* A Provider is also able to send calls, not just receive them. More info [[Configure Provider which can make calls | here]]. In case you want to sell services to a provider you have to create such a provider.
 
== Settings ==

=== General ===
 
[[File:Prov addon.png]]
 
* '''Name''' – a name for the provider, for informational purposes.
* '''Technology''' – out of SIP/IAX2/ZAP/H323, choose which technology your provider uses - that is, the way you connect to the provider.
* '''Tariff''' – the list of rates the provider charges you.
* '''DTMF Mode''' – the available options are inband, info, RFC2833, and auto. Choose the one used by your provider.
* '''Location''' – which Localization rules are set to apply to INCOMING calls COMING FROM this provider.
* '''Automatic Number Identification''' - used in a special case explained here: [[Provider with ANI]]
* '''Ringing Timeout''' – this setting lets you set for how long this provider should be dialed before giving up. The default value is 60 seconds, and the minimum value is 30 seconds.
* '''Call Timeout''' – this setting lets you set the duration of the whole call: ringing and the period after connection.
* '''Device ID''' – informational data.
* '''Call limit''' – explained here: [[Simultaneous call limitation]].
* '''Hidden?''' – [[Hide_provider|hide/unhide provider]]

* '''Balance''' – provider balance. Provider balance increases when calls are made via that provider. Currently it is not available to decrease provider's balance. You can read more about [[Provider Billing Addon|Provider Billing Addon here]].

* '''Balance limit''' – this is the limit of balance that can be reached by the provider. After he reaches the specified balance, calls aren't sent through this provider anymore. Balance limit value of zero means that balance is unlimited.
 

=== Authentication ===

==== For ZAP providers ====

* '''Channel''' – which channel (or channel group) to use on a PRI/BRI/PSTN card. Channels and groups should be configured in zapata.conf.
 
==== For SIP/IAX2 providers ====

* '''Login''' – username if your provider asks for it.
* '''Password''' – password used for authentication by your provider.
* '''Authenticate by IP''' – just check the checkbox (Login/Password can be left empty).
** Make sure you put the correct Hostname/[[IP address]]!
* '''Register''' – should we register the provider? (option is not visible when '''Dynamic''' is selected in '''Network related''' section)
* '''Register extension''' – if the provider asks to use an extension for registration (usually they don't), enter it here.
 

=== Provider CallerID ===

'''CallerID''' – consists of two parts: Name and Number. The "Number" part is transferred by default on all technologies (GSM, PSTN, SIP etc), but the "Name" part only on some. The number you see on your mobile phone when someone is calling you is the "Number" part.
 
[[File:provider_callerid.png]]
 
* '''Name''' – the "Name" part of CallerID.
* '''Number''' – the "Number" part of CallerID.
* '''Random Number from Number Pool''' – the "Number" part of CallerID selected from [[Number Pool]]
.
'''NOTE:''' If you have this configured all incoming call from this provider will have that CallerID.
If you leave these fields empty, calls coming from this provider will have CallerID set by the Provider. It should almost always be this way.
 

=== Network related ===

* '''Hostname''' – hostname of the provider. Subnetting is supported.
* '''Server IP''' – the provider's IP. The value can be "dynamic", which means that provider can change its IP. To discover this value, do ping to the provider's hostname.
* '''Port''' – port used to connect to the provider (default: 5060 for SIP, 4569 for IAX2, 1720 for H323).
 
====H323 specific network settings====

* '''Fast Start''' – an option specific to H323; it is either on or off.
* '''h245 Tunneling''' – an option specific to H323; it is either on or of
 
====SIP specific network settings====

* '''Media control''' – canreinvite/transfer. Do you want your server to stay in the media path between your clients and your provider? Disable if you have a lot of 1-second answered calls.
* '''NAT''' – the available options are: yes, no, never, and route. For a detailed explanation of these settings, refer to [http://www.voip-info.org/wiki-Asterisk+sip+nat here].
 
====IAX2 specific network settings====

* '''IAX2 Trunking Mode''' – enable/disable trunking mode, which allows multiple voice streams to share a single "trunk" to another server, reducing overhead created by IP packets. More details [http://wiki.kolmisoft.com/index.php/Trunks here]
 
====SIP and IAX2 specific network settings====

* '''Qualify''' – how often to "ask" the provider about availability. More details [http://www.voip-info.org/wiki/view/Asterisk+sip+qualify here] for SIP providers and [http://www.voip-info.org/wiki/view/Asterisk+iax+qualify here] for IAX2 providers.
 

=== Servers ===
Allows to select on which Asterisk servers this provider should be used.
 
=== Codecs ===

Choose the codecs your provider uses.

NOTE: When no fields are checked, all codecs are available. ([[Image:asterisk_icon.png]] For example, settings in sip.conf or iax.conf are effective).
 
=== Advanced ===

* '''Grace time''' – if call duration is less than Grace Time, it will not be accounted.
* Progressinband:
**yes – when "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio.
**no – send 180 Ringing if 183 has not yet been sent, establishing an audio path. If the audio path is established already (with 183), then send in-band ringing (this is the way Asterisk historically behaved because of buggy phones like Polycom's).
**never – whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behavior of Asterisk.
* '''Video support''' – does your provider support Video over IP? More info [http://www.voip-info.org/wiki/view/Asterisk+video here].
* '''[[Duplicate call prevention | Allow duplicate calls]]''' – the default setting is "no".
* '''Interpret NO ANSWER as FAILED''' – default option is no (This will change the provider in LCR like it is failed, in last calls you will see
the real hangupcause – no answer)
* '''Interpret BUSY as FAILED''' – default option is no (This will change the provider in LCR like it is failed, in last calls you will see
the real hangupcause – busy)
* '''Fake Ring''' – should system generate Fake Ring or not? Use only if your Device/Provider has bad protocol implementation and does not send RING signal properly.
* '''Common Use''' – Provider can be used by Resellers which are able to create their own Providers. Reseller will be billed using this Provider. If provider is marked as common use and is used in one of reseller's LCR or is in Terminators list - admin is not allowed to delete this provider or remove common use option.
* '''Anti-resale (Auto-answer)''' – when option is set to YES, MOR answers the call before sending it to provider in order to generate FAS. Work only when provider is the originator of the call. This option does not affect the billing.
* '''Transport''' – lets you choose protocol <nowiki>[tcp, udp]</nowiki> for data transfer. Appears only if Provider is SIP and when Asterisk 1.8 is enabled. Default value: ''udp''. If TCP is used, it has to be [[How_to_enable_TCP_for_Asterisk |enabled in Asterisk]].
* '''Time limit per day''' - Total time available for provider per day. When limit is reached another provider in LCR is used (total time is calculated after rounding rules and minimal time adjustment, so if your provider tariff has increment higher than 1 or minimal time set, then total provider time may be different than actual call time).
* '''Limit up to X, during Y seconds''' – you can set limit how many calls (X) can be done in some interval of seconds (Y). [[Last Calls]] will show '''FAILED([[Hangupcause Codes |243]])'''.
* '''Enforce same codec as on Leg A''' - this option takes effect only when Device/Provider is used as Terminator. When this option is enabled, system checks what codec is used on Leg A. Then if Leg A codec matches one of codec enabled on Codecs list, system enforces callee to use that codec, by offering only that single codec in SDP data.
 

=== SIP Specific ===

* '''Fromuser''' – [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+fromuser more details]
* '''Fromdomain''' – [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+fromuser more details]
* '''Trustrpid''' – This defines whether or not Remote-Party-ID is trusted. It's defined in http://tools.ietf.org/id/draft-ietf-sip-privacy-04.txt
* '''Sendrpid''' – defines whether a Remote-Party-ID SIP header should be sent. It defaults to "No". This field is often used by wholesale VoIP providers to provide calling party identity regardless of the privacy settings (the From SIP header).
* '''Insecure''' – [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure more details]
* '''T.38 Support''' – should we support T.38 pass-through?
* '''[[SIP 302 Reinvite Support]]''' – turns this SIP feature on or off.
* '''P-Asserted-Identity usage''' – The P-Asserted-Identity header field can be used to convey the proven identity of the originator of a request within a trusted network.
* '''Periodic check''' - Turns on or off functionality which disables provider if it is unreachable.
 

===Debug===

* '''Save Call Log''' – when this option is enabled you can access Call info information in last calls page.
 

=See also=

* [[H323 Provider settings]]
* [[Quick Providers changes]]
* [[Providers Statistics]]
* [[NO ANSWER/BUSY interpretation for providers]]
* [[I have a problem with Provider]]

Device settings

2015-07-07T14:59:19Z

Rita: /* Advanced */

<mkmeta>MOR Device Settings Explained</mkmeta>
=General=

* '''[[Registration Status]]''' - available only for SIP devices. If device is assigned to server A but registers to server B - status will not be shown (devices must register to the server they are assigned to). If no status is displayed - device has not tried to register or noone called to that device.
* '''Accountcode''' – the unique ID of the device in the system.
* '''Description''' – for informational purposes.
* '''PIN''' – device PIN for authentication/authorization.
* '''Device group''' – to which group the device belongs (not used most of the time).
* '''Type''' – what type of device it is. Device type '''cannot''' be changed once it is created.
* '''Extension''' – a short number by which this device can be reached; must be unique in the system.
* '''DTMF Mode''' – in which protocol phone button presses are sent over network. The available options are: inband, info, RFC2833, and auto. Choose which one your provider uses (RFC2833 is used most often). This setting applies to SIP, H.323 and IAX2 protocols. For ZAP devices edit the configuration files in /etc/asterisk.
* '''Location''' – the default is Global. Choose the appropriate location based on [[Localization]] needs.
* '''Ringing Timeout''' – allows to limit the ringing duration in seconds. Leave 0 for unlimited. Minimal value is 10 seconds.
* '''Call Timeout''' – allows to limit answered call billsec. Leave 0 for unlimited.
* '''[[Trunks | Trunk]]''' – No/Yes/Yes with ANI – Is this device Trunk with/without ANI?
* '''[[Simultaneous call limitation | Call Limit]]''' – how many simultaneous calls can a user make?
* '''[[Multi_Server_support | Server]]''' – allows to choose server in multiple servers system. (While Carrier Class Addon is active, all Virtual and Fax devices are auto-assigned to all present Asterisk servers and cannot be re-assigned. Disabling Carrier Class Addon all Virtual and Fax devices will be assigned to default Asterisk server chosen from Addons > Carrier Class > Settings dropdown).
 

=Authentication=

===For dahdi devices:===

* '''Channel''' – which channel (or channel group) to use on PRI/BRI/PSTN card. Channels and groups should be configured in ''zapata.conf''.
 
===For SIP, H323 and IAX2 Devices:===

'''IP Authentication:'''

* '''IP Address''' – Device IP address in one of the following formats:
** '''IP''' – understands simple IP address, IP address with it's Subnet Mask and IP address with it's Range. For example ''192.168.0.1'', ''192.168.0.1/24'' and ''192.168.0.0-255''.
* '''Accept calls from any port''' – this option lets you receive calls from different Ports. This option is just on SIP (when [[SIP balancer]] in use) and H323.
* '''Port''' – Device Port.

'''Dynamic:'''

* '''Username''' – username you enter in your Device.
* '''Password''' – password you enter in your Device.
* '''IP Address''' – shows Device IP address.
* '''Port''' – shows Device Port.
 

=CallerID=
 
[[File:device_callerid.png]]
 
A CallerID consists of two parts: Name and Number. The "Number" part is transferred by default by all technologies (GSM, PSTN, SIP etc), but the "Name" part is transferred only by some. The number you see on your mobile phone when someone is calling you is the "Number" part.

* '''Name''' – the "Name" part of the CallerID.
* '''Number''' – the "Number" part of the CallerID. Only numerical values can be entered in this field.

NOTE: if you leave these fields empty, the user can set the CallerID name by himself. Usually it is not advisable to allow the user do it on his own. If CallerID number field is empty, system will take device username as CallerID number (if device is username/password authenticated).

* '''Number from DID''' - use the DID as the CallerID (only available when the device has DIDs), this option just sets CallerID Number to be equal to DID. Next time you will edit Device's settings, first option - Number will be checked

* '''Control by DIDs (CID from DIDs)''' – only available when the device has DIDs.

This setting lets you control what a CID user can enter in his devices. CID numbers should be from the set of the device's DIDs. They are mainly used when the user's PBX is connected over Trunk and many DIDs are routed to this Trunk. Calls coming out from this PBX must have a CallerID Number from the set of DIDs assigned to this Trunk (PBX). If "CID from DIDs" is checked, the system checks whether the device's CID number is from DIDs assigned to this device. If no DIDs are assigned, this value is ignored. In order for this setting to be active, CID Name/Number fields must be empty to allow the user to enter any CID number he wants. If "CID from DIDs" is checked and user enters any CID (not from his DIDs), the system will change his CID by his first available DID (selected from database by lowest ID).

* '''Control by CIDs''' – system checks incoming CallerID (Device's CallerID which is set on the Device). If CallerID matches one of the Device's CIDs - then such CallerID will be allowed. If it does not match - it will be changed to the selected CID.

* '''Control by Destination''' – This is advanced option which works by such algorithm:

# system checks Destination and checks DIDs assigned to calling Device. Then it finds 'nearest' DID to the Destination by subtracting DID from Destination in numerical form (Example, DID: 11111, Destination: 11112, |DID-Destination|=|1111-11112|=1) If Device has several DIDs - the lowest value is selected and this DID is set as CallerID.
#If Device has no DIDs - then User DIDs are used.
#If Device does not have DIDs and User has no DIDs, then CallerID set in Number field will be used.
#If nothing is set here - CallerID coming from Device will be used.

* '''Random Number from Number Pool''' – This functionality gives you an ability to send different caller number on each Call. Available from MOR X4. More information: [http://wiki.kolmisoft.com/index.php/Number_Pool Number Pool].

* '''Unknown''' - This option lets you set CallerID number to unknown.

* '''Copy Leg A Name to Leg B Number''' – Leg A name will be used as Leg B number.
 

=Network Related=

===For H323:===

* '''Fast start''' – option for faster H.245.
* '''h245 Tunneling''' – use H.245 without opening a second TCP/IP channel.
More info you can find [http://toncar.cz/Tutorials/VoIP/VoIP_Protocols_H323_Call_Signalling_Optimizations.html here].
 
===For SIP and IAX2===

* '''Media control''' – can reinvite. Do you want your server to stay in media path between your clients and your provider? Please note that this option will work only if it is supported from both, device and provider, sides and it allows to handle SIP packets only. Moreover, no codec mismatch between device and provider is available. Default value is ''No''. '''WARNING''': This option can cause [[Call was billed incorrectly|incorrect billing]].
** canreinvite = yes "allow RTP media direct"
** canreinvite = no "deny re-invites"
** canreinvite = nonat "allow reinvite when local, deny reinvite when NAT"
** canreinvite = update "use UPDATE instead of INVITE"
** canreinvite = update,nonat "use UPDATE when local, deny when NAT"
* '''NAT''' – the available options are: yes, no, never, force_rport, comedia. For a detailed explanation of these settings, refer to [http://www.voip-info.org/wiki/view/Asterisk+sip+nat here].
* '''Qualify''' – how often to "ask" the device about availability. More details [http://www.voip-info.org/wiki/view/Asterisk+sip+qualify here] for SIP devices and [http://www.voip-info.org/wiki/view/Asterisk+iax+qualify here] for IAX2 devices.
* '''IAX2 Trunking Mode''' – enable/disable trunking mode, which allows multiple voice streams to share a single "trunk" to another server, reducing overhead created by IP packets. Only on IAX2.
 

=Groups=

* [[What are callgroups and pickupgroups | Call Group]] – to which Call Group this device belongs.
* [[What are callgroups and pickupgroups | Pickup Group]] – which Call Groups this device can pick up.
 
=Voicemail=

* '''Email''' – where to send received Voicemail.
* '''Password''' – the digital password the user enters when he calls the Voicemail number to hear his messages.
 
=Codecs=

[[Which codecs should I choose for devices | Choose the codecs]] your provider uses.

NOTES:
* When no fields are checked, all codecs are available – for example, settings in sip.conf or iax.conf are effective.
* If the Provider and the Device do not have similar codecs, no call can be established.
* '''Enforce same codec as on Leg A''' - this option takes effect only when Device/Provider is used as Terminator. When this option is enabled, system checks what codec is used on Leg A. Then if Leg A codec matches one of codec enabled on Codecs list, system enforces callee to use that codec, by offering only that single codec in SDP data.
 

=Allowed Addresses=

'''IP, MASK''' – permits traffic from this IP only. You can find a detailed explanation [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+permit-deny-mask here].

You can enable multiple allowed addresses however this option should be turned on to be visible in main device settings window. Go to '''SETTINGS''' -> '''Setup''' -> '''Settings''' -> '''Various''' tab and choose ''Yes'' for ''Enable "Allowed addresses" option for Devices:''.
 
[[File:Settings visual advanced.png]]
 
And click ''Save changes''.

Now you will see an additional section in SIP and IAX2 Devices edit window:
 
[[File:Deviceconfig allowedaddr.png]]
 
'''Allowed addresses''' describes from what IP's or defined subnets accept traffic.
NOTE that this setting take effect only for '''Dynamic authentication''' devices. For '''IP authentication''' devices, this setting is not applied.

If you do not clearly understand what these settings are used for, leave them with default values.
 

=Advanced=

* '''Fromuser/Fromdomain''' – used when calling TO this peer FROM Asterisk. If you're using _register=>_ with another SIP proxy, this setting can come in handy since some SIP networks only allow users in the right domain with the correct user name.
* '''Trustrpid''' – defines whether or not Remote-Party-ID is trusted. It's defined in http://tools.ietf.org/id/draft-ietf-sip-privacy-04.txt.
* '''Sendrpid''' – defines whether a Remote-Party-ID SIP header should be sent. The default setting is "no". This field is often used by wholesale VoIP providers to provide calling party identity regardless of the privacy settings (the From SIP header).
* [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure Insecure]
** port: ignore the port number where request came from. '''WARNING:''' do NOT enable it on username/password authenticated devices.
** invite: don't require authentication of incoming INVITEs. '''WARNING:''' do NOT enable it on username/password authenticated devices.
** port, invite: don't require initial INVITE to authenticate and ignore the port where the request came from. '''WARNING:''' do NOT enable it on username/password authenticated devices.

Both of these settings are enabled by default for [[Authentication | IP Authenticated]] devices, but they can be changed, unless you are using Carrier Class Addon, which forces the device to ignore the port number and the authentication from an incoming request.

* '''[[SIP 302 Redirect Support | SIP 302 Redirect support]]''' – SIP Redirect support for the device. Disabled by default.
* '''Custom SIP Header''' - adds custom header to SIP request. Format is '''header: value''' (for example '''x-My-Custom-Header: my value''')
* '''[http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband Progressinband]''':
** yes – when "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio.
** no – send 180 Ringing if 183 has not yet been sent, establishing an audio path. If the audio path is established already (with 183), then send in-band ringing (this is the way Asterisk historically behaved because of buggy phones like Polycom's).
** never – whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behavior of Asterisk.
NOTE: if Progressinband does not work, add "prematuremedia=no" to sip.conf and reload Asterisk.
* '''Video support''' – does your provider support Video over IP? More info [http://www.voip-info.org/wiki/view/Asterisk+video here].
* '''[[Duplicate call prevention | Allow duplicate calls]]''' – the default setting is "no".
* '''Language''' – sets IVR language
* '''Use ANI (if available) for CallerID:''' – When the call comes the information about who is calling is found in CallerID field. This field is used to determine who is calling. When the call comes through PRI/SS7 channels - then additional information is available who is calling in ANI field (in call's data channel) so sometimes CallerID might be empty or anonymous, but the caller can be found in ANI field. This option allows to use ANI field as CallerID to determine and recognize who is calling.
* '''Incoming Call CallerID Presentation''' – sets CallerID Presentation. Please note that this setting applies only for incoming calls. More information can be found [http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerPres here] and [http://www.voip-info.org/wiki/view/Asterisk+cmd+CallingPres here]
* '''Change Failed Code To''' – if call fails change Hangup Cause Code to this value. This works only for outgoing calls from device. '''Not for Incoming'''
* '''Forward DID''' – it allows to forward call to DID which is assigned to Authorization by PIN or Calling Cards dial plan. After user enters PIN of any device or card, call gets connected with destination.
* '''Anti-resale (Auto-answer)''' – when option is set to YES, MOR answers the call before sending it to provider in order to generate FAS. This does not affect billing in MOR (User is not billed for extra time).
* '''Transport''' – lets you choose protocol(s) for data transfer. Appears only if device is SIP and when Asterisk 1.8 is enabled. Default value: ''udp''. If TCP is used, it has to be [[How_to_enable_TCP_for_Asterisk |enabled in Asterisk]].
* '''T.38 Support''' – should T.38 pass-through be supported
* '''SRTP Encryption''' – should SRTP protocol be used for calls
* '''Block callerid if (number) simultaneous calls come from it''' – blocks CallerID if the entered number of simultaneous calls come from it

* '''Limit up to (number) calls, during (number) seconds''' – allows to set calls per second limit in some period. [[File:call_limit_per_period.png]]
* '''Outbound Proxy''' – send outbound signaling to this proxy, not directly to the peer (Internal Asterisk option).
 

=Tell Options=

* '''Tell rate before call''' - should MOR tell announce minute price (rounded to cents) before every call. Default value is "no".
* '''Tell balance before call''' – should MOR tell the user his balance every time he tries to dial? The default setting is "no".
* '''Tell time''' - should MOR tell the user his remaining time every time he tries to dial? The default setting is "no".
** Time is told in MINUTES only. Currently it is not possible to tell both in minutes and seconds.
* '''Tell remaining time when left''' – when some time is left, MOR will tell the remaining time to talk (in seconds).
* '''Repeat remaining time when left''' – repeats the remaining time when some time is left (in seconds).
 
=Debug=

* [[SIP debug info|'''Process SIPCHANINFO''']] - shows SIP channel info in [[Asterisk CLI]] and saves this information on database.
* '''Save Call Log''' - saves whole call log on database. Log can be overviewed by click [[File:Icon_callinfo.png]] in [[Last Calls]].
NOTE: debug should be enabled only if you are experiencing any problems. It should be disable in any other cases, because it stores lot of information on database.
 
This option is DISABLED globally by default. If you want it to be enabled, please contact Kolmisoft staff.
 

=Recordings=

This section is available when [[Recordings Addon]] is installed in the system.
 
=Comment=

Devices can have Comments, which gives information about the Device.
 
=See also=

* [[Grace_time | Grace time]]
* [[PAP_device_configuration | PAP device configuration]]
* [[H323 Device settings]]

I HAVE A PROBLEM

2015-07-07T08:09:37Z

Rita:

Before proceeding, you need to update MOR GUI to the newest version because many problems can be fixed just by updating GUI.

* [[Update MOR GUI]]

After the update, check your problem. If it still exists - continue.

 
----
 

With '''WHAT''' do you have a problem?

 

I have a problem with:
* [[I have a problem with Auto Dialer| Auto Dialer]]
* [[I have a problem with Accountant| Accountant]]
* [[I have a problem with API| API]]
* [[I have a problem with Asterisk | Asterisk]]
* [[I have a problem with Authorization by PIN | Authorization by PIN]]
* [[I have a problem with Backups | Backups]]
* [[I have a problem with Balance | Balance]]
* [[I have a problem with Callback | Callback]]
* [[I have a problem with Calling Cards | Calling Cards]]
* [[I have a problem with CallerID | CallerID]]
* <big><big><big>[[I have a problem with Calls | Calls]]</big></big></big>
* [[I have a problem with CallFlow | Callflow]]
* [[I have a problem with Codecs | Codecs]]
* [[I have a problem with Currencies | Currencies]]
* [[I have a problem with CSV | CSV]]
* [[I have a problem with Devices | Devices]]
* [[I have a problem with DIDs | DIDs]]
* [[I have a problem with DTMF | DTMF]]
* [[I have a problem with Email | Email]]
* [[I have a problem with Fax | Fax]]
* [[I have a problem with Google Maps| Google Maps]]
* [[I have a problem with GUI | GUI]]
* [[I have a problem with HGC Codes| HGC Codes]]
* [[I have a problem with Invoices | Invoices]]
* [[I have a problem with IVR | IVR]]
* [[I have a problem with Languages | Languages]]
* [[I have a problem with MOR FREE| MOR FREE]]
* [[I have a problem with MySQL| MySQL]]
* [[I have a problem with Network| Network]]
* [[I have a problem with Payment Gateway | Payment Gateway]] (PayPal, WebMoney and similar)
* [[I have a problem with Payments | Payments]]
* [[I have a problem with PAI | PAI]]
* [[I have a problem with PBX Functions | PBX functions]]
* [[I have a problem with Provider| Provider]]
* [[I_have_a_problem_with_Queues | Queues ]]
* [[I have a problem with Recordings | Recordings]]
* [[I have a problem with Registration | Registration]]
* [[I have a problem with Ring Group | Ring Group]]
* [[I have a problem with Security| Security]]
* [[I have a problem with Server| Server]]
** [[I have a problem with Hard Drive | Hard Drive (HDD/SSD)]]
* [[I have a problem with SIP TLS | SIP TLS]]
* [[I have a problem with SMS | SMS]]
* [[I have a problem with Statistics| Statistics]]
* [[I have a problem with Subscriptions| Subscriptions]]
* [[I have a problem with SVN | SVN]]
* [[I have a problem with Rates | Tariffs/Rates]]
* [[I have a problem with Taxes | Taxes]]
* [[I have a problem with Time | Time]]
* [[I have a problem with Voicemail| Voicemail]]
* [[I have a problem with Voucher | Vouchers]]
* [[I have a problem with RFC | RFC]]
* [[I have a problem with MNP | MNP]]
* [[I have a problem with SIP Proxy | SIP Proxy/Balancer]]
* [[I have a problem with Call Flood | Active calls flood (too many connecting calls)]]
* [[Other|Other]]

I have a problem with Queues

2015-07-07T08:08:19Z

Rita: Created page with '* Caller is not able to join the queue'

* [[Caller_is_not_able_to_join_the_queue | Caller is not able to join the queue]]

Caller is not able to join the queue

2015-07-07T08:06:42Z

Rita:

If a caller is not able to join a queue and you see these messages in the Asterisk log:
 
WARNING[11553]: app_queue.c:6323 queue_exec: Unable to join queue 'queue_name'
This means that you have set parameters [[Call_Queues#Caller_Options |"Join empty"]] incorrectly in the Queue settings.
 
For example, if none of the option are checked in the column "Join empty", caller will always be allowed to join the queue, even if there are no agents available.

Caller is not able to join the queue

2015-07-07T08:06:20Z

Rita: Created page with 'If a caller is not able to join a queue and you see these messages in the Asterisk log: WARNING[11553]: app_queue.c:6323 queue_exec: Unable to join queue 'queue_name' This …'

If a caller is not able to join a queue and you see these messages in the Asterisk log:
 
WARNING[11553]: app_queue.c:6323 queue_exec: Unable to join queue 'queue_name'
This means that you have set parameters [[Call_Queues#Caller_Options |"Join empty"]] incorrectly in the Queue settings.
For example, if none of the option are checked in the column "Join empty", caller will always be allowed to join the queue, even if there are no agents available.

Providers

2015-07-02T13:34:52Z

Rita: /* Advanced */

= Description =

In the old PSTN world, Providers were called Trunks. Providers are your route out of your VoIP network to the outside world.

Providers can be one of several types: ZAP, SIP, IAX2, and H323. The type determines which technology is used to connect to the Provider.

At the very start, you need to create a Provider through which you will dial out to PSTN or other VoIP networks.

Every Provider will charge you for calling a destination through his service (although the rate might be zero). The rate to every destination is different. Because of this, we need to have a Tariff (or Price List) for each Provider. This Tariff will tell our billing what price we will pay to a destination when using this Provider's services. In other words, this will be our Selfcost.
 
=Usage=

Go to '''SETTINGS –> Billing –> Providers'''.
 
[[File:providers_path.png]]
 
Here you can see a list of all Providers:
 
[[File:providers_list.png]]
 
* Click [[image:add.png]] '''New Provider''' to create new Provider, enter the Name for a new Provider, select its Technology, Tariff, Asterisk server which should be used for this provider and click Create.
* Click [[image:icon_edit.png]] icon to change settings for old Provider.

* Click [[Image:icon_check.png]] icon to disable Provider and it will not be used while it is cross in Active column. It affects only outgoing calls. Incoming calls from this provider will be accepted.
* Click [[Image:icon_cross.png]] icon to enable Provider if you want to use it again.
NOTE:
* You can't create a new Provider if there are no Tariffs available for Providers.
* A Provider is also able to send calls, not just receive them. More info [[Configure Provider which can make calls | here]]. In case you want to sell services to a provider you have to create such a provider.
 
== Settings ==

=== General ===
 
[[File:Prov addon.png]]
 
* '''Name''' – a name for the provider, for informational purposes.
* '''Technology''' – out of SIP/IAX2/ZAP/H323, choose which technology your provider uses - that is, the way you connect to the provider.
* '''Tariff''' – the list of rates the provider charges you.
* '''DTMF Mode''' – the available options are inband, info, RFC2833, and auto. Choose the one used by your provider.
* '''Location''' – which Localization rules are set to apply to INCOMING calls COMING FROM this provider.
* '''Automatic Number Identification''' - used in a special case explained here: [[Provider with ANI]]
* '''Ringing Timeout''' – this setting lets you set for how long this provider should be dialed before giving up. The default value is 60 seconds, and the minimum value is 30 seconds.
* '''Call Timeout''' – this setting lets you set the duration of the whole call: ringing and the period after connection.
* '''Device ID''' – informational data.
* '''Call limit''' – explained here: [[Simultaneous call limitation]].
* '''Hidden?''' – [[Hide_provider|hide/unhide provider]]

* '''Balance''' – provider balance. Provider balance increases when calls are made via that provider. Currently it is not available to decrease provider's balance. You can read more about [[Provider Billing Addon|Provider Billing Addon here]].

* '''Balance limit''' – this is the limit of balance that can be reached by the provider. After he reaches the specified balance, calls aren't sent through this provider anymore. Balance limit value of zero means that balance is unlimited.
 

=== Authentication ===

==== For ZAP providers ====

* '''Channel''' – which channel (or channel group) to use on a PRI/BRI/PSTN card. Channels and groups should be configured in zapata.conf.
 
==== For SIP/IAX2 providers ====

* '''Login''' – username if your provider asks for it.
* '''Password''' – password used for authentication by your provider.
* '''Authenticate by IP''' – just check the checkbox (Login/Password can be left empty).
** Make sure you put the correct Hostname/[[IP address]]!
* '''Register''' – should we register the provider? (option is not visible when '''Dynamic''' is selected in '''Network related''' section)
* '''Register extension''' – if the provider asks to use an extension for registration (usually they don't), enter it here.
 

=== Provider CallerID ===

'''CallerID''' – consists of two parts: Name and Number. The "Number" part is transferred by default on all technologies (GSM, PSTN, SIP etc), but the "Name" part only on some. The number you see on your mobile phone when someone is calling you is the "Number" part.
 
[[File:provider_callerid.png]]
 
* '''Name''' – the "Name" part of CallerID.
* '''Number''' – the "Number" part of CallerID.
* '''Random Number from Number Pool''' – the "Number" part of CallerID selected from [[Number Pool]]
.
'''NOTE:''' If you have this configured all incoming call from this provider will have that CallerID.
If you leave these fields empty, calls coming from this provider will have CallerID set by the Provider. It should almost always be this way.
 

=== Network related ===

* '''Hostname''' – hostname of the provider. Subnetting is supported.
* '''Server IP''' – the provider's IP. The value can be "dynamic", which means that provider can change its IP. To discover this value, do ping to the provider's hostname.
* '''Port''' – port used to connect to the provider (default: 5060 for SIP, 4569 for IAX2, 1720 for H323).
 
====H323 specific network settings====

* '''Fast Start''' – an option specific to H323; it is either on or off.
* '''h245 Tunneling''' – an option specific to H323; it is either on or of
 
====SIP specific network settings====

* '''Media control''' – canreinvite/transfer. Do you want your server to stay in the media path between your clients and your provider? Disable if you have a lot of 1-second answered calls.
* '''NAT''' – the available options are: yes, no, never, and route. For a detailed explanation of these settings, refer to [http://www.voip-info.org/wiki-Asterisk+sip+nat here].
 
====IAX2 specific network settings====

* '''IAX2 Trunking Mode''' – enable/disable trunking mode, which allows multiple voice streams to share a single "trunk" to another server, reducing overhead created by IP packets. More details [http://wiki.kolmisoft.com/index.php/Trunks here]
 
====SIP and IAX2 specific network settings====

* '''Qualify''' – how often to "ask" the provider about availability. More details [http://www.voip-info.org/wiki/view/Asterisk+sip+qualify here] for SIP providers and [http://www.voip-info.org/wiki/view/Asterisk+iax+qualify here] for IAX2 providers.
 

=== Servers ===
Allows to select on which Asterisk servers this provider should be used.
 
=== Codecs ===

Choose the codecs your provider uses.

NOTE: When no fields are checked, all codecs are available. ([[Image:asterisk_icon.png]] For example, settings in sip.conf or iax.conf are effective).
 
=== Advanced ===

* '''Grace time''' – if call duration is less than Grace Time, it will not be accounted.
* Progressinband:
**yes – when "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio.
**no – send 180 Ringing if 183 has not yet been sent, establishing an audio path. If the audio path is established already (with 183), then send in-band ringing (this is the way Asterisk historically behaved because of buggy phones like Polycom's).
**never – whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behavior of Asterisk.
* '''Video support''' – does your provider support Video over IP? More info [http://www.voip-info.org/wiki/view/Asterisk+video here].
* '''[[Duplicate call prevention | Allow duplicate calls]]''' – the default setting is "no".
* '''Interpret NO ANSWER as FAILED''' – default option is no (This will change the provider in LCR like it is failed, in last calls you will see
the real hangupcause – no answer)
* '''Interpret BUSY as FAILED''' – default option is no (This will change the provider in LCR like it is failed, in last calls you will see
the real hangupcause – busy)
* '''Fake Ring''' – should system generate Fake Ring or not? Use only if your Device/Provider has bad protocol implementation and does not send RING signal properly.
* '''Common Use''' – Provider can be used by Resellers which are able to create their own Providers. Reseller will be billed using this Provider. If provider is marked as common use and is used in one of reseller's LCR or is in Terminators list - admin is not allowed to delete this provider or remove common use option.
* '''Anti-resale (Auto-answer)''' – when option is set to YES, MOR answers the call before sending it to provider in order to generate FAS.
* '''Transport''' – lets you choose protocol <nowiki>[tcp, udp]</nowiki> for data transfer. Appears only if Provider is SIP and when Asterisk 1.8 is enabled. Default value: ''udp''. If TCP is used, it has to be [[How_to_enable_TCP_for_Asterisk |enabled in Asterisk]].
* '''Time limit per day''' - Total time available for provider per day. When limit is reached another provider in LCR is used (total time is calculated after rounding rules and minimal time adjustment, so if your provider tariff has increment higher than 1 or minimal time set, then total provider time may be different than actual call time).
* '''Limit up to X, during Y seconds''' – you can set limit how many calls (X) can be done in some interval of seconds (Y). [[Last Calls]] will show '''FAILED([[Hangupcause Codes |243]])'''.
* '''Enforce same codec as on Leg A''' - this option takes effect only when Device/Provider is used as Terminator. When this option is enabled, system checks what codec is used on Leg A. Then if Leg A codec matches one of codec enabled on Codecs list, system enforces callee to use that codec, by offering only that single codec in SDP data.
 

=== SIP Specific ===

* '''Fromuser''' – [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+fromuser more details]
* '''Fromdomain''' – [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+fromuser more details]
* '''Trustrpid''' – This defines whether or not Remote-Party-ID is trusted. It's defined in http://tools.ietf.org/id/draft-ietf-sip-privacy-04.txt
* '''Sendrpid''' – defines whether a Remote-Party-ID SIP header should be sent. It defaults to "No". This field is often used by wholesale VoIP providers to provide calling party identity regardless of the privacy settings (the From SIP header).
* '''Insecure''' – [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure more details]
* '''T.38 Support''' – should we support T.38 pass-through?
* '''[[SIP 302 Reinvite Support]]''' – turns this SIP feature on or off.
* '''P-Asserted-Identity usage''' – The P-Asserted-Identity header field can be used to convey the proven identity of the originator of a request within a trusted network.
* '''Periodic check''' - Turns on or off functionality which disables provider if it is unreachable.
 

===Debug===

* '''Save Call Log''' – when this option is enabled you can access Call info information in last calls page.
 

=See also=

* [[H323 Provider settings]]
* [[Quick Providers changes]]
* [[Providers Statistics]]
* [[NO ANSWER/BUSY interpretation for providers]]
* [[I have a problem with Provider]]

Device settings

2015-07-02T13:32:26Z

Rita: /* Advanced */

<mkmeta>MOR Device Settings Explained</mkmeta>
=General=

* '''[[Registration Status]]''' - available only for SIP devices. If device is assigned to server A but registers to server B - status will not be shown (devices must register to the server they are assigned to). If no status is displayed - device has not tried to register or noone called to that device.
* '''Accountcode''' – the unique ID of the device in the system.
* '''Description''' – for informational purposes.
* '''PIN''' – device PIN for authentication/authorization.
* '''Device group''' – to which group the device belongs (not used most of the time).
* '''Type''' – what type of device it is. Device type '''cannot''' be changed once it is created.
* '''Extension''' – a short number by which this device can be reached; must be unique in the system.
* '''DTMF Mode''' – in which protocol phone button presses are sent over network. The available options are: inband, info, RFC2833, and auto. Choose which one your provider uses (RFC2833 is used most often). This setting applies to SIP, H.323 and IAX2 protocols. For ZAP devices edit the configuration files in /etc/asterisk.
* '''Location''' – the default is Global. Choose the appropriate location based on [[Localization]] needs.
* '''Ringing Timeout''' – allows to limit the ringing duration in seconds. Leave 0 for unlimited. Minimal value is 10 seconds.
* '''Call Timeout''' – allows to limit answered call billsec. Leave 0 for unlimited.
* '''[[Trunks | Trunk]]''' – No/Yes/Yes with ANI – Is this device Trunk with/without ANI?
* '''[[Simultaneous call limitation | Call Limit]]''' – how many simultaneous calls can a user make?
* '''[[Multi_Server_support | Server]]''' – allows to choose server in multiple servers system. (While Carrier Class Addon is active, all Virtual and Fax devices are auto-assigned to all present Asterisk servers and cannot be re-assigned. Disabling Carrier Class Addon all Virtual and Fax devices will be assigned to default Asterisk server chosen from Addons > Carrier Class > Settings dropdown).
 

=Authentication=

===For dahdi devices:===

* '''Channel''' – which channel (or channel group) to use on PRI/BRI/PSTN card. Channels and groups should be configured in ''zapata.conf''.
 
===For SIP, H323 and IAX2 Devices:===

'''IP Authentication:'''

* '''IP Address''' – Device IP address in one of the following formats:
** '''IP''' – understands simple IP address, IP address with it's Subnet Mask and IP address with it's Range. For example ''192.168.0.1'', ''192.168.0.1/24'' and ''192.168.0.0-255''.
* '''Accept calls from any port''' – this option lets you receive calls from different Ports. This option is just on SIP (when [[SIP balancer]] in use) and H323.
* '''Port''' – Device Port.

'''Dynamic:'''

* '''Username''' – username you enter in your Device.
* '''Password''' – password you enter in your Device.
* '''IP Address''' – shows Device IP address.
* '''Port''' – shows Device Port.
 

=CallerID=
 
[[File:device_callerid.png]]
 
A CallerID consists of two parts: Name and Number. The "Number" part is transferred by default by all technologies (GSM, PSTN, SIP etc), but the "Name" part is transferred only by some. The number you see on your mobile phone when someone is calling you is the "Number" part.

* '''Name''' – the "Name" part of the CallerID.
* '''Number''' – the "Number" part of the CallerID. Only numerical values can be entered in this field.

NOTE: if you leave these fields empty, the user can set the CallerID name by himself. Usually it is not advisable to allow the user do it on his own. If CallerID number field is empty, system will take device username as CallerID number (if device is username/password authenticated).

* '''Number from DID''' - use the DID as the CallerID (only available when the device has DIDs), this option just sets CallerID Number to be equal to DID. Next time you will edit Device's settings, first option - Number will be checked

* '''Control by DIDs (CID from DIDs)''' – only available when the device has DIDs.

This setting lets you control what a CID user can enter in his devices. CID numbers should be from the set of the device's DIDs. They are mainly used when the user's PBX is connected over Trunk and many DIDs are routed to this Trunk. Calls coming out from this PBX must have a CallerID Number from the set of DIDs assigned to this Trunk (PBX). If "CID from DIDs" is checked, the system checks whether the device's CID number is from DIDs assigned to this device. If no DIDs are assigned, this value is ignored. In order for this setting to be active, CID Name/Number fields must be empty to allow the user to enter any CID number he wants. If "CID from DIDs" is checked and user enters any CID (not from his DIDs), the system will change his CID by his first available DID (selected from database by lowest ID).

* '''Control by CIDs''' – system checks incoming CallerID (Device's CallerID which is set on the Device). If CallerID matches one of the Device's CIDs - then such CallerID will be allowed. If it does not match - it will be changed to the selected CID.

* '''Control by Destination''' – This is advanced option which works by such algorithm:

# system checks Destination and checks DIDs assigned to calling Device. Then it finds 'nearest' DID to the Destination by subtracting DID from Destination in numerical form (Example, DID: 11111, Destination: 11112, |DID-Destination|=|1111-11112|=1) If Device has several DIDs - the lowest value is selected and this DID is set as CallerID.
#If Device has no DIDs - then User DIDs are used.
#If Device does not have DIDs and User has no DIDs, then CallerID set in Number field will be used.
#If nothing is set here - CallerID coming from Device will be used.

* '''Random Number from Number Pool''' – This functionality gives you an ability to send different caller number on each Call. Available from MOR X4. More information: [http://wiki.kolmisoft.com/index.php/Number_Pool Number Pool].

* '''Unknown''' - This option lets you set CallerID number to unknown.

* '''Copy Leg A Name to Leg B Number''' – Leg A name will be used as Leg B number.
 

=Network Related=

===For H323:===

* '''Fast start''' – option for faster H.245.
* '''h245 Tunneling''' – use H.245 without opening a second TCP/IP channel.
More info you can find [http://toncar.cz/Tutorials/VoIP/VoIP_Protocols_H323_Call_Signalling_Optimizations.html here].
 
===For SIP and IAX2===

* '''Media control''' – can reinvite. Do you want your server to stay in media path between your clients and your provider? Please note that this option will work only if it is supported from both, device and provider, sides and it allows to handle SIP packets only. Moreover, no codec mismatch between device and provider is available. Default value is ''No''. '''WARNING''': This option can cause [[Call was billed incorrectly|incorrect billing]].
** canreinvite = yes "allow RTP media direct"
** canreinvite = no "deny re-invites"
** canreinvite = nonat "allow reinvite when local, deny reinvite when NAT"
** canreinvite = update "use UPDATE instead of INVITE"
** canreinvite = update,nonat "use UPDATE when local, deny when NAT"
* '''NAT''' – the available options are: yes, no, never, force_rport, comedia. For a detailed explanation of these settings, refer to [http://www.voip-info.org/wiki/view/Asterisk+sip+nat here].
* '''Qualify''' – how often to "ask" the device about availability. More details [http://www.voip-info.org/wiki/view/Asterisk+sip+qualify here] for SIP devices and [http://www.voip-info.org/wiki/view/Asterisk+iax+qualify here] for IAX2 devices.
* '''IAX2 Trunking Mode''' – enable/disable trunking mode, which allows multiple voice streams to share a single "trunk" to another server, reducing overhead created by IP packets. Only on IAX2.
 

=Groups=

* [[What are callgroups and pickupgroups | Call Group]] – to which Call Group this device belongs.
* [[What are callgroups and pickupgroups | Pickup Group]] – which Call Groups this device can pick up.
 
=Voicemail=

* '''Email''' – where to send received Voicemail.
* '''Password''' – the digital password the user enters when he calls the Voicemail number to hear his messages.
 
=Codecs=

[[Which codecs should I choose for devices | Choose the codecs]] your provider uses.

NOTES:
* When no fields are checked, all codecs are available – for example, settings in sip.conf or iax.conf are effective.
* If the Provider and the Device do not have similar codecs, no call can be established.
* '''Enforce same codec as on Leg A''' - this option takes effect only when Device/Provider is used as Terminator. When this option is enabled, system checks what codec is used on Leg A. Then if Leg A codec matches one of codec enabled on Codecs list, system enforces callee to use that codec, by offering only that single codec in SDP data.
 

=Allowed Addresses=

'''IP, MASK''' – permits traffic from this IP only. You can find a detailed explanation [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+permit-deny-mask here].

You can enable multiple allowed addresses however this option should be turned on to be visible in main device settings window. Go to '''SETTINGS''' -> '''Setup''' -> '''Settings''' -> '''Various''' tab and choose ''Yes'' for ''Enable "Allowed addresses" option for Devices:''.
 
[[File:Settings visual advanced.png]]
 
And click ''Save changes''.

Now you will see an additional section in SIP and IAX2 Devices edit window:
 
[[File:Deviceconfig allowedaddr.png]]
 
'''Allowed addresses''' describes from what IP's or defined subnets accept traffic.
NOTE that this setting take effect only for '''Dynamic authentication''' devices. For '''IP authentication''' devices, this setting is not applied.

If you do not clearly understand what these settings are used for, leave them with default values.
 

=Advanced=

* '''Fromuser/Fromdomain''' – used when calling TO this peer FROM Asterisk. If you're using _register=>_ with another SIP proxy, this setting can come in handy since some SIP networks only allow users in the right domain with the correct user name.
* '''Trustrpid''' – defines whether or not Remote-Party-ID is trusted. It's defined in http://tools.ietf.org/id/draft-ietf-sip-privacy-04.txt.
* '''Sendrpid''' – defines whether a Remote-Party-ID SIP header should be sent. The default setting is "no". This field is often used by wholesale VoIP providers to provide calling party identity regardless of the privacy settings (the From SIP header).
* [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure Insecure]
** port: ignore the port number where request came from. '''WARNING:''' do NOT enable it on username/password authenticated devices.
** invite: don't require authentication of incoming INVITEs. '''WARNING:''' do NOT enable it on username/password authenticated devices.
** port, invite: don't require initial INVITE to authenticate and ignore the port where the request came from. '''WARNING:''' do NOT enable it on username/password authenticated devices.

Both of these settings are enabled by default for [[Authentication | IP Authenticated]] devices, but they can be changed, unless you are using Carrier Class Addon, which forces the device to ignore the port number and the authentication from an incoming request.

* '''[[SIP 302 Redirect Support | SIP 302 Redirect support]]''' – SIP Redirect support for the device. Disabled by default.
* '''Custom SIP Header''' - adds custom header to SIP request. Format is '''header: value''' (for example '''x-My-Custom-Header: my value''')
* '''[http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband Progressinband]''':
** yes – when "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio.
** no – send 180 Ringing if 183 has not yet been sent, establishing an audio path. If the audio path is established already (with 183), then send in-band ringing (this is the way Asterisk historically behaved because of buggy phones like Polycom's).
** never – whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behavior of Asterisk.
NOTE: if Progressinband does not work, add "prematuremedia=no" to sip.conf and reload Asterisk.
* '''Video support''' – does your provider support Video over IP? More info [http://www.voip-info.org/wiki/view/Asterisk+video here].
* '''[[Duplicate call prevention | Allow duplicate calls]]''' – the default setting is "no".
* '''Language''' – sets IVR language
* '''Use ANI (if available) for CallerID:''' – When the call comes the information about who is calling is found in CallerID field. This field is used to determine who is calling. When the call comes through PRI/SS7 channels - then additional information is available who is calling in ANI field (in call's data channel) so sometimes CallerID might be empty or anonymous, but the caller can be found in ANI field. This option allows to use ANI field as CallerID to determine and recognize who is calling.
* '''Incoming Call CallerID Presentation''' – sets CallerID Presentation. Please note that this setting applies only for incoming calls. More information can be found [http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerPres here] and [http://www.voip-info.org/wiki/view/Asterisk+cmd+CallingPres here]
* '''Change Failed Code To''' – if call fails change Hangup Cause Code to this value. This works only for outgoing calls from device. '''Not for Incoming'''
* '''Forward DID''' – it allows to forward call to DID which is assigned to Authorization by PIN or Calling Cards dial plan. After user enters PIN of any device or card, call gets connected with destination.
* '''Anti-resale (Auto-answer)''' – when option is set to YES, MOR answers the call before sending it to provider in order to generate FAS.
* '''Transport''' – lets you choose protocol(s) for data transfer. Appears only if device is SIP and when Asterisk 1.8 is enabled. Default value: ''udp''. If TCP is used, it has to be [[How_to_enable_TCP_for_Asterisk |enabled in Asterisk]].
* '''T.38 Support''' – should T.38 pass-through be supported
* '''SRTP Encryption''' – should SRTP protocol be used for calls
* '''Block callerid if (number) simultaneous calls come from it''' – blocks CallerID if the entered number of simultaneous calls come from it

* '''Limit up to (number) calls, during (number) seconds''' – allows to set calls per second limit in some period. [[File:call_limit_per_period.png]]
* '''Outbound Proxy''' – send outbound signaling to this proxy, not directly to the peer (Internal Asterisk option).
 

=Tell Options=

* '''Tell rate before call''' - should MOR tell announce minute price (rounded to cents) before every call. Default value is "no".
* '''Tell balance before call''' – should MOR tell the user his balance every time he tries to dial? The default setting is "no".
* '''Tell time''' - should MOR tell the user his remaining time every time he tries to dial? The default setting is "no".
** Time is told in MINUTES only. Currently it is not possible to tell both in minutes and seconds.
* '''Tell remaining time when left''' – when some time is left, MOR will tell the remaining time to talk (in seconds).
* '''Repeat remaining time when left''' – repeats the remaining time when some time is left (in seconds).
 
=Debug=

* [[SIP debug info|'''Process SIPCHANINFO''']] - shows SIP channel info in [[Asterisk CLI]] and saves this information on database.
* '''Save Call Log''' - saves whole call log on database. Log can be overviewed by click [[File:Icon_callinfo.png]] in [[Last Calls]].
NOTE: debug should be enabled only if you are experiencing any problems. It should be disable in any other cases, because it stores lot of information on database.
 
This option is DISABLED globally by default. If you want it to be enabled, please contact Kolmisoft staff.
 

=Recordings=

This section is available when [[Recordings Addon]] is installed in the system.
 
=Comment=

Devices can have Comments, which gives information about the Device.
 
=See also=

* [[Grace_time | Grace time]]
* [[PAP_device_configuration | PAP device configuration]]
* [[H323 Device settings]]

How to enable TCP for Asterisk

2015-07-02T13:30:19Z

Rita: Created page with 'To allow SIP TCP clients to connect to MOR, it is necessary to enable TCP first. To enable TCP, edit sip.conf at /etc/asterisk directory. Add these lines in General section:…'

To allow SIP TCP clients to connect to MOR, it is necessary to enable TCP first.
 
To enable TCP, edit sip.conf at /etc/asterisk directory. Add these lines in General section:
[general]
tcpenable=yes
tcpbindaddr=0.0.0.0
Once it is done, reload sip in asterisk CLI, or run
asterisk -rx 'sip reload'
It is important to also choose TCP as Transport in the [[Device_settings|device settings]].

MOR API sms send

2015-06-29T08:32:14Z

Rita: /* Returns */

= Description =
SMS API is used to integrate MOR with other external systems like CRM, ERP or other for SMS sending through them.

'''Note:''' SMS API can use only '''Simple Users'''.
 

= Usage =

* Call: /api/sms_send
* Methods: POST, GET(if allowed, not recomended)
 
===Parametres===

* lcr_id - LCR ID in MOR database. '''Required'''.
* u - username to login. '''Required'''.
* dst - Destination number. '''Required'''.
* src - Source number. '''Required'''.
* message - Message that you want to send. '''Required'''.
* [[MOR API hash construction | hash]] - SHA1 hash constructed using '''API_Secret_Key''' ([[MOR API hash construction | More described in Constructing hash]]). '''Required'''.

'''* Note that message must be in [http://meta.wikimedia.org/wiki/Help:URL URL Encode]'''
 
===Request===

We have lcr_id = 1, dst = 937567337911, src = 9375783123767, message = hi, API key = 123456
**We send:
HTML POST: http://<server_ip>/billing/api/sms_send?u=username&lcr_id=1&dst=937567337911&src=9375783123767&message=hi&hash=cc0e27123e27f08de2fbc68a2c8ba0b213654687

'''* Note that username and password are not included in hash'''
 

===Returns===

'''Success'''
<page>
<response>
<status>ok</status>
<message>
<message_id> message id </message_id>
<sms_status_code_tip>SMS is sent</sms_status_code_tip>
<price> price </price>
<currency> currency </currency>
</message>
</response>
</page>
 
----

'''Errors'''

* <error>There is no message or it is empty</error> – message text not found.
* <error>Wrong source</error> – source number is wrong.
* <error>Wrong destination</error> – destination number is wrong.
* <error>There is no such LCR</error> – LCR id not found.
* <error>User is not subscribed to sms service</error> – user is not [[SMS_Addon#Configuration | Subscribed]] for sms usage.
* <error>System owner does not have rate for this destination</error> - SMS status code and tip [[SMS Addon - Status codes]]
* <error>Bad login</error> – User with Username, defined in parameter '''u''', is not found.
* <error>Access Denied</error> – User is not Simple User.
* <error>Incorrect hash</error> - Hash was not correct. Check API_Secret_Key and order or params while concatenating hash_string.
* <error>Insufficient balance</error> - The User does not have enough balance to send an SMS.
 

= See also =

* [[MOR API]]

Quick Forwards

2015-06-25T11:43:48Z

Rita: /* Quick forwards based on diversion number */

=About=

Some DIDs can be assigned to Quick Forwards functionality.
When this is done, each user can enter a different number to be forwarded when he calls a DID.

The main advantage of this function '''is to use a limited number of DIDs to allow different users to call these DIDs and be forwarded to different numbers''' (which are assigned by users). Moreover, Quick Forward dial plans are often beneficial for caller (instead of dialing directly, he dials a Forward Number and then Forward Number dials a DID - both of these connections have exceptional prices, so call is often cheaper).

'''Example'''

We have several DIDs assigned to Quick Forwards functionality:

* 1111111111.
* 2222222222.

We have two users with these CallerIDs:

* User A with CallerID 8888888888.
* User B with CallerID 9999999999.

User A sets his forwards:

* From DID 1111111111 forward to 3333333333.
* From DID 2222222222 forward to 4444444444.

User B sets his forwards:

* From DID 1111111111 forward to 5555555555.
* From DID 2222222222 forward to 6666666666.

* When User A dials DID 1111111111, his CallerID (8888888888) is checked, the user is recognized, and the call is automatically forwarded to number 3333333333.
* When User A dials DID 2222222222, his CallerID (8888888888) is checked, the user is recognized, and the call is automatically forwarded to number 4444444444.

* When User B dials DID 1111111111, his CallerID (9999999999) is checked, the user is recognized, and the call is automatically forwarded to number 5555555555.
* When User B dials DID 2222222222, his CallerID (9999999999) is checked, the user is recognized, and the call is automatically forwarded to number 6666666666

'''IMPORTANT''': Correct CallerIDs should be set for users in order for them to be recognized them. Only the system administrator can do this.

* If the system doesn't recognize the caller, it asks for a PIN number of the device to be entered.
* If a user is dialing a DID that has not been assigned a Forward Number, the user is asked to enter such a number.

The user can manipulate his Quick Forwards in his menu. He needs to log in to GUI to change numbers.

'''Please note that DIDs which follow a quick forward rule can only be assigned to Quick Forwards dial plan!'''

==Quick forwards based on diversion number==


By default quick forwards are based on CallerID but you can change it to be based on user's diversion number. But diversion field can be used instead of CallerID. It comes in a SIP packet and points at the device which forwarded a call before it got into a system. for example, if you put a forward to system's DID on your cellphone and X calls you by default system will catch Xs CallerID, which is not in the system, so it will ask for PIN. Meanwhile if quick forwards are based on diversion system will see that the call belongs to you and it will act respectively.

Moreover, Default Device allows not to bill the user who calls to that DID but bill an other user instead. To see why it is useful here's another example: let's assume that calling to DID costs the same as calling to a premium rate line. When you call to such a line you pay at once. MORs owner gets payment from DIDs provider (which itself gets it from the user). Quick forwards based on diversion allows creating such services where user does not have to pay for anything, i.e. forwards for him are "free" because he pays for a DID.

To make such a dial plan go to '''SETTINGS''' -> '''Billing''' -> '''DIDs''' -> '''Dial Plans'''. Click ''New dial plan'', you have to create a Quick Forward dial plan:

[[File:Dialplans quickforawrds diversion.png]]

Select ''Use Diversion instead of CallerID'' and click [[File:icon_add.png]].
 
'''Example''': User A (CallerID 1111) calls User B (CallerID 2222). User B forwards the call to DID (which is assigned to Quickforwards in MOR).
 
If option "Use Diversion instead of CallerID" is NOT enabled, MOR checks User's A callerID. If CallerID 1111 is found in the system, the call is forwarded to the number, that User A assigned to the DID in Quickforward menu. However, in this situation, User A is usually not a user in MOR system, so the CallerID 1111 is not found and call fails.
 
If option "Use Diversion instead of CallerID" is enabled, MOR checks User's B callerID. If CallerID 2222 is found in the system, the call is forwarded to the number, that User B assigned to the DID in Quickforward menu.

== Why is the Quick Forwards window empty? ==

The window may be empty because one or more DIDs should be assigned to Quickforwards [[Dial_Plans | Dial Plan]] before calls can be forwarded.

== Tell Balance/Tell Time ==

By default calls via Quick forward DID will not play Balance and Time. If you want it to go to '''SETTINGS''' -> '''Billing''' -> '''DIDs''' -> '''Quickforwards''' -> '''Quickforward Settings'''. You will see this window:

[[file:Quickforwards_telltime.png]]

Select ''Tell Balance: '' and/or ''Tell Time: '' and click ''Change''.
 
Tell Time/Tell Balance options also can be found in device's edit window:

[[File:Devices telloptions.png]]

Here you can also select ''Global''. Value (''Yes'' or ''No'') in the brackets depends on what is selected in quickforward settings. In this example there is ''Global (No)'' to both options because in quickforward settings both ''Tell Time'' and ''Tell Balance'' are set to ''No''.

=See also=
* [[DID_Management|DID Management]]
* [[Quickforwards_rules|Quickforwards rules]]

Device settings

2015-06-12T08:53:09Z

Rita: /* Advanced */

<mkmeta>MOR Device Settings Explained</mkmeta>
=General=

* '''[[Registration Status]]''' - available only for SIP devices. If device is assigned to server A but registers to server B - status will not be shown (devices must register to the server they are assigned to). If no status is displayed - device has not tried to register or noone called to that device.
* '''Accountcode''' – the unique ID of the device in the system.
* '''Description''' – for informational purposes.
* '''PIN''' – device PIN for authentication/authorization.
* '''Device group''' – to which group the device belongs (not used most of the time).
* '''Type''' – what type of device it is. Device type '''cannot''' be changed once it is created.
* '''Extension''' – a short number by which this device can be reached; must be unique in the system.
* '''DTMF Mode''' – in which protocol phone button presses are sent over network. The available options are: inband, info, RFC2833, and auto. Choose which one your provider uses (RFC2833 is used most often). This setting applies to SIP, H.323 and IAX2 protocols. For ZAP devices edit the configuration files in /etc/asterisk.
* '''Location''' – the default is Global. Choose the appropriate location based on [[Localization]] needs.
* '''Ringing Timeout''' – allows to limit the ringing duration in seconds. Leave 0 for unlimited. Minimal value is 10 seconds.
* '''Call Timeout''' – allows to limit answered call billsec. Leave 0 for unlimited.
* '''[[Trunks | Trunk]]''' – No/Yes/Yes with ANI – Is this device Trunk with/without ANI?
* '''[[Simultaneous call limitation | Call Limit]]''' – how many simultaneous calls can a user make?
* '''[[Multi_Server_support | Server]]''' – allows to choose server in multiple servers system. (While Carrier Class Addon is active, all Virtual and Fax devices are auto-assigned to all present Asterisk servers and cannot be re-assigned. Disabling Carrier Class Addon all Virtual and Fax devices will be assigned to default Asterisk server chosen from Addons > Carrier Class > Settings dropdown).
 

=Authentication=

===For dahdi devices:===

* '''Channel''' – which channel (or channel group) to use on PRI/BRI/PSTN card. Channels and groups should be configured in ''zapata.conf''.
 
===For SIP, H323 and IAX2 Devices:===

'''IP Authentication:'''

* '''IP Address''' – Device IP address in one of the following formats:
** '''IP''' – understands simple IP address, IP address with it's Subnet Mask and IP address with it's Range. For example ''192.168.0.1'', ''192.168.0.1/24'' and ''192.168.0.0-255''.
* '''Accept calls from any port''' – this option lets you receive calls from different Ports. This option is just on SIP (when [[SIP balancer]] in use) and H323.
* '''Port''' – Device Port.

'''Dynamic:'''

* '''Username''' – username you enter in your Device.
* '''Password''' – password you enter in your Device.
* '''IP Address''' – shows Device IP address.
* '''Port''' – shows Device Port.
 

=CallerID=
 
[[File:device_callerid.png]]
 
A CallerID consists of two parts: Name and Number. The "Number" part is transferred by default by all technologies (GSM, PSTN, SIP etc), but the "Name" part is transferred only by some. The number you see on your mobile phone when someone is calling you is the "Number" part.

* '''Name''' – the "Name" part of the CallerID.
* '''Number''' – the "Number" part of the CallerID. Only numerical values can be entered in this field.

NOTE: if you leave these fields empty, the user can set the CallerID name by himself. Usually it is not advisable to allow the user do it on his own. If CallerID number field is empty, system will take device username as CallerID number (if device is username/password authenticated).

* '''Number from DID''' - use the DID as the CallerID (only available when the device has DIDs), this option just sets CallerID Number to be equal to DID. Next time you will edit Device's settings, first option - Number will be checked

* '''Control by DIDs (CID from DIDs)''' – only available when the device has DIDs.

This setting lets you control what a CID user can enter in his devices. CID numbers should be from the set of the device's DIDs. They are mainly used when the user's PBX is connected over Trunk and many DIDs are routed to this Trunk. Calls coming out from this PBX must have a CallerID Number from the set of DIDs assigned to this Trunk (PBX). If "CID from DIDs" is checked, the system checks whether the device's CID number is from DIDs assigned to this device. If no DIDs are assigned, this value is ignored. In order for this setting to be active, CID Name/Number fields must be empty to allow the user to enter any CID number he wants. If "CID from DIDs" is checked and user enters any CID (not from his DIDs), the system will change his CID by his first available DID (selected from database by lowest ID).

* '''Control by CIDs''' – system checks incoming CallerID (Device's CallerID which is set on the Device). If CallerID matches one of the Device's CIDs - then such CallerID will be allowed. If it does not match - it will be changed to the selected CID.

* '''Control by Destination''' – This is advanced option which works by such algorithm:

# system checks Destination and checks DIDs assigned to calling Device. Then it finds 'nearest' DID to the Destination by subtracting DID from Destination in numerical form (Example, DID: 11111, Destination: 11112, |DID-Destination|=|1111-11112|=1) If Device has several DIDs - the lowest value is selected and this DID is set as CallerID.
#If Device has no DIDs - then User DIDs are used.
#If Device does not have DIDs and User has no DIDs, then CallerID set in Number field will be used.
#If nothing is set here - CallerID coming from Device will be used.

* '''Random Number from Number Pool''' – This functionality gives you an ability to send different caller number on each Call. Available from MOR X4. More information: [http://wiki.kolmisoft.com/index.php/Number_Pool Number Pool].

* '''Unknown''' - This option lets you set CallerID number to unknown.

* '''Copy Leg A Name to Leg B Number''' – Leg A name will be used as Leg B number.
 

=Network Related=

===For H323:===

* '''Fast start''' – option for faster H.245.
* '''h245 Tunneling''' – use H.245 without opening a second TCP/IP channel.
More info you can find [http://toncar.cz/Tutorials/VoIP/VoIP_Protocols_H323_Call_Signalling_Optimizations.html here].
 
===For SIP and IAX2===

* '''Media control''' – can reinvite. Do you want your server to stay in media path between your clients and your provider? Please note that this option will work only if it is supported from both, device and provider, sides and it allows to handle SIP packets only. Moreover, no codec mismatch between device and provider is available. Default value is ''No''. '''WARNING''': This option can cause [[Call was billed incorrectly|incorrect billing]].
** canreinvite = yes "allow RTP media direct"
** canreinvite = no "deny re-invites"
** canreinvite = nonat "allow reinvite when local, deny reinvite when NAT"
** canreinvite = update "use UPDATE instead of INVITE"
** canreinvite = update,nonat "use UPDATE when local, deny when NAT"
* '''NAT''' – the available options are: yes, no, never, force_rport, comedia. For a detailed explanation of these settings, refer to [http://www.voip-info.org/wiki/view/Asterisk+sip+nat here].
* '''Qualify''' – how often to "ask" the device about availability. More details [http://www.voip-info.org/wiki/view/Asterisk+sip+qualify here] for SIP devices and [http://www.voip-info.org/wiki/view/Asterisk+iax+qualify here] for IAX2 devices.
* '''IAX2 Trunking Mode''' – enable/disable trunking mode, which allows multiple voice streams to share a single "trunk" to another server, reducing overhead created by IP packets. Only on IAX2.
 

=Groups=

* [[What are callgroups and pickupgroups | Call Group]] – to which Call Group this device belongs.
* [[What are callgroups and pickupgroups | Pickup Group]] – which Call Groups this device can pick up.
 
=Voicemail=

* '''Email''' – where to send received Voicemail.
* '''Password''' – the digital password the user enters when he calls the Voicemail number to hear his messages.
 
=Codecs=

[[Which codecs should I choose for devices | Choose the codecs]] your provider uses.

NOTES:
* When no fields are checked, all codecs are available – for example, settings in sip.conf or iax.conf are effective.
* If the Provider and the Device do not have similar codecs, no call can be established.
* '''Enforce same codec as on Leg A''' - this option takes effect only when Device/Provider is used as Terminator. When this option is enabled, system checks what codec is used on Leg A. Then if Leg A codec matches one of codec enabled on Codecs list, system enforces callee to use that codec, by offering only that single codec in SDP data.
 

=Allowed Addresses=

'''IP, MASK''' – permits traffic from this IP only. You can find a detailed explanation [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+permit-deny-mask here].

You can enable multiple allowed addresses however this option should be turned on to be visible in main device settings window. Go to '''SETTINGS''' -> '''Setup''' -> '''Settings''' -> '''Various''' tab and choose ''Yes'' for ''Enable "Allowed addresses" option for Devices:''.
 
[[File:Settings visual advanced.png]]
 
And click ''Save changes''.

Now you will see an additional section in SIP and IAX2 Devices edit window:
 
[[File:Deviceconfig allowedaddr.png]]
 
'''Allowed addresses''' describes from what IP's or defined subnets accept traffic.
NOTE that this setting take effect only for '''Dynamic authentication''' devices. For '''IP authentication''' devices, this setting is not applied.

If you do not clearly understand what these settings are used for, leave them with default values.
 

=Advanced=

* '''Fromuser/Fromdomain''' – used when calling TO this peer FROM Asterisk. If you're using _register=>_ with another SIP proxy, this setting can come in handy since some SIP networks only allow users in the right domain with the correct user name.
* '''Trustrpid''' – defines whether or not Remote-Party-ID is trusted. It's defined in http://tools.ietf.org/id/draft-ietf-sip-privacy-04.txt.
* '''Sendrpid''' – defines whether a Remote-Party-ID SIP header should be sent. The default setting is "no". This field is often used by wholesale VoIP providers to provide calling party identity regardless of the privacy settings (the From SIP header).
* [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure Insecure]
** port: ignore the port number where request came from. '''WARNING:''' do NOT enable it on username/password authenticated devices.
** invite: don't require authentication of incoming INVITEs. '''WARNING:''' do NOT enable it on username/password authenticated devices.
** port, invite: don't require initial INVITE to authenticate and ignore the port where the request came from. '''WARNING:''' do NOT enable it on username/password authenticated devices.

Both of these settings are enabled by default for [[Authentication | IP Authenticated]] devices, but they can be changed, unless you are using Carrier Class Addon, which forces the device to ignore the port number and the authentication from an incoming request.

* '''[[SIP 302 Redirect Support | SIP 302 Redirect support]]''' – SIP Redirect support for the device. Disabled by default.
* '''Custom SIP Header''' - adds custom header to SIP request. Format is '''header: value''' (for example '''x-My-Custom-Header: my value''')
* '''[http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband Progressinband]''':
** yes – when "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio.
** no – send 180 Ringing if 183 has not yet been sent, establishing an audio path. If the audio path is established already (with 183), then send in-band ringing (this is the way Asterisk historically behaved because of buggy phones like Polycom's).
** never – whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behavior of Asterisk.
NOTE: if Progressinband does not work, add "prematuremedia=no" to sip.conf and reload Asterisk.
* '''Video support''' – does your provider support Video over IP? More info [http://www.voip-info.org/wiki/view/Asterisk+video here].
* '''[[Duplicate call prevention | Allow duplicate calls]]''' – the default setting is "no".
* '''Language''' – sets IVR language
* '''Use ANI (if available) for CallerID:''' – When the call comes the information about who is calling is found in CallerID field. This field is used to determine who is calling. When the call comes through PRI/SS7 channels - then additional information is available who is calling in ANI field (in call's data channel) so sometimes CallerID might be empty or anonymous, but the caller can be found in ANI field. This option allows to use ANI field as CallerID to determine and recognize who is calling.
* '''Incoming Call CallerID Presentation''' – sets CallerID Presentation. Please note that this setting applies only for incoming calls. More information can be found [http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerPres here] and [http://www.voip-info.org/wiki/view/Asterisk+cmd+CallingPres here]
* '''Change Failed Code To''' – if call fails change Hangup Cause Code to this value. This works only for outgoing calls from device. '''Not for Incoming'''
* '''Forward DID''' – it allows to forward call to DID which is assigned to Authorization by PIN or Calling Cards dial plan. After user enters PIN of any device or card, call gets connected with destination.
* '''Anti-resale (Auto-answer)''' – when option is set to YES, MOR answers the call before sending it to provider in order to generate FAS.
* '''Transport''' – lets you choose protocol(s) for data transfer. Appears only if device is SIP and when Asterisk 1.8 is enabled. Default value: ''udp''.
* '''T.38 Support''' – should T.38 pass-through be supported
* '''SRTP Encryption''' – should SRTP protocol be used for calls
* '''Block callerid if (number) simultaneous calls come from it''' – blocks CallerID if the entered number of simultaneous calls come from it

* '''Limit up to (number) calls, during (number) seconds''' – allows to set calls per second limit in some period. [[File:call_limit_per_period.png]]
* '''Outbound Proxy''' – send outbound signaling to this proxy, not directly to the peer (Internal Asterisk option).
 

=Tell Options=

* '''Tell rate before call''' - should MOR tell announce minute price (rounded to cents) before every call. Default value is "no".
* '''Tell balance before call''' – should MOR tell the user his balance every time he tries to dial? The default setting is "no".
* '''Tell time''' - should MOR tell the user his remaining time every time he tries to dial? The default setting is "no".
** Time is told in MINUTES only. Currently it is not possible to tell both in minutes and seconds.
* '''Tell remaining time when left''' – when some time is left, MOR will tell the remaining time to talk (in seconds).
* '''Repeat remaining time when left''' – repeats the remaining time when some time is left (in seconds).
 
=Debug=

* [[SIP debug info|'''Process SIPCHANINFO''']] - shows SIP channel info in [[Asterisk CLI]] and saves this information on database.
* '''Save Call Log''' - saves whole call log on database. Log can be overviewed by click [[File:Icon_callinfo.png]] in [[Last Calls]].
NOTE: debug should be enabled only if you are experiencing any problems. It should be disable in any other cases, because it stores lot of information on database.
 
This option is DISABLED globally by default. If you want it to be enabled, please contact Kolmisoft staff.
 

=Recordings=

This section is available when [[Recordings Addon]] is installed in the system.
 
=Comment=

Devices can have Comments, which gives information about the Device.
 
=See also=

* [[Grace_time | Grace time]]
* [[PAP_device_configuration | PAP device configuration]]
* [[H323 Device settings]]

Providers

2015-06-12T08:52:02Z

Rita: /* Advanced */

= Description =

In the old PSTN world, Providers were called Trunks. Providers are your route out of your VoIP network to the outside world.

Providers can be one of several types: ZAP, SIP, IAX2, and H323. The type determines which technology is used to connect to the Provider.

At the very start, you need to create a Provider through which you will dial out to PSTN or other VoIP networks.

Every Provider will charge you for calling a destination through his service (although the rate might be zero). The rate to every destination is different. Because of this, we need to have a Tariff (or Price List) for each Provider. This Tariff will tell our billing what price we will pay to a destination when using this Provider's services. In other words, this will be our Selfcost.
 
=Usage=

Go to '''SETTINGS –> Billing –> Providers'''.
 
[[File:providers_path.png]]
 
Here you can see a list of all Providers:
 
[[File:providers_list.png]]
 
* Click [[image:add.png]] '''New Provider''' to create new Provider, enter the Name for a new Provider, select its Technology, Tariff, Asterisk server which should be used for this provider and click Create.
* Click [[image:icon_edit.png]] icon to change settings for old Provider.

* Click [[Image:icon_check.png]] icon to disable Provider and it will not be used while it is cross in Active column. It affects only outgoing calls. Incoming calls from this provider will be accepted.
* Click [[Image:icon_cross.png]] icon to enable Provider if you want to use it again.
NOTE:
* You can't create a new Provider if there are no Tariffs available for Providers.
* A Provider is also able to send calls, not just receive them. More info [[Configure Provider which can make calls | here]]. In case you want to sell services to a provider you have to create such a provider.
 
== Settings ==

=== General ===
 
[[File:Prov addon.png]]
 
* '''Name''' – a name for the provider, for informational purposes.
* '''Technology''' – out of SIP/IAX2/ZAP/H323, choose which technology your provider uses - that is, the way you connect to the provider.
* '''Tariff''' – the list of rates the provider charges you.
* '''DTMF Mode''' – the available options are inband, info, RFC2833, and auto. Choose the one used by your provider.
* '''Location''' – which Localization rules are set to apply to INCOMING calls COMING FROM this provider.
* '''Automatic Number Identification''' - used in a special case explained here: [[Provider with ANI]]
* '''Ringing Timeout''' – this setting lets you set for how long this provider should be dialed before giving up. The default value is 60 seconds, and the minimum value is 30 seconds.
* '''Call Timeout''' – this setting lets you set the duration of the whole call: ringing and the period after connection.
* '''Device ID''' – informational data.
* '''Call limit''' – explained here: [[Simultaneous call limitation]].
* '''Hidden?''' – [[Hide_provider|hide/unhide provider]]

* '''Balance''' – provider balance. Provider balance increases when calls are made via that provider. Currently it is not available to decrease provider's balance. You can read more about [[Provider Billing Addon|Provider Billing Addon here]].

* '''Balance limit''' – this is the limit of balance that can be reached by the provider. After he reaches the specified balance, calls aren't sent through this provider anymore. Balance limit value of zero means that balance is unlimited.
 

=== Authentication ===

==== For ZAP providers ====

* '''Channel''' – which channel (or channel group) to use on a PRI/BRI/PSTN card. Channels and groups should be configured in zapata.conf.
 
==== For SIP/IAX2 providers ====

* '''Login''' – username if your provider asks for it.
* '''Password''' – password used for authentication by your provider.
* '''Authenticate by IP''' – just check the checkbox (Login/Password can be left empty).
** Make sure you put the correct Hostname/[[IP address]]!
* '''Register''' – should we register the provider? (option is not visible when '''Dynamic''' is selected in '''Network related''' section)
* '''Register extension''' – if the provider asks to use an extension for registration (usually they don't), enter it here.
 

=== Provider CallerID ===

'''CallerID''' – consists of two parts: Name and Number. The "Number" part is transferred by default on all technologies (GSM, PSTN, SIP etc), but the "Name" part only on some. The number you see on your mobile phone when someone is calling you is the "Number" part.
 
[[File:provider_callerid.png]]
 
* '''Name''' – the "Name" part of CallerID.
* '''Number''' – the "Number" part of CallerID.
* '''Random Number from Number Pool''' – the "Number" part of CallerID selected from [[Number Pool]]
.
'''NOTE:''' If you have this configured all incoming call from this provider will have that CallerID.
If you leave these fields empty, calls coming from this provider will have CallerID set by the Provider. It should almost always be this way.
 

=== Network related ===

* '''Hostname''' – hostname of the provider. Subnetting is supported.
* '''Server IP''' – the provider's IP. The value can be "dynamic", which means that provider can change its IP. To discover this value, do ping to the provider's hostname.
* '''Port''' – port used to connect to the provider (default: 5060 for SIP, 4569 for IAX2, 1720 for H323).
 
====H323 specific network settings====

* '''Fast Start''' – an option specific to H323; it is either on or off.
* '''h245 Tunneling''' – an option specific to H323; it is either on or of
 
====SIP specific network settings====

* '''Media control''' – canreinvite/transfer. Do you want your server to stay in the media path between your clients and your provider? Disable if you have a lot of 1-second answered calls.
* '''NAT''' – the available options are: yes, no, never, and route. For a detailed explanation of these settings, refer to [http://www.voip-info.org/wiki-Asterisk+sip+nat here].
 
====IAX2 specific network settings====

* '''IAX2 Trunking Mode''' – enable/disable trunking mode, which allows multiple voice streams to share a single "trunk" to another server, reducing overhead created by IP packets. More details [http://wiki.kolmisoft.com/index.php/Trunks here]
 
====SIP and IAX2 specific network settings====

* '''Qualify''' – how often to "ask" the provider about availability. More details [http://www.voip-info.org/wiki/view/Asterisk+sip+qualify here] for SIP providers and [http://www.voip-info.org/wiki/view/Asterisk+iax+qualify here] for IAX2 providers.
 

=== Servers ===
Allows to select on which Asterisk servers this provider should be used.
 
=== Codecs ===

Choose the codecs your provider uses.

NOTE: When no fields are checked, all codecs are available. ([[Image:asterisk_icon.png]] For example, settings in sip.conf or iax.conf are effective).
 
=== Advanced ===

* '''Grace time''' – if call duration is less than Grace Time, it will not be accounted.
* Progressinband:
**yes – when "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio.
**no – send 180 Ringing if 183 has not yet been sent, establishing an audio path. If the audio path is established already (with 183), then send in-band ringing (this is the way Asterisk historically behaved because of buggy phones like Polycom's).
**never – whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behavior of Asterisk.
* '''Video support''' – does your provider support Video over IP? More info [http://www.voip-info.org/wiki/view/Asterisk+video here].
* '''[[Duplicate call prevention | Allow duplicate calls]]''' – the default setting is "no".
* '''Interpret NO ANSWER as FAILED''' – default option is no (This will change the provider in LCR like it is failed, in last calls you will see
the real hangupcause – no answer)
* '''Interpret BUSY as FAILED''' – default option is no (This will change the provider in LCR like it is failed, in last calls you will see
the real hangupcause – busy)
* '''Fake Ring''' – should system generate Fake Ring or not? Use only if your Device/Provider has bad protocol implementation and does not send RING signal properly.
* '''Common Use''' – Provider can be used by Resellers which are able to create their own Providers. Reseller will be billed using this Provider. If provider is marked as common use and is used in one of reseller's LCR or is in Terminators list - admin is not allowed to delete this provider or remove common use option.
* '''Anti-resale (Auto-answer)''' – when option is set to YES, MOR answers the call before sending it to provider in order to generate FAS.
* '''Transport''' – lets you choose protocol <nowiki>[tcp, udp]</nowiki> for data transfer. Appears only if Provider is SIP and when Asterisk 1.8 is enabled. Default value: ''udp''.
* '''Time limit per day''' - Total time available for provider per day. When limit is reached another provider in LCR is used (total time is calculated after rounding rules and minimal time adjustment, so if your provider tariff has increment higher than 1 or minimal time set, then total provider time may be different than actual call time).
* '''Limit up to X, during Y seconds''' – you can set limit how many calls (X) can be done in some interval of seconds (Y). [[Last Calls]] will show '''FAILED([[Hangupcause Codes |243]])'''.
* '''Enforce same codec as on Leg A''' - this option takes effect only when Device/Provider is used as Terminator. When this option is enabled, system checks what codec is used on Leg A. Then if Leg A codec matches one of codec enabled on Codecs list, system enforces callee to use that codec, by offering only that single codec in SDP data.
 

=== SIP Specific ===

* '''Fromuser''' – [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+fromuser more details]
* '''Fromdomain''' – [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+fromuser more details]
* '''Trustrpid''' – This defines whether or not Remote-Party-ID is trusted. It's defined in http://tools.ietf.org/id/draft-ietf-sip-privacy-04.txt
* '''Sendrpid''' – defines whether a Remote-Party-ID SIP header should be sent. It defaults to "No". This field is often used by wholesale VoIP providers to provide calling party identity regardless of the privacy settings (the From SIP header).
* '''Insecure''' – [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure more details]
* '''T.38 Support''' – should we support T.38 pass-through?
* '''[[SIP 302 Reinvite Support]]''' – turns this SIP feature on or off.
* '''P-Asserted-Identity usage''' – The P-Asserted-Identity header field can be used to convey the proven identity of the originator of a request within a trusted network.
* '''Periodic check''' - Turns on or off functionality which disables provider if it is unreachable.
 

===Debug===

* '''Save Call Log''' – when this option is enabled you can access Call info information in last calls page.
 

=See also=

* [[H323 Provider settings]]
* [[Quick Providers changes]]
* [[Providers Statistics]]
* [[NO ANSWER/BUSY interpretation for providers]]
* [[I have a problem with Provider]]

MOR GUI Translation

2015-06-11T05:13:50Z

Rita: /* If you want to translate MOR into another language, it's very easy */

==MOR is translated into the following languages==

* English (Kolmisoft)
* Lithuanian (Kolmisoft)
* Spanish (Guillermo Salas M.(RazaMetaL)/Cesar Bermudez)
* Dutch (Pascal de Vos/Kay Reijnen)
* Italian (Giovanni Bosa)
* Albanian (Sokol Shehetila)
* Russian (ViPCall Network)
* Brazilian Portuguese (Fernando Soares)
* Estonian (Liivar Paas)
* Bulgarian (HemuZ.org)
* Swedish (VoIP Consult)
* German (Martin Melheritz)
* Armenian (Voipshop)
* French (Sébastien Keller)
* Polish (John)
* Romanian (Andrei Ivanoiu)
* Turkish (Koko Kokov)
* Indonesian (Dedi Setiawan/www.clarisense.co.id)
* Hungarian (John)
* Slovenian (Tom)
* Greek (Konstantinos Liadakis)
* Serbian (Alessandro Marzini and Ivex)
* Arabic (Amr Elgbaly)
* Macedonian (Ivo Trajkovski)
* Belarussian (Zhukov Oleg & Ko)
* Chinese (Steven Wang)
* Urdu (Andrew T. Wells)
* Ukrainian (Kontekst.net)

==If you want to translate MOR into another language, it's very easy==

# Download the MOR package or even better request the newest translation files from info@kolmisoft.com. This email address is protected from spam bots - you need Javascript enabled to view it.
# Find /lang directory (if MOR is installed, it's in '''/home/mor/lang'''). '''NOTE!''' From MOR X5 language files moved to '''/home/mor/config/locales'''.
# Make a copy of a en.rb file and name it accordingly. For example, if you are translating to Russian, name the file ru.rb (if you know another already translated language, select the appropriate file).
# Open that file in editor.
# Translate according to following rule:
# Let's say we have the line:
# l.store 'answered_calls', 'Answered calls'.
# So you need to translate only 'Answered calls'.
# Let's say we are translating to Norwegian, so the final line would be:
# l.store 'answered_calls', 'Besvarte Anrop'.
# Translate all remaining lines.
# '''Make file UTF-8 Standard.'''
# Please send it to info@kolmisoft.com. This email address is protected from spam bots - you need Javascript enabled to view it.

In order to test your new translated file, you need to take the following steps:

* Make sure your new translated file is in /home/mor/config/locales directory (for MOR X4 and older: /home/mor/lang).
* Restart Apache (/etc/init.d/httpd restart) and you should be able to test your translated file: go to SETTINGS -> Setup -> Settings -> Translations to find it).

Thank you for your contributions!

 

==See also==

* [[Translations]]
* [[Change flag for translation]]

Configuration from GUI

2015-06-03T12:50:22Z

Rita: /* See also */

=Where I can find this settings menu?=

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

[[File:Settings_path.png]]
 
== Global ==

'''SETTINGS –> Setup –> Settings –> [[Global_Settings | Global]]'''

* '''Company''' – name of the company.
* '''Company email''' – email address of the company.
* '''Version''' – version number.
* '''Copyright Title''' – copyright title supports html text format.
* '''Browser Title''' – browser title for MOR GUI.
* '''Front Page Text''' – some text to place on front page.
* '''Login Page Text''' – some text to place on login page
 

== Registration ==

'''SETTINGS –> Setup –> Settings –> Registration'''

* '''Enabled?''' – do we want to let users [[Online registration | register online]] automatically?
* '''Hide registration link in Login page?''' – tick this option if you do not want users to be able to register online.
* '''Default country'''
* '''Asterisk server IP''' – what IP should user connect his softphone/server to?
* '''Send email to user after registration''' – just yes or no.
* '''Send email to admin after registration''' – just yes or no.
* '''[[ReCAPTCHA]] configuration'''
* '''[[User_Agreement|Terms and conditions]]'''
* '''Allow to create user and device with matching username and password''' – just yes or no. If this setting is not enabled - username and password from MOR GUI will have to be different than device's username and password.
* '''Enable VAT checking for EU companies''' – you can choose whether to check '''TAX Registration Number:''' during the [[Online registration]] or not. Done over [http://ec.europa.eu/taxation_customs/vies/?locale=lt EU page].
* '''Allow blank VAT for EU companies''' – allows to leave blank '''TAX Registration Number:''' field during the [[Online registration]]
 
===See also===

* [[Default user]]
 

== Invoices ==

'''SETTINGS –> Setup –> Settings –> Invoices'''

Invoice configuration is divided into '''Prepaid''' and '''Postpaid'''.
 
=== Settings ===

* '''Number start''' – how the invoice number should start.
* '''Number length''' – the length in digits AFTER Number Start – e.g. Number Start is not counted into this length. This value is active only for first number type.
* '''Number type''' – select more appropriate number format.
* '''Invoice period start''' – when invoices are issued, default is first day of the month.
* '''Show calls in detailed invoice''' – yes or no.
* '''[[Outstanding balance(Debt) | Show balance line]]''' – shows balance line in Invoice PDF files when Invoice is generated from first day of month till last one.
* '''Round finals to 2 decimals''' – round or not total in Invoices to 2 decimals.
* '''Use short file name'''
* '''Invoice Page Limit:''' – invoice will be generated till this page limit. Additional page is added with an explanation that page limit was reached.
 

=== Details ===

* '''Address format''' – changes field positioning in the invoice for the address. You must select it in order to show address in the invoice.
* '''Address line 1-4''' – address info.
* '''Bank details line 1-5''' – bank details info.
* '''Invoice balance line''' – line for balance (e.g. "Your current balance is")
* '''Invoice to pay Line'''
* '''End title''' – a title for the end of your invoice.

* '''Add Average rate in detailed invoice''' – adds average rate in detailed invoice.
* '''Show Time in Minute''' – shows time in mm:ss instead of hh:mm:ss, for example, not 02:13:42 but 133:42.
 

=== Additional ===

* '''Show additional details on separate page''' – enables additional custom page at the end of the invoice.
* '''Additional details''' – the information to be displayed in a custom page at the end of the invoice.
 
=== Functionality ===

* '''Invoice allow recalculate after send''' – allows recalculating invoices after being sent.
 
===See also===

* [[Invoices]]
* [[Invoice configuration]]
* [[Invoices (MOR X6)]]
 

== Emails ==

'''SETTINGS –> Setup –> Settings –> Emails'''

* ''' Sending globally enabled''' – if this is checked, you will be able to use Mass Mailing and the system will send emails upon user registration.
* '''SMTP server''' – hostname of the server.
* '''Port''' – port of SMTP server.
* '''Login''' – username to your SMTP server.
* '''Password''' – password to your SMTP server.
* '''Batch size''' – value for Mass Mailing, or how many emails to send at once.
* '''From''' – email address to put into From: field in email. It is required to fill in(correctly) this field if email sending is enabled.
* '''Test email sending''' – save settings and press this to test email sending. Do not forget to save settings first!

'''P.S. Remember to set admin's e-mail in admin's personal settings. Without this test email sending will not work!'''

 
===See also===

* [[Emails|Mass Emails]]
* [[Email variables]]
 
== Various ==

'''SETTINGS –> Setup –> Settings –> Various'''

* '''Days for DID to close''' – how long to keep DID closed before making it free.
* '''Agreement number length''' – number length in digits.
* '''Change dahdi''' – Sometimes this is used to change "Zap" to something else in GUI, e.g. to "PSTN", mainly because not everybody knows what Zap stands for. PSTN is more common.
* '''Device PIN length''' – length for automatically generated PIN codes for devices.
* '''Device range MIN''' – from which extension automatic extension assigning starts.
* '''Device range MAX''' – till which extension automatic extension assigning goes.
** For example, from this interval: [Device range MIN, Device range MAX] MOR will select extensions for newly created Devices.
* '''Enable "Allowed addresses" option for Devices:''' – allows [[Device_settings#Allowed_Addresses | Allowed Addresses]] functionality in Devices settings.
* '''Enable "Allowed addresses" option for Providers:''' – allows Allowed Addresses functionality in Providers settings.
* '''User wholesale tariff enabled''' – enables user wholesale tariffs.
* '''Play IVR for unauthenticated call'''
* '''Banned CLIs default IVR''' – default IVR to route banned numbers.
* '''Do not allow duplicate device user names'''
* '''Do not allow users to change their email addresses'''
* '''Allow short passwords in devices'''
* '''Logout link''' – a user will be redirected to this link after pressing "Logout"
* '''Change ANSWER to FAILED if HGC not equal to 16 for Users''' – changes the Hangupcause for users in call reports from Answered to Failed if Hangupcause is not 16.
* '''Archive Calls older than''' – how old Calls should be archived?
* '''Archive Calls every day at''' – when Calls should be archive process started?
* '''Stop Archiving Calls at''' – when Calls should be archive process stopped ?
* '''Show Usernames on PDF/CSV Export Files in Last Calls''' – if checked, when particular User in Last Calls page search is selected, Last Calls PDF/CSV file name will be composed also from Name of selected User. If you are looking for All Users, no User Name will be used. (Name of User will be shown in Name of CSV/PDF file).
* '''Do not allow PREPAID User balance drop below zero (block them instead)'''

* '''Do not block POSTPAID User when balance goes bellow zero on subscription count'''
* '''Allow identical email addresses to different Users''' – allows to create Users or Register users with identical email addresses. After checking this setting you will be not able to check '''Show forgot password''', because this functionality require unique email in all system. Setting can be unchecked if all emails are unique in system.
* '''Allow same IP and Port combinations in multiple Devices''' – allows to create Devices with the same IP and Port combination.
* '''Server free space limit''' – indicator when system should inform admin about low [[Multi_Server_support|server]] space. By default 20%. Notice will appear in [[Integrity Check]] and every hour will be send an email "Server low free space" template which can be found and edited in [[Emails]] page.
 

== Tax ==

'''SETTINGS –> Setup –> Settings –> Tax'''

Settings are explained here: [[Multi-Tax system]]
 
== Backups ==

'''SETTINGS –> Setup –> Settings –> Backups'''

Configuration is explained here: [[Backup system]]
 
== API ==

'''SETTINGS –> Setup –> Settings –> API'''

* '''Allow API''' – enable/disable API.
* '''Allow GET API''' – enable/disable GET in API - use only for testing!
* '''API Secret Key''' – secret key for API. This will be used as a password for signing queries to MOR API, check [[MOR_API | MOR API documentation]] for more information
* '''XML API Extension''' – .NET does not follow standards (surprise!); enable this setting if you are using it to get XML starting with HTTP tags.
* '''Allow API Login Redirect''' – enable/disable
* '''API allow registration''' – enable/disable registration through API
* '''API allow Payments''' – enable/disable payments through API
* '''API Payment confirmation''' – enable/disable payments confirmation through API
* '''[[Balance_in_phone|Allow devices to check balance over HTTP]]''' – enable/disable device balance checking
* '''[[MOR_API_rate_get|Allow devices to check rate over HTTP]]''' – enable/disable device rate checking
* '''Disable hash checking''' – lets you skip hash parameter using API
* '''Uniquehash''' – shows your uniquehash
 
===See also===

[[MOR API]]
 

== Functionality ==

'''SETTINGS –> Setup –> Settings –> Functionality'''
 
===FAX===

* '''Fax device enabled''' – enable/disable Fax device in MOR. When admin enables FAX, Functionality tab appears in Reseller's GUI configuration. He can then enable or disabled fax for his own users himself.
* '''Fax2Email From Sender''' – there is a field "From:" in an email form. When MOR sends Fax to Email, "Fax2Email From Sender" value is displayed in "From:" field.
 

===CSV===

* '''CSV Column Separator''' – how columns are separated in imported/exported CSV files.
* '''CSV Decimal Separator''' – how decimal parts in numbers are separated in imported/exported CSV files.
* ''' Archive CSV file when size reaches''' – if CSV file reaches a certain size, the system will archive it to save time to download it (0 means to archive always)
 
===Active Calls===

* '''How many maximum [[Active_Calls|Active Calls]] to show in GUI''' – how many [[Active_Calls| Active Calls]] to show in GUI
* '''[[Active_Calls| Active Calls]] refresh interval''' – [[Active_Calls|Active Calls]] refresh interval in seconds
* '''Show [[Active_Calls|Active Calls]] for Users''' – tick if you want to allow users to see [[Active_Calls|Active Calls]]
* '''Show Servers in [[Active_Calls|Active Calls]]'''
* '''Show DIDs in [[Active_Calls|Active Calls]]'''
 

===Tariffs/Rates===

* '''Show rates for users''' – allows users to see rates
* '''Show rates without Tax''' – shows rates without TAX
* '''Show [[Advanced_Rates|Advanced Rates]] for users''' – allows users to see [[Advanced_Rates|Advanced Rates]]
* '''Show zero rates in LCR tariff export:''' – shows zero rates in LCR tariff exported file.
NOTE: if [[Custom Rates]] and/or Tariff Rates are set for the user, only the custom rates will be shown in the details of destination rates.
 

==Privacy==

'''SETTINGS –> Setup –> Settings –> Privacy'''

Settings are explained here: [[Privacy_to_hide_Destination_ends|Privacy to hide Destination ends]]
 
== Visual ==

'''SETTINGS –> Setup –> Settings –> Visual'''

* '''Items per page''' – the number of items to display per one page
* '''[[Global_date_format| Default date format]]''' – System default date format
* '''Default Time Format''' – System default time format
* '''Number digits''' – how many digits to display after a comma
* '''Hide Quick Stats in Start page''' – hide Quick stats from start page
* '''Hide device passwords for users''' – do not display device passwords for users
* '''Hide HELP banner''' – hide HELP banner in GUI
* '''Hide "I want to" help links'''
* '''[[Remove_Manual_link_from_admin_GUI|Hide "Manual" link]]''' – hides "Manual" link
* '''Show logo on first page''' - untick to hide logo on the start page
* '''Show Full Source (CallerID Name and Number)'''
* '''Hide payment options for postpaid users''' - will hide the payment option for all postpaid users (including Resellers if they are postpaid).
* '''Hide non-completed payments for user''' – hides non-completed payments (the ones which are started in MOR, but are not completed in a payment gateway website)
* '''Show only main page''' – this is a very advanced feature for those who want to hide default MOR menu tree on the left so that they could build their own menu using advanced techniques without modifying MOR GUI sources.
* '''Show forgot password''' – turns on '''[[MOR_forgot_user_password | Forgot password]]''' functionality.
* '''Hide Recordings for all users''' – hides recordings for all system users from menu and device settings.
* '''Show recordings with zero billsec''' – shows recordings with zero billsec
* '''[[Last_Calls#Faster_Page_Loading|Count Last Calls total values only after click]]''' – select to load Total Values lines separately in order to dramatically improve Last calls and Old calls loading time.
* '''Default Number Decimal''' – with this option you can set default number decimal for all GUI
* '''Show user_billsec for Users instead of billsec''' – shows user_billsec instead of billsec in Users invoices and last calls.

* '''Show Calls statistics to User for last''' – number of Calls statistics to show in days. For example, if you enter 5 days, users will only see 5 days old calls, they will not be able to check older calls.
* '''Show Device and it's CallerID in Last Calls''' – shows additional column in [[Last Calls]] page with Device and its CallerID information. Click to reach that Device settings page.
 

== Server load ==

'''SETTINGS –> Setup –> Settings –> Server load'''
 
Settings for GUI Server and Database Server

* '''HDD utilisation higher than''' – Percentage of CPU time during which I/O requests were issued to the device (bandwidth utilization for the device).
* '''CPU General Load higher than''' – CPU General load higher than – Percentage of the CPU for user processes.
* '''CPU Load Stats 1 higher than''' – average load per one minute on the system
* '''CPU MySQL process higher than''' – Percentage of the CPU for MySQL process
* '''CPU Ruby process higher than''' – Percentage of the CPU for Ruby process
* '''CPU Asterisk process (if present) higher than''' – Percentage of the CPU for Asterisk process
 
* '''Delete Server Load Stats older than''' – deletes internal servers load statistics older than defined number of days.
 
=See also=

* [[Server is overloaded]]

MOR API user balance get

2015-05-18T13:50:25Z

Rita: /* Request */

=Description=
This feature enables you to check the balance over http.
 
=Configuration=

1. Go to '''SETTINGS –> Setup –> Settings'''
 
[[File:settings_path.png]]
 

2. In API tab check the marked option:
 
[[File:Settings_various_balance_over_http.png]]
 

=Usage=

* From '''MOR X4''' call: '''/api/user_balance_get'''
* For Backwards-compatibility old name 'balance' is also usable. Call: '''/api/balance'''
* Methods: POST, GET(if allowed, not recomended)
 
===Parametres===

Parameters which are '''included''' into hash(if it is used):

* username – username of user whose balance you want to get. '''Required'''.
* currency – currency code or value USER in which case balance will be returned in user's currency. '''Optional'''.

'''Note that parameters are shown in the same order as they have to be when generating the hash. Last parameter in a row is API_Secret_Key'''
 
Parameters which are '''not included''' into hash:

* [[MOR API hash construction | hash]] - SHA1 hash constructed using parameters above and API_Secret_Key ([[MOR API hash construction | More described in Constructing hash]]). '''Required'''.
* user_currency = 1 – in this case balance will be returned in user's currency with name of users currency. '''Optional'''.
 
'''NOTE''' Without '''currency''' or '''user_currency''' parameters API request will return Balance in [[Currencies#Default_Currency|Default System Currency]]. If user_currency is used currency parameter will be ignored.
 

===Request===

We have username=USERNAME, API Secret Key = secret

We send:

HTML POST: http://<SERVER_IP>/billing/api/user_balance_get?username=USERNAME&hash=79e75e75f9cd6821e7b661250a45294b4579f262

Optionaly you can specify currency, valid values are USER, in that case balance is returned in user's currency. And currency code(EUR, USD etc.)

HTML POST: http://<SERVER_IP>/billing/api/user_balance_get?username=USERNAME&currency={USER|CURRENCY_CODE}&hash=79e75e75f9cd6821e7j5h89f6a45294b45g8u9f5
 

===Returns===

'''Success''' returns balance of user.

<?xml version="1.0" encoding="UTF-8"?>
<page>
<balance> user balance </balance>
</page>

With user_currency = 1 parameter.
<?xml version="1.0" encoding="UTF-8"?>
<page>
<balance> user balance </balance>
<currency> user currency </currency>
</page>

 
----

'''Errors'''

When hash is incorrect.

<?xml version="1.0" encoding="UTF-8"?>
<status>
<error>Incorrect hash</error>
</status>

When user is not found.
<?xml version="1.0" encoding="UTF-8"?>
<page>
<error>User was not found</error>
</page>

When feature is off.
<?xml version="1.0" encoding="UTF-8"?>
<page>
<error>Feature disabled</error>
</page>
 

=See Also=

* [[MOR API]]