Kolmisoft Wiki - User contributions [en]

Didx.net

2014-11-11T15:39:00Z

Algirdas:

=What is didx.net=
DIDX (DIDXchange) is a place where VOIP companies buy and sell DID (Direct Inward Dialing) phone numbers.

=How to configure a DID purchased from didx.net to work with MOR?=
1. Go to purchased DIDs:
 
[[File:Didx_purchased_dids.png]]
 
2. Then select a DID for edit:
 
[[File:Didx_edit_purchased_did.png]]
 

3. Select "New SIP"

In this field you have to enter your DID and server address with the following syntax: DID@example.com

For example:

You have purchased a DID: 37068111888
Your MOR server address is: 127.0.0.1

Then you have to put in this field:
37068111888@127.0.0.1

'''Don't forget to press the update button!'''
 
[[File:Didx_purchased_dids_addr_did.png]]

 
4. Now login to MOR GUI and go to '''SETTINGS - Billing -> DIDs'''
 
[[File:Did_path.png]]
 
5. Now create a new DID and leave it free at the moment. To create a DID press the marked icon:
 
[[File:Dids_new_did.png]]
 
6. Now go back to DIDs list: '''SETTINGS - Billing -> DIDs'''
 
[[File:Did_path.png]]
 
7. Edit the DID you created:
 
[[File:Didx_edit_did.png]]
 
8. Select a user you want to reserve a DID for:
 
[[File:Didx_edit_did_reserve_for_user.png]]
 
9. Reserve the DID for a user - press the marked button:
 
[[File:Didx_edit_did_reserve_for_user_commit.png]]
 
10. Now you have to select the device you configured earlier with callflow, in this case it will be SIP/1001:
 
[[File:Didx_edit_did_select_device.png]]
 
11. Save the changes by pressing the marked button:
 
[[File:Didx edit did select device commit.png]]
 

= How to allow incoming connections from DIDx.net =

== DIDx.net used IP's ==

DIDX may send you calls from any of the IP's below.

67.15.180.14 - didx.net
67.15.128.14 - sip1.didx.net
67.15.128.18 - sip2.didx.net
209.62.66.242 - sip3.didx.net
198.101.50.4 - sip4.didx.net
198.101.50.2 - sip5.didx.net
74.55.75.30 - sip6.didx.net
174.133.195.194 - sip7.didx.org
67.19.199.170 - sip1.virtualphoneline.com
88.208.247.34 - eu1.didx.org
88.208.208.34 - eu2.didx.org
74.52.4.234 - sip9.didx.net

== Configuration on MOR ==

In order to receive incoming connections from DIDx you can ether create IP authenticated Providers for all IP's or you can do the following:
It is possible to implement multiple DIDx IP addresses by making these changes:

1. Create a provider in MOR, with a name didx: 
[[File:provider_create_didx.png]] 
2. Afterwards, enter any IP (or leave blank) in Provider Network settings, and write down Provider Device ID value, you
will need it later. 
[[File:provider_settings_didx.png]] 
3. Now connect to your server over ssh and locate this file /etc/asterisk/extensions_mor_didx.conf and find these lines:
[from-didx]
exten => _X.,1,Set(CDR(ACCOUNTCODE)=0)
exten => _X.,2,Goto(mor,${EXTEN},1)
Now you need to change '''(ACCOUNTCODE)=0''' value (0) to Provider Device ID value as follows:
exten => _X.,1,Set(CDR(ACCOUNTCODE)=221)
exten => _X.,2,Goto(mor,${EXTEN},1)
 
4. Make sure, file /etc/asterisk/sip.conf has this line included:
#include sip_didx.conf
 
NOTE: sip_didx.conf file contains list of DIDWW server IPs, so this file must be included if you are using DIDx DIDs. Without this list, all incoming calls from DIDx will not be allowed by MOR. 
 
5. [[How_to_restart_Asterisk_server | Restart Asterisk]]
 
Now your system is updated with all the DIDx IP addresses that are listed in their recommendations.
 
'''Now there is no need to create one Provider for each IP'''
 

'''That's all'''

Didx.net

2014-11-11T14:48:25Z

Algirdas:

=What is didx.net=
DIDX (DIDXchange) is a place where VOIP companies buy and sell DID (Direct Inward Dialing) phone numbers.

=How to configure a DID purchased from didx.net to work with MOR?=
1. Go to purchased DIDs:
 
[[File:Didx_purchased_dids.png]]
 
2. Then select a DID for edit:
 
[[File:Didx_edit_purchased_did.png]]
 

3. Select "New SIP"

In this field you have to enter your DID and server address with the following syntax: DID@example.com

For example:

You have purchased a DID: 37068111888
Your MOR server address is: 127.0.0.1

Then you have to put in this field:
37068111888@127.0.0.1

'''Don't forget to press the update button!'''
 
[[File:Didx_purchased_dids_addr_did.png]]

 
4. Now login to MOR GUI and go to '''SETTINGS - Billing -> DIDs'''
 
[[File:Did_path.png]]
 
5. Now create a new DID and leave it free at the moment. To create a DID press the marked icon:
 
[[File:Dids_new_did.png]]
 
6. Now go back to DIDs list: '''SETTINGS - Billing -> DIDs'''
 
[[File:Did_path.png]]
 
7. Edit the DID you created:
 
[[File:Didx_edit_did.png]]
 
8. Select a user you want to reserve a DID for:
 
[[File:Didx_edit_did_reserve_for_user.png]]
 
9. Reserve the DID for a user - press the marked button:
 
[[File:Didx_edit_did_reserve_for_user_commit.png]]
 
10. Now you have to select the device you configured earlier with callflow, in this case it will be SIP/1001:
 
[[File:Didx_edit_did_select_device.png]]
 
11. Save the changes by pressing the marked button:
 
[[File:Didx edit did select device commit.png]]
 

= How to allow incoming connections from DIDx.net =

== DIDx.net used IP's ==

DIDX may send you calls from any of the IP's below.

67.15.180.14 - didx.net
67.15.128.14 - sip1.didx.net
67.15.128.18 - sip2.didx.net
209.62.66.242 - sip3.didx.net
198.101.50.4 - sip4.didx.net
198.101.50.2 - sip5.didx.net
74.55.75.30 - sip6.didx.net
174.133.195.194 - sip7.didx.org
67.19.199.170 - sip1.virtualphoneline.com
88.208.247.34 - eu1.didx.org
88.208.208.34 - eu2.didx.org
74.52.4.234 - sip9.didx.net

== Configuration on MOR ==

In order to receive incoming connections from DIDx you can ether create IP authenticated Providers for all IP's or you can do the following:
It is possible to implement multiple DIDx IP addresses by making these changes:

1. Create a provider in MOR, with a name didx: 
[[File:provider_create_didx.png]] 
2. Afterwards, enter any IP (or leave blank) in Provider Network settings, and write down Provider Device ID value, you
will need it later. 
[[File:provider_settings_didx.png]] 
3. Now connect to your server over ssh and locate this file /etc/asterisk/extensions_mor_didww.conf and find these lines:
[from-didx]
exten => _X.,1,Set(CDR(ACCOUNTCODE)=0)
exten => _X.,2,Goto(mor,${EXTEN},1)
Now you need to change '''(ACCOUNTCODE)=0''' value (0) to Provider Device ID value as follows:
exten => _X.,1,Set(CDR(ACCOUNTCODE)=221)
exten => _X.,2,Goto(mor,${EXTEN},1)
 
4. Make sure, file /etc/asterisk/sip.conf has this line included:
#include sip_didx.conf
 
NOTE: sip_didx.conf file contains list of DIDWW server IPs, so this file must be included if you are using DIDx DIDs. Without this list, all incoming calls from DIDx will not be allowed by MOR. 
 
5. [[How_to_restart_Asterisk_server | Restart Asterisk]]
 
Now your system is updated with all the DIDx IP addresses that are listed in their recommendations.
 
'''Now there is no need to create one Provider for each IP'''
 

'''That's all'''

Didx.net

2014-11-11T14:47:13Z

Algirdas:

=What is didx.net=
DIDX (DIDXchange) is a place where VOIP companies buy and sell DID (Direct Inward Dialing) phone numbers.

=How to configure a DID purchased from didx.net to work with MOR?=
1. Go to purchased DIDs:
 
[[File:Didx_purchased_dids.png]]
 
2. Then select a DID for edit:
 
[[File:Didx_edit_purchased_did.png]]
 

3. Select "New SIP"

In this field you have to enter your DID and server address with the following syntax: DID@example.com

For example:

You have purchased a DID: 37068111888
Your MOR server address is: 127.0.0.1

Then you have to put in this field:
37068111888@127.0.0.1

'''Don't forget to press the update button!'''
 
[[File:Didx_purchased_dids_addr_did.png]]

 
4. Now login to MOR GUI and go to '''SETTINGS - Billing -> DIDs'''
 
[[File:Did_path.png]]
 
5. Now create a new DID and leave it free at the moment. To create a DID press the marked icon:
 
[[File:Dids_new_did.png]]
 
6. Now go back to DIDs list: '''SETTINGS - Billing -> DIDs'''
 
[[File:Did_path.png]]
 
7. Edit the DID you created:
 
[[File:Didx_edit_did.png]]
 
8. Select a user you want to reserve a DID for:
 
[[File:Didx_edit_did_reserve_for_user.png]]
 
9. Reserve the DID for a user - press the marked button:
 
[[File:Didx_edit_did_reserve_for_user_commit.png]]
 
10. Now you have to select the device you configured earlier with callflow, in this case it will be SIP/1001:
 
[[File:Didx_edit_did_select_device.png]]
 
11. Save the changes by pressing the marked button:
 
[[File:Didx edit did select device commit.png]]
 

= How to allow incoming connections from DIDx.net =

== DIDx.net used IP ==

DIDX may send you calls from any of the IP's below.

67.15.180.14 - didx.net
67.15.128.14 - sip1.didx.net
67.15.128.18 - sip2.didx.net
209.62.66.242 - sip3.didx.net
198.101.50.4 - sip4.didx.net
198.101.50.2 - sip5.didx.net
74.55.75.30 - sip6.didx.net
174.133.195.194 - sip7.didx.org
67.19.199.170 - sip1.virtualphoneline.com
88.208.247.34 - eu1.didx.org
88.208.208.34 - eu2.didx.org
74.52.4.234 - sip9.didx.net

== Configuration on MOR ==

In order to receive incoming connections from DIDx you can ether create IP authenticated Providers for all IP's or you can do the following:
It is possible to implement multiple DIDx IP addresses by making these changes:

1. Create a provider in MOR, with a name didx: 
[[File:provider_create_didx.png]] 
2. Afterwards, enter any IP (or leave blank) in Provider Network settings, and write down Provider Device ID value, you
will need it later. 
[[File:provider_settings_didx.png]] 
3. Now connect to your server over ssh and locate this file /etc/asterisk/extensions_mor_didww.conf and find these lines:
[from-didww]
exten => _X.,1,Set(CDR(ACCOUNTCODE)=0)
exten => _X.,2,Goto(mor,${EXTEN},1)
Now you need to change '''(ACCOUNTCODE)=0''' value (0) to Provider Device ID value as follows:
exten => _X.,1,Set(CDR(ACCOUNTCODE)=221)
exten => _X.,2,Goto(mor,${EXTEN},1)
 
4. Make sure, file /etc/asterisk/sip.conf has this line included:
#include sip_didx.conf
 
NOTE: sip_didx.conf file contains list of DIDWW server IPs, so this file must be included if you are using DIDx DIDs. Without this list, all incoming calls from DIDx will not be allowed by MOR. 
 
5. [[How_to_restart_Asterisk_server | Restart Asterisk]]
 
Now your system is updated with all the DIDx IP addresses that are listed in their recommendations.
 
'''Now there is no need to create one Provider for each IP'''
 

'''That's all'''

File:Provider settings didx.png

2014-11-11T14:46:59Z

Algirdas: Create DIDx provider step two.

Create DIDx provider step two.

File:Provider create didx.png

2014-11-11T14:46:22Z

Algirdas: Create provider DIDx first step

Create provider DIDx first step

Installation

2014-10-17T10:18:38Z

Algirdas:

Installation procedure:

* Choose the hardware according to [[Recommended_hardware_for_MOR_server | recommended specifications]]

* Prepare a server with fresh minimal [[Centos_installation |Centos 6.5 installation]]

* '''IMPORTANT!''' Make sure that minimum requirements are 100 GB to be assign to the root " / "

* Open [[Ports_which_should_be_opened |necessary ports]]

* [http://www.kolmisoft.com/contacts Send us] IP/root

Installation will be done by Kolmisoft engineers within 48 hours.

200 MOR can't determine who is calling

2014-10-17T07:03:54Z

Algirdas: /* How to disable such call attempts */

[[Image:audio.png]] SYSTEM ERROR. CODE 200. The system cannot determine who is calling.
The IP from which the call comes is not entered in the system or is entered incorrectly.
Some other fields in configuration may also be missing.
Please consult the online manual at wiki.kolmisoft.com.

The Provider/Device is not authenticated.

* Check settings for Provider/Device.
* For provider - make sure '''hostname''' and '''IP address''' have correct values.
** Both these fields should be filled.
** If '''hostname''' is not assigned, then it should have an IP address, with the same value as the '''IP address''' field.
* Check the '''port''' setting.
** The call may not be coming through the default SIP port 5060, but from 5061. Check this.
** '''Port''' refers to the SOURCE port. That means the port the call is coming FROM, not the port it is coming TO.
* If you are using H323 in file ''h323.conf'', add '''UserByAlias=no'''.
* If you are using ZAP/H323 provider, you may have forgotten about the '''accountcode=X''' setting. ([http://wiki.kolmisoft.com/index.php/MOR_PRO_v0.5_documentation#How_to_create_Zap.2FH323_providers More info])
* Make sure accountcode for dialing device is not 0 (ERROR[3433]: app_mor_authentication.c:11 mor_get_user_by_acc: Accountcode cannot be 0!)

 
=== The Main Reason ===

You forget to describe IP in MOR from which the call is coming.

Example:

[Dec 17 08:37:42] WARNING[21726]: app_mor.c:3557 mor_get_user_by_acc: User not found by accountcode:
[Dec 17 08:37:42] ERROR[21726]: app_mor.c:667 mor_authorize: '''MOR can't determine who is calling. Make sure accountcode is set for caller (Provider or Device).'''
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:692 mor_authorize: Caller type: Local
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:706 mor_authorize: Localized destination: 5143161536
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3872 process_sipchaninfo: ============== SIPCHANINFO ===============
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3897 process_sipchaninfo: '''Peer IP: 64.34.135.88'''
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3898 process_sipchaninfo: Source IP: 64.34.135.88
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3899 process_sipchaninfo: From: sip:7052058393@64.34.164.254
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3900 process_sipchaninfo: Contact: sip:7052058393@64.34.164.254
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3901 process_sipchaninfo: Useragent: Voice Network Inc 1b
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3902 process_sipchaninfo: Peername:
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3903 process_sipchaninfo: T38Passthrough: 0
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3907 process_sipchaninfo: ==========================================

The above means that MOR does not know about IP 64.34.135.88.

'''SOLUTION:''' assign this IP to some Provider (or Device) in MOR.

'''The most common reason why people get this error'''

The reason is, that very often MOR administrators forget about Trustrpid, Sendrpid, Insecure: port/invite options in device settings window. They must match incoming call settings. The image below shows settings that are usually correct.

[[Image:note21.png]]

 

=== How to disable such call attempts ===

By default MOR allows guest calls (unauthenticated calls) so the calls would be processed and would show up in the STATISTICS->Calls->Last Calls after they are rejected and the Admin could see such call attempts as well. This is very useful when you are trying to setup a new device and the call does not go through due to bad authentication.
However, you can disallow such calls to be processed at all by changing the "allowguest" option in sip.con file. To do so, simply open the sip.conf file with any editor(in the example we will be using "vi")

vi /etc/asterisk/sip.conf

Then find the allowguest option and change it to:

allowguest=no

After this save the file and reload asterisk so new setting would take affect:

asterisk -rx 'sip reload'

----
See also:
* [[Hangupcause Codes]]

200 MOR can't determine who is calling

2014-10-17T07:03:21Z

Algirdas: /* How to disable such call attempts */

[[Image:audio.png]] SYSTEM ERROR. CODE 200. The system cannot determine who is calling.
The IP from which the call comes is not entered in the system or is entered incorrectly.
Some other fields in configuration may also be missing.
Please consult the online manual at wiki.kolmisoft.com.

The Provider/Device is not authenticated.

* Check settings for Provider/Device.
* For provider - make sure '''hostname''' and '''IP address''' have correct values.
** Both these fields should be filled.
** If '''hostname''' is not assigned, then it should have an IP address, with the same value as the '''IP address''' field.
* Check the '''port''' setting.
** The call may not be coming through the default SIP port 5060, but from 5061. Check this.
** '''Port''' refers to the SOURCE port. That means the port the call is coming FROM, not the port it is coming TO.
* If you are using H323 in file ''h323.conf'', add '''UserByAlias=no'''.
* If you are using ZAP/H323 provider, you may have forgotten about the '''accountcode=X''' setting. ([http://wiki.kolmisoft.com/index.php/MOR_PRO_v0.5_documentation#How_to_create_Zap.2FH323_providers More info])
* Make sure accountcode for dialing device is not 0 (ERROR[3433]: app_mor_authentication.c:11 mor_get_user_by_acc: Accountcode cannot be 0!)

 
=== The Main Reason ===

You forget to describe IP in MOR from which the call is coming.

Example:

[Dec 17 08:37:42] WARNING[21726]: app_mor.c:3557 mor_get_user_by_acc: User not found by accountcode:
[Dec 17 08:37:42] ERROR[21726]: app_mor.c:667 mor_authorize: '''MOR can't determine who is calling. Make sure accountcode is set for caller (Provider or Device).'''
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:692 mor_authorize: Caller type: Local
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:706 mor_authorize: Localized destination: 5143161536
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3872 process_sipchaninfo: ============== SIPCHANINFO ===============
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3897 process_sipchaninfo: '''Peer IP: 64.34.135.88'''
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3898 process_sipchaninfo: Source IP: 64.34.135.88
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3899 process_sipchaninfo: From: sip:7052058393@64.34.164.254
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3900 process_sipchaninfo: Contact: sip:7052058393@64.34.164.254
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3901 process_sipchaninfo: Useragent: Voice Network Inc 1b
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3902 process_sipchaninfo: Peername:
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3903 process_sipchaninfo: T38Passthrough: 0
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3907 process_sipchaninfo: ==========================================

The above means that MOR does not know about IP 64.34.135.88.

'''SOLUTION:''' assign this IP to some Provider (or Device) in MOR.

'''The most common reason why people get this error'''

The reason is, that very often MOR administrators forget about Trustrpid, Sendrpid, Insecure: port/invite options in device settings window. They must match incoming call settings. The image below shows settings that are usually correct.

[[Image:note21.png]]

 

=== How to disable such call attempts ===

By default MOR allows guest calls (unauthenticated calls) so the calls would be processed and would show up in the STATISTICS->Calls->Last Calls after they are rejected and the Admin could see such call attempts as well.
This is very useful when you are trying to setup a new device and the call does not go through due to bad authentication.
However, you can disallow such calls to be processed at all by changing the "allowguest" option in sip.con file. To do so, simply open the sip.conf file with any editor(in the example we will be using "vi")

vi /etc/asterisk/sip.conf

Then find the allowguest option and change it to:

allowguest=no

After this save the file and reload asterisk so new setting would take affect:

asterisk -rx 'sip reload'

----
See also:
* [[Hangupcause Codes]]

200 MOR can't determine who is calling

2014-10-17T07:01:44Z

Algirdas:

[[Image:audio.png]] SYSTEM ERROR. CODE 200. The system cannot determine who is calling.
The IP from which the call comes is not entered in the system or is entered incorrectly.
Some other fields in configuration may also be missing.
Please consult the online manual at wiki.kolmisoft.com.

The Provider/Device is not authenticated.

* Check settings for Provider/Device.
* For provider - make sure '''hostname''' and '''IP address''' have correct values.
** Both these fields should be filled.
** If '''hostname''' is not assigned, then it should have an IP address, with the same value as the '''IP address''' field.
* Check the '''port''' setting.
** The call may not be coming through the default SIP port 5060, but from 5061. Check this.
** '''Port''' refers to the SOURCE port. That means the port the call is coming FROM, not the port it is coming TO.
* If you are using H323 in file ''h323.conf'', add '''UserByAlias=no'''.
* If you are using ZAP/H323 provider, you may have forgotten about the '''accountcode=X''' setting. ([http://wiki.kolmisoft.com/index.php/MOR_PRO_v0.5_documentation#How_to_create_Zap.2FH323_providers More info])
* Make sure accountcode for dialing device is not 0 (ERROR[3433]: app_mor_authentication.c:11 mor_get_user_by_acc: Accountcode cannot be 0!)

 
=== The Main Reason ===

You forget to describe IP in MOR from which the call is coming.

Example:

[Dec 17 08:37:42] WARNING[21726]: app_mor.c:3557 mor_get_user_by_acc: User not found by accountcode:
[Dec 17 08:37:42] ERROR[21726]: app_mor.c:667 mor_authorize: '''MOR can't determine who is calling. Make sure accountcode is set for caller (Provider or Device).'''
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:692 mor_authorize: Caller type: Local
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:706 mor_authorize: Localized destination: 5143161536
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3872 process_sipchaninfo: ============== SIPCHANINFO ===============
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3897 process_sipchaninfo: '''Peer IP: 64.34.135.88'''
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3898 process_sipchaninfo: Source IP: 64.34.135.88
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3899 process_sipchaninfo: From: sip:7052058393@64.34.164.254
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3900 process_sipchaninfo: Contact: sip:7052058393@64.34.164.254
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3901 process_sipchaninfo: Useragent: Voice Network Inc 1b
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3902 process_sipchaninfo: Peername:
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3903 process_sipchaninfo: T38Passthrough: 0
[Dec 17 08:37:42] NOTICE[21726]: app_mor.c:3907 process_sipchaninfo: ==========================================

The above means that MOR does not know about IP 64.34.135.88.

'''SOLUTION:''' assign this IP to some Provider (or Device) in MOR.

'''The most common reason why people get this error'''

The reason is, that very often MOR administrators forget about Trustrpid, Sendrpid, Insecure: port/invite options in device settings window. They must match incoming call settings. The image below shows settings that are usually correct.

[[Image:note21.png]]

 

=== How to disable such call attempts ===

By default MOR allows guest calls (unauthenticated calls) so the calls would be processed and would show up in the STATISTICS->Calls->Last Calls and the Admin could see such call attempts as well.
This is very useful when you are trying to setup a new device and the call does not go through due to bad authentication.
However, you can disallow such calls to be processed at all by changing the "allowguest" option in sip.con file. To do so, simply open the sip.conf file with any editor(in the example we will be using "vi")

vi /etc/asterisk/sip.conf

Then find the allowguest option and change it to:

allowguest=no

After this save the file and reload asterisk so new setting would take affect:

asterisk -rx 'sip reload'

----
See also:
* [[Hangupcause Codes]]

Upgrade MOR GUI

2014-10-14T09:05:22Z

Algirdas: moved Upgrade MOR GUI to Update MOR GUI: Wrong Page name

#REDIRECT [[Update MOR GUI]]

Update MOR GUI

2014-10-14T09:05:22Z

Algirdas: moved Upgrade MOR GUI to Update MOR GUI: Wrong Page name

=What is MOR GUI Update?=
MOR GUI update is a procedure to update your graphical user interface (GUI) to the newest version.
=Why should I update my GUI?=
Some features are being implemented after official version release, so if you find, that you lack some feature documented in wiki - you should update your GUI to receive that feature.

Please check for important changes [[Important_changes | here]]. It could be that you will need some changes on configuration after update.

'''P.S. Use the procedure that matches your version, otherwise it might break your system. To update to a newer version - please contact the Kolmisoft team'''

=How to determine what GUI version I have?=
Please run this command to determine your current MOR GUI version:
 
svn info /home/mor | grep URL | awk -F"/" '{print $NF}'
You will get a similar output:
[root@localhost ~]# svn info /home/mor | grep URL | awk -F"/" '{print $NF}'
9

 

=What else should I know?=
MOR GUI and MOR core versions '''must match'''. Always update your GUI according to core version. MOR core version can be determined by running this command:
 
asterisk -rx "mor show status" | grep Version
You will get a similar output:
[root@localhost ~]# asterisk -rx "mor show status" | grep Version
Version: '''9.1.22'''

In this example we have MOR with version 9 core. So to update GUI you should follow the GUI update guide for version 9
 
=How to update the GUI?=

 
----
 

== MOR 8 update ==

To update '''[[MOR 8 documentation | MOR 8]]''' GUI run such commands over SSH as root:

rm -fr /usr/src/mor
svn co http://svn.kolmisoft.com/mor/install_script/trunk/ /usr/src/mor
/usr/src/mor/upgrade/0.8/hard_fix.sh

 
----
 

== MOR 9 update ==

To upgrade '''[[MOR 9 documentation | MOR 9]]''' GUI run such commands over SSH as root:

rm -fr /usr/src/mor
svn co http://svn.kolmisoft.com/mor/install_script/trunk/ /usr/src/mor
/usr/src/mor/upgrade/9/fix.sh

 
----
 

== MOR 10 update ==

To update '''[[MOR 10 documentation | MOR 10]]''' GUI run such commands over SSH as root:

rm -fr /usr/src/mor
svn co http://svn.kolmisoft.com/mor/install_script/trunk/ /usr/src/mor
/usr/src/mor/upgrade/10/fix.sh

== MOR 11 update ==

To update '''[[MOR_11 | MOR 11]]''' GUI run such commands over SSH as root:

rm -fr /usr/src/mor
svn co http://svn.kolmisoft.com/mor/install_script/trunk/ /usr/src/mor
/usr/src/mor/upgrade/11/fix.sh

== MOR 12 update ==

To update '''[[MOR_12 | MOR 12]]''' GUI run such commands over SSH as root:

rm -fr /usr/src/mor
svn co http://svn.kolmisoft.com/mor/install_script/trunk/ /usr/src/mor
/usr/src/mor/upgrade/12.126/fix.sh

== MOR X3 update ==

To update '''[[MOR_X3 | MOR X3]]''' GUI run such commands over SSH as root:

rm -fr /usr/src/mor
svn co http://svn.kolmisoft.com/mor/install_script/trunk/ /usr/src/mor
/usr/src/mor/upgrade/12/fix.sh

== MOR X4 update ==

To update '''[[MOR_X4 | MOR X4]]''' GUI run such commands over SSH as root:

rm -fr /usr/src/mor
svn co http://svn.kolmisoft.com/mor/install_script/trunk/ /usr/src/mor
/usr/src/mor/upgrade/x4/fix.sh

== MOR X5 update ==

To update '''[[MOR_X5 | MOR X5]]''' GUI run such commands over SSH as root:

rm -fr /usr/src/mor
svn co http://svn.kolmisoft.com/mor/install_script/trunk/ /usr/src/mor
/usr/src/mor/x5/update.sh

Training

2014-09-15T08:57:56Z

Algirdas: /* FREE Online Group Training every week */

[[File:training.jpg|right]]
= Types of Training =

 
== FREE Online Group Training every week ==

* Online Skype call with screen-sharing, video, audio
* Every Client is eligible for this Training
* Training sessions are every Tuesday 3:00 PM (UTC +2 Lithuania time)
* To participate in the weekly training send a contact invitation to skype name: kolmisoft_training.
* You can send the skype invitation during the session as well. You will be added to the call.
* Your contact will be deleted after the session, so please send a new invitation before the next weeks training if you wish to participate.

 

== Personalized Training ==
* Specially for only one client at agreed time for agreed questions
* Online with screen-sharing, video, audio

 
= Training purposes =

Kolmisoft provides training for the following basic purposes:

* To understand how MOR works and to learn how to use MOR for your specific business operation. For example:
** Add provider
** Upload tariff
** Generate calling cards
** Configure IVR
** etc
* To understand how to start and prosper in the following business models:
** Retail Business (PC2Phone, PC2PC, Phone2Phone)
** Wholesale Business (Termination/Origination)
** Callback (Over Call, Email, SMS)
** Calling Card (using PIN and PINless)
** Call Shop
** Auto Dialer

 
= Training prices and plans =

'''Group Online training is FREE.'''

Personalized Training's price is 100 EUR/h.

Also for start-up companies or for companies who want to add new business direction to their services, we recommend special plan which is focused on '''[[How to make money]]''' and covers much more:

* Training material;
* Detailed Webinar;
* 1 month Phone/Skype/Email/Web/Video consultations;
* Financial calculators to clearly see your investments and profit;
* TO-DO step-by-step checklists for starting your business processes;
* Ready agreement forms for your clients, suppliers, partners;
* FULL Infrastructure for 1 month - you can check what you have learned!
* FREE traffic to start your business now!

 

= How is the training done? =

Live training is done using live meeting platform Webex.

Not live training (if you choose special plan) is done by:

* Webinars
* Emails with useful material
* etc.

 
= FAQ =

Here you can check frequently asked questions related with training.

Training

2014-09-15T08:39:22Z

Algirdas: /* FREE Online Group Training every week */

[[File:training.jpg|right]]
= Types of Training =

 
== FREE Online Group Training every week ==

* Online Skype call with screen-sharing, video, audio
* Every Client is eligible for this Training
* Training sessions are every Tuesday 3:00 PM (UTC +2 Lithuania time)
* To participate in the weekly training send a contact invitation to skype name: kolmisoft_training.
* You can sent the skype invitation during the session as well. You will be added to the call.
* Your contact will be deleted after the session, so please send a new invitation before the next weeks training if you wish to participate.

 

== Personalized Training ==
* Specially for only one client at agreed time for agreed questions
* Online with screen-sharing, video, audio

 
= Training purposes =

Kolmisoft provides training for the following basic purposes:

* To understand how MOR works and to learn how to use MOR for your specific business operation. For example:
** Add provider
** Upload tariff
** Generate calling cards
** Configure IVR
** etc
* To understand how to start and prosper in the following business models:
** Retail Business (PC2Phone, PC2PC, Phone2Phone)
** Wholesale Business (Termination/Origination)
** Callback (Over Call, Email, SMS)
** Calling Card (using PIN and PINless)
** Call Shop
** Auto Dialer

 
= Training prices and plans =

'''Group Online training is FREE.'''

Personalized Training's price is 100 EUR/h.

Also for start-up companies or for companies who want to add new business direction to their services, we recommend special plan which is focused on '''[[How to make money]]''' and covers much more:

* Training material;
* Detailed Webinar;
* 1 month Phone/Skype/Email/Web/Video consultations;
* Financial calculators to clearly see your investments and profit;
* TO-DO step-by-step checklists for starting your business processes;
* Ready agreement forms for your clients, suppliers, partners;
* FULL Infrastructure for 1 month - you can check what you have learned!
* FREE traffic to start your business now!

 

= How is the training done? =

Live training is done using live meeting platform Webex.

Not live training (if you choose special plan) is done by:

* Webinars
* Emails with useful material
* etc.

 
= FAQ =

Here you can check frequently asked questions related with training.

I have a problem with Call Flood

2014-07-15T17:05:48Z

Algirdas:

If in active calls you see a lot of pending calls an none of them are connecting:

[[File:flood.jpg]]

This can indicate that asterisk is overloaded, in such case setting the "maxload" value in asterisk.conf should solve the issue.
What you should do:
1. Open asterisk.conf file with any editor:

vi /etc/asterisk/asterisk.conf

2. Add or edit the "maxload" line, sett the value to be equal to the number of cores one your server (or lover).

maxload = "number of cores"

3. Restart asterisk:

service asterisk restart

File:Flood.jpg

2014-07-15T17:04:03Z

Algirdas: uploaded a new version of "File:Flood.jpg"

I have a problem with Call Flood

2014-07-15T17:01:45Z

Algirdas:

If in active calls you see a lot of pending calls an none of them are connecting:

[[File:flood.jpg]]

This can indicate that asterisk server is overloaded, in such case setting the "maxload" value in asterisk.conf should solve the issue.
What you should do:
1. Open asterisk.conf file with any editor:

vi /etc/asterisk/asterisk.conf

2. Add or edit the "maxload" line, sett the value to be equal to the number of cores one your server (or lover).

maxload = "number of cores"

3. Restart asterisk:

service asterisk restart

File:Flood.jpg

2014-07-15T17:00:01Z

Algirdas: uploaded a new version of "File:Flood.jpg"

File:Flood.jpg

2014-07-15T16:57:41Z

Algirdas:

I have a problem with Call Flood

2014-07-15T16:54:59Z

Algirdas: Created page with 'If in active calls you see a lot of pending calls an none of them are connecting: File:flood.jpg This can indicate that asterisk server is overloaded, in such case setting th…'

If in active calls you see a lot of pending calls an none of them are connecting:
[[File:flood.jpg]]
This can indicate that asterisk server is overloaded, in such case setting the "maxload" value in asterisk.conf should solve the issue.
What you should do:
1. Open asterisk.conf file with any editor:

vi /etc/asterisk/asterisk.conf

2. Add or edit the "maxload" line, sett the value to be equal to the number of cores one your server (or lover).

maxload = "number of cores"

3. Restart asterisk:

service asterisk restart

I HAVE A PROBLEM

2014-07-15T16:43:51Z

Algirdas:

[[Image:problems.gif|right]]Before proceeding, you need to upgrade MOR GUI to the newest version because many problems can be fixed just by upgrading GUI.

* [[Upgrade MOR GUI]]

After the upgrade, 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 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 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 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]]

Registration Status

2014-05-23T13:54:51Z

Algirdas: /* Status */

=About=
This feature was developed to make MOR administrator's life easier. With this feature an administrator is able to track SIP device status from GUI and solve problems faster.
 
==Technical details (requirements)==
These requirements must be fulfilled in order for this functionality to work correctly:

*Works only with SIP devices.
*Device must try to register or someone should call to that device in order for that device status to be visible.
*Device must register to an Asterisk server where it is assigned.
 
===Example: ===

If a device is assigned to a server A but registers to server B - status will not be shown (devices must register to the server they are assigned to).

 

==Status==

[[File:bullet_green.png]] '''OK''' - Device is registered, all is OK

[[File:bullet_yellow.png]] '''LAGGED''' - Device is registered but connection to it is poor

[[File:bullet_red.png]] '''UNREACHABLE''' - Device tried to register before, but currently it is unreachable

[[File:bullet_black.png]] '''UNKNOWN''' - means the device state status could not be found. This is NOT an alternative message to the previous '''UNREACHABLE''' message. This status is created when you add a new device and the system does not yet have any data about its registration status. It also occurs if a device unregistered improperly.

[[File:bullet_white.png]] '''Unmonitored''' - Device should not register to server so its status is unmonitored, this device made calls, so server is aware of its existence

'''NO STATUS''' - Device never tried to register to Server and Server does not know anything about this device.

 

==Device list==
 
[[File:Device_status.png]]
 
==Device edit==
 
[[File:Device_status_edit_device.png]]
 
==Very technical details==
''You will probably never need these.''

These files must exist:

* /usr/local/mor/mor_retrieve_peers
* /etc/cron.d/mor_minute_actions
 
===Logs===
If you want to get a lot of logs put DEBUG = 1 in mor.conf

Logs for this feature can be found here:

*/var/log/mor/ami_debug.log
 

=See also=
* [[Device Registration Status is not visible in the system]]
* [http://kb.smartvox.co.uk/voip-sip/sip-registration-explained/ SIP Registration Explained]

Registration Status

2014-05-23T13:54:31Z

Algirdas: /* Status */

=About=
This feature was developed to make MOR administrator's life easier. With this feature an administrator is able to track SIP device status from GUI and solve problems faster.
 
==Technical details (requirements)==
These requirements must be fulfilled in order for this functionality to work correctly:

*Works only with SIP devices.
*Device must try to register or someone should call to that device in order for that device status to be visible.
*Device must register to an Asterisk server where it is assigned.
 
===Example: ===

If a device is assigned to a server A but registers to server B - status will not be shown (devices must register to the server they are assigned to).

 

==Status==

[[File:bullet_green.png]] '''OK''' - Device is registered, all is OK

[[File:bullet_yellow.png]] '''LAGGED''' - Device is registered but connection to it is poor

[[File:bullet_red.png]] '''UNREACHABLE''' - Device tried to register before, but currently it is unreachable

[[File:bullet_black.png]] '''UNKNOWN''' - means the device state status could not be found. This is NOT an alternative message to the previous '''UNREACHABLE''' message. This status is created when you add a new device and the system does not yet have any data about its registration status it also occurs if a device unregistered improperly.

[[File:bullet_white.png]] '''Unmonitored''' - Device should not register to server so its status is unmonitored, this device made calls, so server is aware of its existence

'''NO STATUS''' - Device never tried to register to Server and Server does not know anything about this device.

 

==Device list==
 
[[File:Device_status.png]]
 
==Device edit==
 
[[File:Device_status_edit_device.png]]
 
==Very technical details==
''You will probably never need these.''

These files must exist:

* /usr/local/mor/mor_retrieve_peers
* /etc/cron.d/mor_minute_actions
 
===Logs===
If you want to get a lot of logs put DEBUG = 1 in mor.conf

Logs for this feature can be found here:

*/var/log/mor/ami_debug.log
 

=See also=
* [[Device Registration Status is not visible in the system]]
* [http://kb.smartvox.co.uk/voip-sip/sip-registration-explained/ SIP Registration Explained]

Recordings Addon

2014-05-22T10:28:41Z

Algirdas: /* Functionality */

=Description=

Recordings functionality can be used to record selected users' calls for Monitoring purpose.
 
== Functionality ==

Recording Addon provides this functionality:

* Record User Calls.
* Send a Recording to email.
* Limit space Users' space for Recordings.
* Record Calls without User knowledge.
* Let Users manage their Recordings (send to email/listen/download/delete).
* Convert recordings from WAV to MP3 format and store them on an external server, to ease the load on the main server.

NOTE: Resellers is able to have the same functions as an administrator if '''Allow to use recording functionality''' option is checked by Admin in resellers settings. In order for the calls to be recorded for resellers users the "Record calls for this Device:" option (located in the device edit window, "Recordings" block) has to be enabled by the reseller for the devices for which the calls should be recorded.
 
=== Limitations ===

* Currently it is not possible to record Calls for a Device that is accessed from IVR. If a call goes like this: IVR -> Device, and the device has the option to record calls for this device, these calls will not be recorded.

* Recordings will not be saved on the server if there will be left less then 10% of free space. Such recordings will be sent directly to the Email (if email is specified in GUI Interface) and will be completely removed from the server.

* Currently it is not possible to record Calls that are made between local extensions, but it is possible to record calls if a local device has a DID assigned to it and the call is made dialing the DID number, so the call would go like this: local extension > DID > local extension.
 

=== Sound file format ===

Recordings are saved into mp3 format. 1mb can save around 4min of recordings.
 

=Configuration=

In order to use Recording, you have to set up which Users' Calls you want to record.

Go to '''ADDONS –> Recordings –> Users''' and you will see this window:
 
[[Image:recordings1.png]]
 

Here you can check the '''Forced record calls for this User''' field for the Users you want to record. By checking '''Allow to use recording functionality''' option you can authorize the User to use the Recordings Addon functionality.Click '''Update''' at the bottom for the changes to take effect.

If you leave '''Server space (Mb)''' 0 - there will be no limitations in data usage (HDD will fill up after some time).

If you want to view/listen/download/delete recorded Calls go to '''ADDONS –> Recordings'''. You will see something like this:
 
[[Image:recordings2.png]]
 
'''NOTE''' Recordings are NOT available in Calls between 2 SIP Devices in MOR.
 
== Record not answered calls ==

In order to enable this functionality few changes a needed: 

1) Edit /etc/asterisk/mor.conf 

vi /etc/asterisk/mor.conf

and set '''record_not_answered_calls =''' '''1''' 

This option record_not_answered_calls - save audio to the file while the channel is not answered (not bridged), default 0 (by default records only answered calls).
 

2) Option Show Recordings with zero billsec should be enabled in GUI. 

To enable it go to '''SETTINGS –> Setup –> Visual''' and check option '''Show recordings with zero billsec'''. 
 
[[Image:recordingams.png]]
 
Now system records not answered calls for all users which are using recording functionality. 

Recordings are located in '''ADDONS –> Recordings.''' 

In recording list you will be able to see recordings without duration. 

Such calls were not answered but recorded.
 
[[Image:recordingams2.png]]
 
==Advanced options==

Manage settings for recordings upload to external server go to '''ADDONS –> Recordings –> Settings'''
 
[[File:recordings_settings_path.png]]
 
you will reach settings page
 
[[File:recordings_addon_settings.png]]
 
* Use external Server - should system send recordings to external server?
* Recordings server - details such as IP/Port/Login where port is SSH default port 22 and Login is default root.
* By default recording files in remote server will be placed at /tmp directory

'''P.S. SSH keys must be configured without password in order scp command could transfer the recordings to other server. It this functionality is not enabled - the recordings are stored in local server'''
 
== Administrator functionality ==

'''ADDONS –> Recordings'''
 
[[File:Recordings_Addons_Recordings_List.png]]
 
Here the System administrator can see all recordings. He can search Recordings by:

* Period.
* Source (CallerID).
* Destination (Dialed number).
* User.
* Device.

With each recording, it is possible to:

* Check its details.
* Check Call details.
* Add comment to it.
* Listen to it.
* Download it.
* Delete it.

Admin can decide:

* Whether or not the user can use Recording Addon functionality (set to record calls, send recordings to email, etc).
* Which user calls should always be recorded (without user knowledge).
 
'''SETTINGS –> Users –> Edit (on selected user)'''

The Recordings section has the following options:

* Allow to use recording functionality - can this user use the Recording addon functionality?

* Forced record calls for this User - record calls for this user (all devices) without user knowledge. This option has the highest priority.
when enabled - records for all User devices will be recorded regardless of other options (value in Device settings: Record calls for this Device: - will be ignored, Server space - dedicated for User will be ignored ant etc )

* Server space - how much HDD space to reserve for recordings of this user. Takes effect only if Admin has not enabled recordings for User himself. Such options should be set on appropriate User/Device by Admin in order to enable this function for User:
User settings
Forced record calls for this User: NO
Device settings:
Record calls for this Device : NO
Keep Recordings on Server after sent to Email: NO

* Send deleted recordings to this email. If HDD space is used, the recording will be deleted and sent to this email. If this email is empty, the recording will be sent to the default user email.
 
'''Device –> Edit window'''

The following options are available for Device:

* '''Record calls for this Device''' - record calls for this Device (without user knowledge). If this option is enabled, recordings must be deleted manually. If this option is set to YES, then server space - dedicated for User will be ignored and records will be placed on server regardless of this value
* '''Show Recordings to the User for this Device''' - allows User to see such recordings
* '''Send Recording to Email''' - when recording is completed, send to it email.
* '''Recordings Email''' - send to this email address.
* '''Keep Recordings on Server after sent to Email''' - should the recording be left on the server after being sent to email? If this option is set to YES, then server space - dedicated for User will be ignored and records will be placed on server regardless of this value
 
[[File:device_recordings_settings.png]]
 

== Recordings bulk management ==

This feature allows an administrator to quickly delete Recordings for selected period or selected Device
 
==== Who can use this functionality? ====

* This feature is only available for MOR administrator
 
==== How to use this functionality ====
Go to '''ADDONS –> Recordings'''
 
[[File:Recordings_menu_path.png]]
 
Click on [[image:icon_edit.png]] '''Bulk management''' link.
 
[[File:Recordings_bulk_management.png]]
 
Here you can check an option to delete by device or by date, select device/date and click '''Delete'''.
 
[[File:Recordings_bulk_management_confirm.png]]
 
Confirm that you really want to delete recordings of selected period/device.
 

= User settings =

The Administrator must authorize the user to use the Recording Addon functionality before User can use its functions. This can be done by checking '''Allow to use recording functionality''' in the User Edit window.
 
'''PERSONAL MENU –> Details'''

* Server space - shows used/reserved HDD space for recordings.
* Email for deleted Recordings - to which email recordings should be sent if there is not enough space for them.
 
'''PERSONAL MENU –> Details –> Devices –> Edit'''

* Record calls for this Device - user can set to record calls for this device or not.
* Send Recording to Email - should recording be sent to email?
* Recordings Email - the email address email to which recordings should be sent.
* Keep Recordings on Server after sent to Email - should recordings be left on the server after being sent to email?
 
'''PERSONAL MENU –> Calls –> Recordings'''

On this page, the User can manage his recordings.
 
= Guides =

== Setup user so he could see his recorded calls ==

1. As admin, go to '''User Edit''' settings [[Image:icon_edit.png]] and check '''Allow to use recording functionality'''. User will be able to use Recording addon functionality.

2. Log in as this User and in '''PERSONAL MENU –> Details –> Devices''', click [[Image:icon_edit.png]] for Device and in the Recordings section check '''Record calls for this Device: YES.'''

3. Now the user will be able to see his calls in '''PERSONAL MENU –> Call –> Recordings'''.
 

== Delete recordings after sending it via e-mail ==

1. As admin, go to '''User Edit''' settings [[Image:icon_edit.png]] and check '''Allow to use recording functionality'''. User will be able to use Recording addon functionality. Do not check '''Forced record calls for this User''' because it this way recordings will '''NOT''' be deleted.

2. Log in as this User and in '''PERSONAL MENU –> Personal Details –> Devices''', click [[Image:icon_edit.png]] for Device and in the Recordings section check '''Keep Recordings on Server after sent to Email: NO.'''

3. Now recordings will be deleted immediately after they will be sent via e-mail.
 

=See also=

* [[MOR Addons]]

Recordings Addon

2014-02-21T09:45:13Z

Algirdas: /* Limitations */

=Description=

Recordings functionality can be used to record selected users' calls for Monitoring purpose.
 
== Functionality ==

Recording Addon provides this functionality:

* Record User Calls.
* Send a Recording to email.
* Limit space Users' space for Recordings.
* Record Calls without User knowledge.
* Let Users manage their Recordings (send to email/listen/download/delete).
* Convert recordings from WAV to MP3 format and store them on an external server, to ease the load on the main server.

NOTE: Resellers is able to have the same functions as an administrator if '''Allow to use recording functionality''' option is checked by Admin in resellers settings.
 
=== Limitations ===

* Currently it is not possible to record Calls for a Device that is accessed from IVR. If a call goes like this: IVR -> Device, and the device has the option to record calls for this device, these calls will not be recorded.

* Recordings will not be saved on the server if there will be left less then 10% of free space. Such recordings will be sent directly to the Email (if email is specified in GUI Interface) and will be completely removed from the server.

* Currently it is not possible to record Calls that are made between local extensions, but it is possible to record calls if a local device has a DID assigned to it and the call is made dialing the DID number, so the call would go like this: local extension > DID > local extension.
 

=== Sound file format ===

Recordings are saved into mp3 format. 1mb can save around 4min of recordings.
 
=Configuration=

In order to use Recording, you have to set up which Users' Calls you want to record.

Go to '''ADDONS –> Recordings –> Users''' and you will see this window:
 
[[Image:recordings1.png]]
 

Here you can check the '''Forced record calls for this User''' field for the Users you want to record. By checking '''Allow to use recording functionality''' option you can authorize the User to use the Recordings Addon functionality.Click '''Update''' at the bottom for the changes to take effect.

If you leave '''Server space (Mb)''' 0 - there will be no limitations in data usage (HDD will fill up after some time).

If you want to view/listen/download/delete recorded Calls go to '''ADDONS –> Recordings'''. You will see something like this:
 
[[Image:recordings2.png]]
 
'''NOTE''' Recordings are NOT available in Calls between 2 SIP Devices in MOR.
 
== Record not answered calls ==

In order to enable this functionality few changes a needed: 

1) Edit /etc/asterisk/mor.conf 

vi /etc/asterisk/mor.conf

and set '''record_not_answered_calls =''' '''1''' 

This option record_not_answered_calls - save audio to the file while the channel is not answered (not bridged), default 0 (by default records only answered calls).
 

2) Option Show Recordings with zero billsec should be enabled in GUI. 

To enable it go to '''SETTINGS –> Setup –> Visual''' and check option '''Show recordings with zero billsec'''. 
 
[[Image:recordingams.png]]
 
Now system records not answered calls for all users which are using recording functionality. 

Recordings are located in '''ADDONS –> Recordings.''' 

In recording list you will be able to see recordings without duration. 

Such calls were not answered but recorded.
 
[[Image:recordingams2.png]]
 
==Advanced options==

Manage settings for recordings upload to external server go to '''ADDONS –> Recordings –> Settings'''
 
[[File:recordings_settings_path.png]]
 
you will reach settings page
 
[[File:recordings_addon_settings.png]]
 
* Use external Server - should system send recordings to external server?
* Recordings server - details such as IP/Port/Login where port is SSH default port 22 and Login is default root.

'''P.S. SSH keys must be configured without password in order scp command could transfer the recordings to other server. It this functionality is not enabled - the recordings are stored in local server'''
 
== Administrator functionality ==

'''ADDONS –> Recordings'''
 
[[File:Recordings_Addons_Recordings_List.png]]
 
Here the System administrator can see all recordings. He can search Recordings by:

* Period.
* Source (CallerID).
* Destination (Dialed number).
* User.
* Device.

With each recording, it is possible to:

* Check its details.
* Check Call details.
* Add comment to it.
* Listen to it.
* Download it.
* Delete it.

Admin can decide:

* Whether or not the user can use Recording Addon functionality (set to record calls, send recordings to email, etc).
* Which user calls should always be recorded (without user knowledge).
 
'''SETTINGS –> Users –> Edit (on selected user)'''

The Recordings section has the following options:

* Allow to use recording functionality - can this user use the Recording addon functionality?
* Forced record calls for this User - record calls for this user (all devices) without user knowledge.
* Server space - how much HDD space to reserve for recordings of this user.
* Send deleted recordings to this email. If HDD space is used, the recording will be deleted and sent to this email. If this email is empty, the recording will be sent to the default user email.
 
'''Device –> Edit window'''

The following options are available for Device:

* '''Record calls for this Device''' - record calls for this Device (without user knowledge). If this option is enabled, recordings must be deleted manually.
* '''Show Recordings to the User for this Device''' - allows User to see such recordings
* '''Send Recording to Email''' - when recording is completed, send to it email.
* '''Recordings Email''' - send to this email address.
* '''Keep Recordings on Server after sent to Email''' - should the recording be left on the server after being sent to email?
 
[[File:device_recordings_settings.png]]
 

== Recordings bulk management ==

This feature allows an administrator to quickly delete Recordings for selected period or selected Device
 
==== Who can use this functionality? ====

* This feature is only available for MOR administrator
 
==== How to use this functionality ====
Go to '''ADDONS –> Recordings'''
 
[[File:Recordings_menu_path.png]]
 
Click on [[image:icon_edit.png]] '''Bulk management''' link.
 
[[File:Recordings_bulk_management.png]]
 
Here you can check an option to delete by device or by date, select device/date and click '''Delete'''.
 
[[File:Recordings_bulk_management_confirm.png]]
 
Confirm that you really want to delete recordings of selected period/device.
 

= User settings =

The Administrator must authorize the user to use the Recording Addon functionality before User can use its functions. This can be done by checking '''Allow to use recording functionality''' in the User Edit window.
 
'''PERSONAL MENU –> Details'''

* Server space - shows used/reserved HDD space for recordings.
* Email for deleted Recordings - to which email recordings should be sent if there is not enough space for them.
 
'''PERSONAL MENU –> Details –> Devices –> Edit'''

* Record calls for this Device - user can set to record calls for this device or not.
* Send Recording to Email - should recording be sent to email?
* Recordings Email - the email address email to which recordings should be sent.
* Keep Recordings on Server after sent to Email - should recordings be left on the server after being sent to email?
 
'''PERSONAL MENU –> Calls –> Recordings'''

On this page, the User can manage his recordings.
 
= Guides =

== Setup user so he could see his recorded calls ==

1. As admin, go to '''User Edit''' settings [[Image:icon_edit.png]] and check '''Allow to use recording functionality'''. User will be able to use Recording addon functionality.

2. Log in as this User and in '''PERSONAL MENU –> Details –> Devices''', click [[Image:icon_edit.png]] for Device and in the Recordings section check '''Record calls for this Device: YES.'''

3. Now the user will be able to see his calls in '''PERSONAL MENU –> Call –> Recordings'''.
 

== Delete recordings after sending it via e-mail ==

1. As admin, go to '''User Edit''' settings [[Image:icon_edit.png]] and check '''Allow to use recording functionality'''. User will be able to use Recording addon functionality. Do not check '''Forced record calls for this User''' because it this way recordings will '''NOT''' be deleted.

2. Log in as this User and in '''PERSONAL MENU –> Personal Details –> Devices''', click [[Image:icon_edit.png]] for Device and in the Recordings section check '''Keep Recordings on Server after sent to Email: NO.'''

3. Now recordings will be deleted immediately after they will be sent via e-mail.
 

=See also=

* [[MOR Addons]]

Recordings Addon

2014-02-21T08:23:48Z

Algirdas: /* Limitations */

=Description=

Recordings functionality can be used to record selected users' calls for Monitoring purpose.
 
== Functionality ==

Recording Addon provides this functionality:

* Record User Calls.
* Send a Recording to email.
* Limit space Users' space for Recordings.
* Record Calls without User knowledge.
* Let Users manage their Recordings (send to email/listen/download/delete).
* Convert recordings from WAV to MP3 format and store them on an external server, to ease the load on the main server.

NOTE: Resellers is able to have the same functions as an administrator if '''Allow to use recording functionality''' option is checked by Admin in resellers settings.
 
=== Limitations ===

* Currently it is not possible to record Calls for a Device that is accessed from IVR. If a call goes like this: IVR -> Device, and the device has the option to record calls for this device, these calls will not be recorded.

* Recordings will not be saved on the server if there will be left less then 10% of free space. Such recordings will be sent directly to the Email (if email is specified in GUI Interface) and will be completely removed from the server.

* Currently it is not possible to record Calls that are made between local extensions, but it is possible to record calls if a local device has a DID assigned to it and the call is made dialing the DID number.
 

=== Sound file format ===

Recordings are saved into mp3 format. 1mb can save around 4min of recordings.
 
=Configuration=

In order to use Recording, you have to set up which Users' Calls you want to record.

Go to '''ADDONS –> Recordings –> Users''' and you will see this window:
 
[[Image:recordings1.png]]
 

Here you can check the '''Forced record calls for this User''' field for the Users you want to record. By checking '''Allow to use recording functionality''' option you can authorize the User to use the Recordings Addon functionality.Click '''Update''' at the bottom for the changes to take effect.

If you leave '''Server space (Mb)''' 0 - there will be no limitations in data usage (HDD will fill up after some time).

If you want to view/listen/download/delete recorded Calls go to '''ADDONS –> Recordings'''. You will see something like this:
 
[[Image:recordings2.png]]
 
'''NOTE''' Recordings are NOT available in Calls between 2 SIP Devices in MOR.
 
== Record not answered calls ==

In order to enable this functionality few changes a needed: 

1) Edit /etc/asterisk/mor.conf 

vi /etc/asterisk/mor.conf

and set '''record_not_answered_calls =''' '''1''' 

This option record_not_answered_calls - save audio to the file while the channel is not answered (not bridged), default 0 (by default records only answered calls).
 

2) Option Show Recordings with zero billsec should be enabled in GUI. 

To enable it go to '''SETTINGS –> Setup –> Visual''' and check option '''Show recordings with zero billsec'''. 
 
[[Image:recordingams.png]]
 
Now system records not answered calls for all users which are using recording functionality. 

Recordings are located in '''ADDONS –> Recordings.''' 

In recording list you will be able to see recordings without duration. 

Such calls were not answered but recorded.
 
[[Image:recordingams2.png]]
 
==Advanced options==

Manage settings for recordings upload to external server go to '''ADDONS –> Recordings –> Settings'''
 
[[File:recordings_settings_path.png]]
 
you will reach settings page
 
[[File:recordings_addon_settings.png]]
 
* Use external Server - should system send recordings to external server?
* Recordings server - details such as IP/Port/Login where port is SSH default port 22 and Login is default root.

'''P.S. SSH keys must be configured without password in order scp command could transfer the recordings to other server. It this functionality is not enabled - the recordings are stored in local server'''
 
== Administrator functionality ==

'''ADDONS –> Recordings'''
 
[[File:Recordings_Addons_Recordings_List.png]]
 
Here the System administrator can see all recordings. He can search Recordings by:

* Period.
* Source (CallerID).
* Destination (Dialed number).
* User.
* Device.

With each recording, it is possible to:

* Check its details.
* Check Call details.
* Add comment to it.
* Listen to it.
* Download it.
* Delete it.

Admin can decide:

* Whether or not the user can use Recording Addon functionality (set to record calls, send recordings to email, etc).
* Which user calls should always be recorded (without user knowledge).
 
'''SETTINGS –> Users –> Edit (on selected user)'''

The Recordings section has the following options:

* Allow to use recording functionality - can this user use the Recording addon functionality?
* Forced record calls for this User - record calls for this user (all devices) without user knowledge.
* Server space - how much HDD space to reserve for recordings of this user.
* Send deleted recordings to this email. If HDD space is used, the recording will be deleted and sent to this email. If this email is empty, the recording will be sent to the default user email.
 
'''Device –> Edit window'''

The following options are available for Device:

* '''Record calls for this Device''' - record calls for this Device (without user knowledge). If this option is enabled, recordings must be deleted manually.
* '''Show Recordings to the User for this Device''' - allows User to see such recordings
* '''Send Recording to Email''' - when recording is completed, send to it email.
* '''Recordings Email''' - send to this email address.
* '''Keep Recordings on Server after sent to Email''' - should the recording be left on the server after being sent to email?
 
[[File:device_recordings_settings.png]]
 

== Recordings bulk management ==

This feature allows an administrator to quickly delete Recordings for selected period or selected Device
 
==== Who can use this functionality? ====

* This feature is only available for MOR administrator
 
==== How to use this functionality ====
Go to '''ADDONS –> Recordings'''
 
[[File:Recordings_menu_path.png]]
 
Click on [[image:icon_edit.png]] '''Bulk management''' link.
 
[[File:Recordings_bulk_management.png]]
 
Here you can check an option to delete by device or by date, select device/date and click '''Delete'''.
 
[[File:Recordings_bulk_management_confirm.png]]
 
Confirm that you really want to delete recordings of selected period/device.
 

= User settings =

The Administrator must authorize the user to use the Recording Addon functionality before User can use its functions. This can be done by checking '''Allow to use recording functionality''' in the User Edit window.
 
'''PERSONAL MENU –> Details'''

* Server space - shows used/reserved HDD space for recordings.
* Email for deleted Recordings - to which email recordings should be sent if there is not enough space for them.
 
'''PERSONAL MENU –> Details –> Devices –> Edit'''

* Record calls for this Device - user can set to record calls for this device or not.
* Send Recording to Email - should recording be sent to email?
* Recordings Email - the email address email to which recordings should be sent.
* Keep Recordings on Server after sent to Email - should recordings be left on the server after being sent to email?
 
'''PERSONAL MENU –> Calls –> Recordings'''

On this page, the User can manage his recordings.
 
= Guides =

== Setup user so he could see his recorded calls ==

1. As admin, go to '''User Edit''' settings [[Image:icon_edit.png]] and check '''Allow to use recording functionality'''. User will be able to use Recording addon functionality.

2. Log in as this User and in '''PERSONAL MENU –> Details –> Devices''', click [[Image:icon_edit.png]] for Device and in the Recordings section check '''Record calls for this Device: YES.'''

3. Now the user will be able to see his calls in '''PERSONAL MENU –> Call –> Recordings'''.
 

== Delete recordings after sending it via e-mail ==

1. As admin, go to '''User Edit''' settings [[Image:icon_edit.png]] and check '''Allow to use recording functionality'''. User will be able to use Recording addon functionality. Do not check '''Forced record calls for this User''' because it this way recordings will '''NOT''' be deleted.

2. Log in as this User and in '''PERSONAL MENU –> Personal Details –> Devices''', click [[Image:icon_edit.png]] for Device and in the Recordings section check '''Keep Recordings on Server after sent to Email: NO.'''

3. Now recordings will be deleted immediately after they will be sent via e-mail.
 

=See also=

* [[MOR Addons]]

Configuration from GUI

2013-12-27T09:27:43Z

Algirdas: /* Visual */

=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]]
 

== 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'''
* '''Do not allow prepaid user balance drop below zero'''
* '''Logout link''' – a user will be redirected to this link after pressing "Logout"
* '''Change ANSWER to FAILED if HGC not equal to 16 for Users''' – changes the Hangupcause for users in call reports from Answered to Failed if Hangupcause is not 16.
 
== Tax ==

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

Settings are explained 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.
* '''Fax2Email From Sender''' – There is a field "From:" in an email form. When MOR sends Fax to Email, "Fax2Email From Sender" value is puted 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]]'''
* '''[[False_ANSWER#FAS_Control_from_GUI| Kill Call if PDD more/less than]]'''
 

===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
* '''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]]

File:Active calls2.png

2013-12-16T14:39:31Z

Algirdas: uploaded a new version of "File:Active calls2.png"

Voxbeam

2013-12-13T17:51:39Z

Algirdas:

== About ==

This documentation shows how to configure your VoxBeam provider on MOR.

== Configuration on VoxBeam side ==

First in your VoxBeam account you need to add you servers IP and choose the prefix you will use for call routing. Also please check the VoxBeam hostname and aveilable codecs which you will configure in MOR.
See the following screen shot with the instructions regarding VoxBeam configuration:

<div style="text-align: center;">[[File:VoxBeam.png]]</div>

== Configuration on MOR side ==

After configuring VoxBem you need to configure MOR. Create a provider by going to SETTINGS->Billing->Providers and pressing on "Add Provider"
Configure the provider according the example and instructions provided in the following screen shot:

<div style="text-align: center;">[[File:VoxBem_prov.png]]</div>

When the provider is created you still need to create a provider rule adding the specified prefix in VoxBeam to the dialed destination. You can do so by going to SETTINGS->Billing-Providers and pressing on the "Rule" icon
next to the created provider.

<div style="text-align: center;">[[File:Rules1.png]]</div>

Create the rule accourding the instructions provided in the following screen show. The rule created in the screen shot adds 0011101 to any dialed destination.

<div style="text-align: center;">[[File:prefix.png]]</div>

And now you should have a fully configured and working VoxBeam provider on MOR

Voxbeam

2013-12-13T10:10:20Z

Algirdas: Created page with 'First in your VoxBeam account you need to add you servers IP and choose the prefix you will use for call routing. Also please check the VoxBeam hostname and aveilable codecs whic…'

First in your VoxBeam account you need to add you servers IP and choose the prefix you will use for call routing. Also please check the VoxBeam hostname and aveilable codecs which you will configure in MOR.
See the following screen shot with the instructions regarding VoxBeam configuration:

[[File:VoxBeam.png]]

After configuring VoxBem you need to configure MOR. Create a provider by going to SETTINGS->Billing->Providers and pressing on "Add Provider"
Configure the provider according the example and instructions provided in the following screen shot:

[[File:VoxBem_prov.png]]

When the provider is created you still need to create a provider rule adding the specified prefix in VoxBeam to the dialed destination. You can do so by going to SETTINGS->Billing-Providers and pressing on the "Rule" icon
next to the created provider.

[[File:Rules1.png]]

Create the rule accourding the instructions provided in the following screen show. The rule created in the screen shot adds 0011101 to any dialed destination.

[[File:prefix.png]]

File:Rules1.png

2013-12-13T10:10:04Z

Algirdas: rules button location

rules button location

File:Prefix.png

2013-12-13T10:05:45Z

Algirdas: Rule adding 0011101 to any dialed destination

Rule adding 0011101 to any dialed destination

File:Rules.png

2013-12-13T10:02:00Z

Algirdas: uploaded a new version of "File:Rules.png": rules button location

File:VoxBem prov.png

2013-12-13T09:58:27Z

Algirdas: Instructions regarding VoxBeam Configuration on MOR side

Instructions regarding VoxBeam Configuration on MOR side

File:VoxBeam.png

2013-12-13T09:55:12Z

Algirdas: VoxBem configuration example with instructions

VoxBem configuration example with instructions

Extending Fail2Ban configuration

2013-09-15T13:09:11Z

Algirdas: /* About */

= About =
This page explains how to tune/extend Fail2Ban the right way, so that Kolmisoft update scripts would not overwrite your changes.

Please note that in order for these extended rules to work Fail2Ban has to be updated. You can update Fail2Ban by running the following script:

/usr/src/mor/sh_scripts/fail2ban_install.sh

== Fail2Ban configuration options explained ==
===/etc/fail2ban/jail.conf===

For example we will take HGC 200 ban configurations which by default ban IP's sending unauthenticated (devices are not registered to MOR system and IP based authentication is not selected) calls after 20 attempts/ 1 minute:

[ast-hgc-200]
enabled = true
filter = asterisk_hgc_200
action = iptables-allports[name=AST_HGC_200, port=5060, protocol=all]
sendmail-banned[name=Asterisk_HGC_200]
logpath = /var/log/asterisk/messages
maxretry = 20
bantime = -1

'''Options explained'''

'''enabled''' = {true,false} - filter enabled or not.

'''filter''' = asterisk_hgc_200 # Filter name. This name matches file /etc/fail2ban/filter.d/asterisk_hgc_200.conf . This file structure is very simple and is self explaining. It contains patterns the fail2ban has to scan in log files. You have to be able read / write perl regexp syntax in order to modify this file.

'''action''' = iptables-allports[name=AST_HGC_200, port=5060, protocol=all] # Action configuration, it matches file /etc/fail2ban/action.d/iptables-allports

sendmail-banned[name=Asterisk_HGC_200] # Action configuration, it matches file /etc/fail2ban/action.d/sendmail-banned

'''logpath''' = /var/log/asterisk/messages # Which log has to be scanned by filter

'''maxretry''' = 20 # How many attempts we tolerate per minute. If there are more entries in log file than defined maxretry value = actions configuration files defined above are launched.

'''bantime''' = -1 # <0 - ban will be forever (till you restart fail2ban/system). >0 - seconds.

== Writing your own modifications for Fail2Ban==

In order Kolmisoft update scripts would not overwrite your changes - you have to put it in file named the same way except the extension - it has to be named ".local" at the end instead of ".conf".

For example - if you would like to modify main configuration jail.conf - you would have to create your own configuration file named jail.local. Put all your changes there and Fail2Ban will use your configuration instead of Kolmisoft.

Important - you don't have to copy the whole jail.conf file to jail.local, you can just put there only one section you would like to behave differently. For example - default Kolmisoft Fail2Ban configuration file jail.conf has these rules:

[ast-hgc-200]
enabled = true
filter = asterisk_hgc_200
action = iptables-allports[name=AST_HGC_200, port=5060, protocol=all]
sendmail-banned[name=Asterisk_HGC_200]
logpath = /var/log/asterisk/messages
maxretry = 20
bantime = -1

If you are an experienced user and you know that there should not be any unauthenticated calls in your system - it is advised to modify this configuration to look like this:

[ast-hgc-200]
enabled = true
filter = asterisk_hgc_200
action = iptables-allports[name=AST_HGC_200, port=5060, protocol=all]
sendmail-banned[name=Asterisk_HGC_200]

logpath = /var/log/asterisk/messages
maxretry = 1
bantime = -1

You have to put this configuration to '''/etc/fail2ban/jail.local'''. This modification will ensure, that the offender will get ban from the first try.

'''Important'''

Every Fail2Ban configuration can be extended the same way - just create another file with same name with ".local" as file extension.

= Extending Fail2Ban to send you an email =

In order Fail2Ban would send you an email on various events you can extend existing Fail2Ban rules by creating jail.local file and adding modified configuration. For example in order on HGC 200 attack you would get an email - you should extend the configuration of jail.local like this:
[ast-hgc-200]
enabled = true
filter = asterisk_hgc_200
action = iptables-allports[name=AST_HGC_200, port=5060, protocol=all]
sendmail-banned[name=Asterisk_HGC_200]
sendmail[name=Asterisk_HGC_200, dest=your@mail.com]
logpath = /var/log/asterisk/messages
maxretry = 1
bantime = -1

The important line here is this: '''sendmail[name=Asterisk_HGC_200, dest=your@mail.com]'''

Extending Fail2Ban configuration

2013-09-15T13:08:30Z

Algirdas: /* About */

= About =
This page explains how to tune/extend Fail2Ban the right way, so that Kolmisoft update scripts would not overwrite your changes.
Please note that in order for these extended rules to work Fail2Ban has to be updated. You can update Fail2Ban by running the following script:
/usr/src/mor/sh_scripts/fail2ban_install.sh

== Fail2Ban configuration options explained ==
===/etc/fail2ban/jail.conf===

For example we will take HGC 200 ban configurations which by default ban IP's sending unauthenticated (devices are not registered to MOR system and IP based authentication is not selected) calls after 20 attempts/ 1 minute:

[ast-hgc-200]
enabled = true
filter = asterisk_hgc_200
action = iptables-allports[name=AST_HGC_200, port=5060, protocol=all]
sendmail-banned[name=Asterisk_HGC_200]
logpath = /var/log/asterisk/messages
maxretry = 20
bantime = -1

'''Options explained'''

'''enabled''' = {true,false} - filter enabled or not.

'''filter''' = asterisk_hgc_200 # Filter name. This name matches file /etc/fail2ban/filter.d/asterisk_hgc_200.conf . This file structure is very simple and is self explaining. It contains patterns the fail2ban has to scan in log files. You have to be able read / write perl regexp syntax in order to modify this file.

'''action''' = iptables-allports[name=AST_HGC_200, port=5060, protocol=all] # Action configuration, it matches file /etc/fail2ban/action.d/iptables-allports

sendmail-banned[name=Asterisk_HGC_200] # Action configuration, it matches file /etc/fail2ban/action.d/sendmail-banned

'''logpath''' = /var/log/asterisk/messages # Which log has to be scanned by filter

'''maxretry''' = 20 # How many attempts we tolerate per minute. If there are more entries in log file than defined maxretry value = actions configuration files defined above are launched.

'''bantime''' = -1 # <0 - ban will be forever (till you restart fail2ban/system). >0 - seconds.

== Writing your own modifications for Fail2Ban==

In order Kolmisoft update scripts would not overwrite your changes - you have to put it in file named the same way except the extension - it has to be named ".local" at the end instead of ".conf".

For example - if you would like to modify main configuration jail.conf - you would have to create your own configuration file named jail.local. Put all your changes there and Fail2Ban will use your configuration instead of Kolmisoft.

Important - you don't have to copy the whole jail.conf file to jail.local, you can just put there only one section you would like to behave differently. For example - default Kolmisoft Fail2Ban configuration file jail.conf has these rules:

[ast-hgc-200]
enabled = true
filter = asterisk_hgc_200
action = iptables-allports[name=AST_HGC_200, port=5060, protocol=all]
sendmail-banned[name=Asterisk_HGC_200]
logpath = /var/log/asterisk/messages
maxretry = 20
bantime = -1

If you are an experienced user and you know that there should not be any unauthenticated calls in your system - it is advised to modify this configuration to look like this:

[ast-hgc-200]
enabled = true
filter = asterisk_hgc_200
action = iptables-allports[name=AST_HGC_200, port=5060, protocol=all]
sendmail-banned[name=Asterisk_HGC_200]

logpath = /var/log/asterisk/messages
maxretry = 1
bantime = -1

You have to put this configuration to '''/etc/fail2ban/jail.local'''. This modification will ensure, that the offender will get ban from the first try.

'''Important'''

Every Fail2Ban configuration can be extended the same way - just create another file with same name with ".local" as file extension.

= Extending Fail2Ban to send you an email =

In order Fail2Ban would send you an email on various events you can extend existing Fail2Ban rules by creating jail.local file and adding modified configuration. For example in order on HGC 200 attack you would get an email - you should extend the configuration of jail.local like this:
[ast-hgc-200]
enabled = true
filter = asterisk_hgc_200
action = iptables-allports[name=AST_HGC_200, port=5060, protocol=all]
sendmail-banned[name=Asterisk_HGC_200]
sendmail[name=Asterisk_HGC_200, dest=your@mail.com]
logpath = /var/log/asterisk/messages
maxretry = 1
bantime = -1

The important line here is this: '''sendmail[name=Asterisk_HGC_200, dest=your@mail.com]'''