Menu Clients¶
This is the list of all fields with their description of the menu Clients
Users¶
This menu has the following fields
Username¶
Username used to login into the panel.
Password¶
Password used to login into the panel.
Group¶
There are 3 groups: admin, agent and client. You can create more or edit any of these groups. Each group can have specific permissions. Check the menu Configuration->User Group.
Group for agent users¶
Select the group that the clients of this retailer used.
Plan¶
Plan that will be used to charge the clients.
Language¶
Language. This languague is used for some system function, but not for the panel language.
Prefix rules¶
Prefix rules. You can see more details at the link https://www.magnusbilling.org/local_prefix.
Active¶
Only active users can login into the panel and make calls
Country¶
Used to CID Callback. The country prefix code will be added before the CID to convert the CID to E164
Activate offer¶
Used to give free minutes. It’s necessary to inform the tariffs that will belongs to the free packages.
CPS Limit¶
CPS(calls per second) limit to this client. The calls that exceed this limit will be send CONGESTION.
Description¶
We did not write the description to this field.
Company website¶
Company website.|Also used to agent panel customization. To agent, set the domain without http or wwww.
Company name¶
Company name. Also used to agent panel customization.|Whether is a agent this name will be used on the login panel. Need set the compnay website and use the agent domain to working the customization
Commercial name¶
Brand name.
State number¶
State number.
Last name¶
Lastname.
First name¶
Firstname.
City¶
City.
State¶
State.
Address¶
Address.
Neighborhood¶
Neighborhood.
Zip code¶
Zipcode.
Phone¶
Landline phone.
Mobile¶
Mobile phone.
Email¶
Email, it’s necessary to send system notifications.
DOC¶
Client document.
VAT¶
Used in some payment methods.
Contract value¶
We did not write the description to this field.
DIST¶
We did not write the description to this field.
Type paid¶
Pos-paid clients can stay with negative balance until the credit limit informed in the field below.
Credit notification daily¶
Enable this option to customer receive daily balance notification Email. You can customize the email on Configuration menu, submenu Email Templates
Credit limit¶
If the user is Post-paid, the user will be able to make calls until he reaches this limit.
Credit notification¶
If the client credit get lower than this field value, MagnusBilling will send an email to the client warning that he is with low credits. IT’S NECESSARY HAVE A REGISTERED SMTP SERVER IN THE SETTINGS MENU.
Enable expire¶
Activate expire. It’s necessary to inform the expiry date in the “Expiry date” field.
Expiration date¶
The date that the user will not be able to make calls anymore.
Call limit¶
The amount of simultaneous calls allowed for this client.
Limit error¶
Warning to be send if the call limit is exceeded.
Record call format¶
Format used to record calls.
Callshop¶
Activate the CallShop module. Only active if you really are going to use it. It’s necessary give permition to the selected group.
Disk space¶
Insert the amount disk space available to record, in GB. Use -1 to save it without limit. It’s necessary to add in the cron the following php command /var/www/html/mbilling/cron.php UserDiskSpace .
SIP account limit¶
The amount of VoIP accounts allowed by this user. Will be necessary give permission to the group to create VoIP accounts.
CallingCard PIN¶
Used to authenticate the CallingCard.
Restriction¶
Used to restrict dialing. Add the numbers in the menu: Users->Restricted numbers.
Use¶
Which number will be used to check the restriction. This option is valid only to outbound calls.
Profit¶
This function is not avaible in Brazil. It’s only used to mobile refills in some countries.
Profit¶
This function is not avaible in Brazil. It’s only used to mobile refills in some countries.
Profit¶
This function is not avaible in Brazil. It’s only used to mobile refills in some countries.
Enable DBBL/Rocket¶
This function is not avaible in Brazil. It’s only used to mobile refills in some countries.
Profit¶
This function is not avaible in Brazil. It’s only used to mobile refills in some countries.
Show selling price¶
This function is not avaible in Brazil. It’s only used to mobile refills in some countries.
Email¶
We did not write the description to this field.
Services email notification¶
We did not write the description to this field.
DID email notification¶
We did not write the description to this field.
SIP Users¶
This menu has the following fields
Username¶
User that this SIP user is associated with.
SIP user¶
Username used to login in a Softphone or any SIP device.
SIP password¶
Password to login in a Softphone or any SIP device.
CallerID¶
The Caller ID number that will be shown in their destination. Your trunk needs to accept CLI.
Alias¶
Alias to dial between sip accounts from the same AccountCode (company).
Disallow¶
Disallow all codecs and then select the codecs available below to enable them to the user.
Codec¶
Select the codecs that the trunk will accept.
Host¶
Dynamic is an option that allows the user to register their account under any IP. If you want to authenticate the user via IP, put the client IP here, let the password field blank and set it to “insecure” to por/invite in the Aditional Informations tab.
Group¶
Videosupport¶
Activate video calls.
Block call regex¶
Block calls using REGEX. To block calls from cellphones, just put it ^55\d\d9. You can see more details at the link https://regex101.com..
Record call¶
Record calls of this SIP user.
Tech prefix¶
Useful option for when it’s necessary to authenticate more than one client via IP that uses the same IP. Common in BBX multi tenant.
Description¶
We did not write the description to this field.
NAT¶
Nat. You can see more details at the link https://www.voip-info.org/asterisk-sip-nat/.
Directmedia¶
If enabled, Asterisk tries to redirect the RTP media stream to go directly from the caller to the callee.
Qualify¶
Sent the “OPTION” package to verify if the user is online.
Sintax:
qualify = xxx | no | yes
Where the XXX is the number of milliseconds used. If “yes”, the time configurated in sip.conf is used, 2 seconds is the standard.
If you activate “qualify”, the Asterisk will sent the command “OPTION” to SIP peer regulary to verify if the device is still online.
If the device don’t answer the “OPTION” in the set period of time, Asterisk will consider the device offline for future calls.
This status can be verified with the funcion “sip show peer XXXX”, this funcion will only provide informations of status for the SIP peer that possess “qualify = yes.
Trunk groups¶
:::::::WARNING::::::. By selecting a trunk group here, the trunk group will be ignored from tariffs and this trunk group will always be used. Only select a trunk group here if you really want all calls from this SIP user to be sent to this trunk group
Context¶
This is the context that the call will be processed, “billing” is the standard option. Only change configuration if you have knowledge of Asterisk.
Dtmfmode¶
DTMF type. You can see more details at the link https://www.voip-info.org/asterisk-sip-dtmfmode/..
Insecure¶
This option need to be “NO” if the host is dynamic, so the IP authentication changes to port,invite.
Deny¶
You can limit SIP traffic of a determined IP or network.
Permit¶
You can allow SIP traffic of a determined IP or network.
Type¶
Standard type is “friend”, in other words, can make and receive calls. You can see more details at the link https://www.voip-info.org/asterisk-sip-type/..
Allowtransfer¶
Ring false¶
Activate false ring. Add rR of the “Dial” command.
Call limit¶
Maximum simultaneous calls allowed for this SIP user.
MOH¶
Waiting music for this SIP user.
URL events notify¶
.
Addparameter¶
The parameters set in here will replace the system default parameters, as well of the trunks, if there’s any.
AMD¶
.
Forward type¶
Resend destination type. This resend will not work in queues.
IVR¶
Select the IVR that you want to to send to calls if the SIP user don’t answer.
Queue¶
Select the queue that you want to to send to calls if the SIP user don’t answer.
Sip user¶
Select the SIP users that you want to to send to calls if the SIP user don’t answer.
Destination¶
Click for more details
We have 3 options, conform the selected type, group, number or custom.
* Group, the group name set here, needs to be exatcly the same group of SIP users that wants to receive the calls, is going to call all SIP users in the group.
* Custom, it’s possible to execute any valid option of the DIAL command of Asterisk, example: SIP/contaSIP,45,tTr
* Number, can be a landline number or mobile number, needs to be in the 55 DDD format
Dial timeout¶
Timeout in seconds to wait for the call to be picked-up. After the timeout will be execute the channeling if it’s configurated.
Enable voicemail¶
Activate voicemail. It’s necessary the configuration of SMTP in Linux to receive the email with the message. You can see more details at the link https://www.magnusbilling.org/br/blog-br/9-novidades/25-configurar-ssmtp-para-enviar-voicemail-no-asterisk.html..
Email¶
Email that will be send the email with the voicemail.
Parameters¶
We did not write the description to this field.
Peer¶
sip show peer
CNL zone¶
We did not write the description to this field.
Calls Online¶
This menu has the following fields
CallerID¶
This menu has the following fields
Username¶
Select user.
CallerID¶
The number to CID authenticate with CallingCard. Use the exact format that you received the CallerID from your DID provider.
Name¶
Optional.
Description¶
CallerID description.
Status¶
Status of the CallerID.
ATA Linksys¶
This menu has the following fields
Serial¶
LinkSys serial number
MAC¶
LinkSys MAC address
User password¶
Username to login in LinkSys settings
Admin password¶
Password to login in LinkSys settings
Antireset¶
Be cautious.*73738# command prevents resetting LinkSys.
Enable_Web_Server¶
Beware! If deactivated, will not be able to login in the Linksys settings.
Dial Tone¶
We did not write the description to this field.
Proxy¶
Proxy 1.
Username¶
SIP user username that will be used in ATA line 1.
Password¶
SIP user password
Use_Pref_Codec¶
Only use the preferred CODEC
Codec¶
Set the preferred CODEC
Register expires¶
Interval in seconds that LinkSys will send a REGISTER to your server. Useful to avoid a loss of connection when you receive a call.
Dial plan¶
Read LinkSys documentation
NAT Mapping¶
It’s recommended to activate this option if ATA is behind NAT.
NAT keep alive¶
It’s recommended to activate this option if ATA is behind NAT.
Proxy¶
Proxy 2.
Username¶
SIP user username that will be used in ATA line 1.
Password¶
VOIP account password.
Use_Pref_Codec¶
Only use preferencial codec.
Codec¶
Settings of preferincial codec.
Register expires¶
Time in seconds that Linksys sends “REGISTER” to the server. If it’s going to get calls in this line, it’s better set it up between 120 and 480 seconds.
Dial plan¶
Read linksys documentation
NAT Mapping¶
It’s recommended to activate this option if ATA is behind NAT.
NAT keep alive¶
It’s recommended to activate this option if ATA is behind NAT.
Enable STUN¶
Activate STUN server.
STUN Test¶
Validate STUN server periodically..
Substitute VIA Addr¶
Replace publia IP in the VIA.
STUN Server¶
STUN server domain.
Restricted Number¶
This menu has the following fields
Callback¶
This menu has the following fields
Buy Credit¶
This menu has the following fields
Iax¶
This menu has the following fields
Username¶
The user whose IAX account will belong
IAX user¶
The user that will be used to authenticate in the softphone
IAX password¶
The Password that will be used to authenticate in the softphone
CallerID¶
This is the CallerID that will be shown in their destination, in external calls the provider will need to permit CLI to be correctly identified in their destination.
Disallow¶
In this option will be possible to deactivate codecs. To deactivate all the codecs and letting avaible to the user only what you select below, use “Use all”
Codec¶
Codecs that will be accepted.
Host¶
“Dynamic” is an option that will let the user register his account in any IP. If you want to to authenticate the user by their IP, fill here the IP of the client, let the password field blank and put “insecure” for the port/invite in the tab “Additional Information”
NAT¶
The client is behind NAT. You can see more details at the link https://www.voip-info.org/asterisk-sip-nat/..
Context¶
This is the context that the call will be processed, by default is set to “billing”. Only alter if you have knowledge of Asterisk.
Qualify¶
Sent the “OPTION” package to verify if the user is online.
Sintax:
qualify = xxx | no | yes
Where the XXX is the number of milliseconds used. If “yes”, the time configurated in sip.conf is used, 2 seconds is the standard.
If you activate “qualify”, the Asterisk will sent the command “OPTION” to SIP peer regulary to verify if the device is still online.
If the device don’t answer the “OPTION” in the set period of time, Asterisk will consider the device offline for future calls.
This status can be verified with the function “sip show peer XXXX”, this funcition will only provide status informations to the SIP peer that have “qualify = yes”.
Dtmfmode¶
Type of DTMF. You can see more details at the link https://www.voip-info.org/asterisk-sip-dtmfmode/..
Insecure¶
If the host is set to “dynamic”, this option will need to be set to “no”. To authenticate via IP and alter to port. You can see more details at the link https://www.voip-info.org/asterisk-sip-insecure/..
Type¶
Default type is “friend”, in other words they can make and receive calls. You can see more details at the link https://www.voip-info.org/asterisk-sip-type/..
Call limit¶
Total of simultaneous calls allowed for this IAX account.
Send Credit¶
This menu has the following fields
User History¶
This menu has the following fields
Menu Billing¶
This is the list of all fields with their description of the menu Billing
Refills¶
This menu has the following fields
Username¶
User that will be realized the refill.
Credit¶
Refill amount. Can be a positive or negative value, if the value is negative will remove from the total amount of credit of the client.
Description¶
Description to the calendar, only for self control.
Add payment¶
This setting is only to your control, the credit will be released to the user anyway if set to Payment NO
Date¶
We did not write the description to this field.
Invoice number¶
Invoice number.
Payment receipt¶
We did not write the description to this field.
Payment Methods¶
This menu has the following fields
Voucher¶
This menu has the following fields
Credit¶
Voucher price. You can see more details at the link https://wiki.magnusbilling.org/en/source/how_to_use_voucher.html..
Plan¶
Plan that will be linked to the client that will use this VOUCHER.
Language¶
Language that will be used.
Prefix rules¶
Rule that will be used in the “prefix rule” field
Quantity¶
The amount of VOUCHERS to be generated.
Description¶
Description to the calendar, only for self control.
Voucher¶
VOUCHER number.
Menu DIDs¶
This is the list of all fields with their description of the menu DIDs
DIDs¶
This menu has the following fields
DID¶
The exact number coming from the context in Asterisk. We recommend you to always use the E164 format. Also you can create intervals. Example 13605040001-13605040009. In this example will be created the DIDs 13605040001 until 13605040009.
Record call¶
Record calls for this DID. Recorded regardless of destination.
Status¶
Only active numbers can receive calls.
Callerid name¶
Use this field to set a CallerID name or leave it blank to use the received CallerID from the DID provider.
Setup price¶
Activation cost. This value will be deducted from the client the moment that the DID is associated with the user.
Monthly price¶
Monthly price. This value will be deducted automatically every month from the user’s balance. If the client doesn’t have enough credit the DID will be cancelled automatically.
Connection charge¶
This is the value that will be charged for each call. Simply by picking up the call, this value will be deducted.
Minimum time to charge¶
We did not write the description to this field.
Buy price initblock¶
We did not write the description to this field.
Buy price increment¶
We did not write the description to this field.
Minimum time to charge¶
Minimum time to tariff the DID. If you set it to 3 any call that with lower duration will not be charged for.
Initial block¶
Minimum time in seconds to buy. If you set it to 30 and the call duration is 10, the call will be billed as 30.
Billing block¶
This defines the block in which the call billing time will be incremented, in seconds. If set to 6 and call duration is 32, the call will be billed as 36.
Charge who¶
The user that will be charged for the DID cost.
Channel limit¶
Maximum simultaneous calls for this DID.
Description¶
Set here the destination!
Regular expression¶
This is a regular expression to tariff the DID depending on who is calling it.
Lets analyze a real example:
Suppose we want to charge 0.10 when we receive a call from a landline and 0.20 if its a mobile phone and block any other format.
In this example we will create rules to identify the CallerID in the format 0 + area code + number, area code + number, or 55 + area code + number.
Take a look at the following image on what the result would look like:
.. image:: ../img/did_regex.png
:scale: 100%
Regular expression for mobile
^[1-9][0-9]9\d{8}$|^0[1-9][0-9]9\d{8}$|^55[1-9][0-9]9\d{8}$
Regular expression for landline
^[1-9][0-9]\d{8}$|^0[1-9][0-9]\d{8}$|^55[1-9][0-9]\d{8}$
Buy price per min¶
We did not write the description to this field.
Sell price per min¶
Price per minute if the number matches the above regular expression.
Block calls from this expression¶
Set to yes to block calls that matches with the above regular expression.
Send the call to callback¶
Send this call to CallBack if it matches with the above regular expression.
Regular expression¶
Same as the first expression. You can see more details at the link https://wiki.magnusbilling.org/en/source/modules/did/did.html#did-expression-1..
Buy price per min¶
We did not write the description to this field.
Sell price per min¶
Price per minute if the number matches the above regular expression.
Block calls from this expression¶
Set to yes to block calls that matches with the above regular expression.
Send the call to callback¶
Send this call to CallBack if it matches with the above regular expression.
Regular expression¶
Same as the first expression. You can see more details at the link https://wiki.magnusbilling.org/en/source/modules/did/did.html#did-expression-1..
Buy price per min¶
We did not write the description to this field.
Sell price per min¶
Price per minute if the number matches the above regular expression.
Block calls from this expression¶
Set to yes to block calls that matches with the above regular expression.
Send the call to callback¶
Send this call to CallBack if it matches with the above regular expression.
CallBack pro¶
Enables CallBack Pro.
Use audio¶
Execute an audio.
Maximum trying¶
How many times will the system try to return the call?
Interval between trying¶
Time interval between each try, in minutes.
Early media¶
Execute an audio before the call is answered. Your DID provider needs to allow early media.
Mon-Fri¶
Example: if your company only callbacks to the callee if the call was placed in between 09:00-12:00 and 14:00-18:00 MON-FRY, between this time interval the workaudio is going to be played and then callback to the callee. You can use multiple time intervals separated by |.
Sat¶
The same but for Saturday.
Sun¶
The same but for Sunday.
Work audio¶
Audio that will be executed when a call is received at the time interval.
Out work audio¶
Audio that will be executed when a call is received out of the time interval.
Country¶
We did not write the description to this field.
Server¶
We did not write the description to this field.
DID Destination¶
This menu has the following fields
DID¶
Select the DID that you want create new destination for.
Username¶
User that will be the owner of this DID.
Status¶
Only active destinations will be used.
Priority¶
You can create up to 5 destinations for your DID. If a try is made and a error is received, MagnusBilling will try to send the call to the next destination priority available. Only works with the “SIP call” type.
Type¶
Type of destination.
Destination¶
Set here the destination!
IVR¶
Select a IVR to send the call to. The IVR needs to belong to the owner of the DID aswell.
Queue¶
Select a Queue to send the call to. The Queue needs to belong to the owner of the DID aswell.
Sip user¶
Select a SIP user to send the call to. The SIP user needs to belong to the owner of the DID aswell.
Context¶
In this field you may use a context in the format supported by Asterisk
Example:
_X. => 1,Dial(SIP/sipaccount,45)
same => n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-CONGESTION,1,Congestion
exten => s-CANCEL,1,Hangup
exten => s-BUSY,1,Busy
exten => s-CHANUNAVAIL,1,SetCallerId(4545454545)
exten => s-CHANUNAVAIL,2,Dial(SIP/sipaccount2,,T)
You should NOT set a name for the context because the name will be set automatically as [did-number-of-the-did]
You may take a look at the context at /etc/asterisk/extensions_magnus_did.conf
DIDs Use¶
This menu has the following fields
IVRs¶
This menu has the following fields
Name¶
Name of the IVR
Username¶
User who owns the IVR
MonFri intervals¶
Weekly interval of attendance, can be configurated with multiples shifts.
Example:
Supposing that the attendance hours are 08h to 12h and 14h to 19h. In this case the rule would be
08:00-12:00|14:00-19:00
Saturday intervals¶
Interval of attendance in saturdays, can be configurated with multiple shifts
Example:
Supposing that the attendance hours in the saturdays are 08h to 13h. In this case the rule would be
08:00-13:00
Sunday intervals¶
Interval of attendance in sundays, can be configurated with multiple shifts
Example:
Supposing that theres no attendance hours in the sundays. In this case the rule would be
00:00-00:00
Use holidays¶
If this option is activated then the system will check if there is a holiday registered for the day, if so, then the audio, not working, will be played.
Work audio¶
Audio to play in the attendance hours.
Out work audio¶
Audio to play when it’s not attendance hours
Option 0¶
Select the destination if the option 0 is pressed. Let it in blank if don’t want any action
Option 1¶
Select the destination if the option 1 is pressed. Let it in blank if don’t want any action
Option 2¶
Select the destination if the option 2 is pressed. Let it in blank if don’t want any action
Option 3¶
Select the destination if the option 3 is pressed. Let it in blank if don’t want any action
Option 4¶
Select the destination if the option 4 is pressed. Let it in blank if don’t want any action
Option 5¶
Select the destination if the option 5 is pressed. Let it in blank if don’t want any action
Option 6¶
Select the destination if the option 6 is pressed. Let it in blank if don’t want any action
Option 7¶
Select the destination if the option 7 is pressed. Let it in blank if don’t want any action
Option 8¶
Select the destination if the option 8 is pressed. Let it in blank if don’t want any action
Option 9¶
Select the destination if the option is pressed. Let it in blank if don’t want any action
Default option¶
Select the destination if none of the options was selected.
Enable known SIP user¶
Activating this option will be able to type an SIP user to call it directly.
Option 0¶
Select the destinationif the option 0 is pressed. Let it in blank if don’t want any action
Option 1¶
Select the destination if the option 1 is pressed. Let it in blank if don’t want any action
Option 2¶
Select the destination if the option 2 is pressed. Let it in blank if don’t want any action
Option 3¶
Select the destination if the option 3 is pressed. Let it in blank if don’t want any action
Option 4¶
Select the destination if the option 4 is pressed. Let it in blank if don’t want any action
Option 5¶
Select the destination if the option 5 is pressed. Let it in blank if don’t want any action
Option 6¶
Select the destination if the option 6 is pressed. Let it in blank if don’t want any action
Option 7¶
Select the destination if the option 7 is pressed. Let it in blank if don’t want any action
Option 8¶
Select the destination if the option 8 is pressed. Let it in blank if don’t want any action
Option 9¶
Select the destination if the option 9 is pressed. Let it in blank if don’t want any action
Default option¶
Select the destination if none of the options was selected.
Queues¶
This menu has the following fields
Username¶
User that owns the queue.
Name¶
Queue name.
Language¶
Queue language.
Strategy¶
Queue strategy.
Ringinuse¶
Call or not the agents of the queue that are in call.
Ring for¶
How long the phone will ring until timeout
Time for another agent¶
The amount of time in seconds that will retry the call.
Time for another call¶
Time in seconds until the agent can receive another call.
Weight¶
Queue Priority.
Periodic announce¶
A set of periodic announcements can be created by separating each announcements to reproduce whit commas. E.g.: queue-periodic-announce,your-call-is-important,please-wait. This data need to be in /var/lib/asterisk/sounds/ directory.
Frequency¶
How often to make a periodic announcement.
Announce position¶
Informs the postition in the queue.
Announce holdtime¶
Should we include an estimated hold time in the position announcements?
Announce frequency¶
How often to announce queue position and/or estimated holdtime to caller 0=off
Join empty¶
Allow calls when theres no one to answer the call.
Leave when empty¶
Hang the calls in queue when there’s no one to answer.
Max wait time¶
Maximum wait time on the queue
Max wait time action¶
SipAccount, IVR, QUEUE or LOCAL channel to send the caller if the maximum wait time is reached. Use: SIP/sip_account, QUEUE/queue_name, IVR/ivr_name OR LOCAL/extension@context.
Ring or playing MOH¶
Play waiting music or tone when the client is in the queue.
Audio musiconhold¶
Import one waiting music to this queue.
Queues Members¶
This menu has the following fields
Buy DID¶
This menu has the following fields
Queue DashBoard¶
This menu has the following fields
DIDww¶
This menu has the following fields
DID History¶
This menu has the following fields
Menu Rates¶
This is the list of all fields with their description of the menu Rates
Plans¶
This menu has the following fields
Tariffs¶
This menu has the following fields
Plan¶
The plan that you want to create a tariff for.
Destination¶
The prefix that you want create a tariff for.
Trunk groups¶
The group of trunks that will be used to send this call.
Sell price¶
The amount that you want to charge per minute.
Initial block¶
Minimum time in seconds to buy. E.g., if set to 30s and the call duration is 21s, will be charged for 30s.
Billing block¶
This defines how the time is incremented after the minimum. E.g, if set to 6s and call duration is 32s, will becharged for 36.
Minimum time to charge¶
Minimun time to tariff. If it’s set to 3, will only tariff calls when the time is equal or more than 3 seconds.
Additional time¶
Aditional time to add to all call duration. If it’s set to 10, will be added 10 seconds to all call time duration, this affects tarrifs.
Connection charge¶
We did not write the description to this field.
Include in offer¶
Set to yes if you want to include this tariff to a package offer.
Status¶
Deactivating Tariffs, MagnusBilling will completely desconsider this tariff. Therefore, deleting or deactivating will have the sam effect.
Prefixes¶
This menu has the following fields
Prefix¶
Prefix code. Prefix will be used to tariff and bill the calls.
Destination¶
Destination name.
User Custom Rates¶
This menu has the following fields
Offers¶
This menu has the following fields
Name¶
Free package name
Package type¶
Type of package, there’s 3 types. Unlimited calls, free calls or free seconds.
Free time to call¶
In this field is where the package avaible quantity configuration will occur.
Example:
* Unlimited calls: In this option the field is blank, because will be allowed to call without any control.
* Free calls: Configure the amount of free calls that you want to give.
* Free seconds: Configure the amount of seconds that you want to allow the client to call.
Billing type¶
This is the period that the package will be calculated.
Look the description:
* Monthly: The system will verify the day of the plan activation + 30 days that the client reached the package limit.
* Weekly: The system will verify the day of the plan activation + 7 days that the client reached the package limit.
Price¶
Price that will be charged monthly to the client.
If on the expiry day the client don’t have the sufficient funds to pay the package MagnusBilling will automatically cancel the package.
In the settings menu, ajusts, exist one option named Package Offer Notification, this value means how many days are left until the expiration of the package, the system will try to charge the subscription, if the client don’t have the balance, MagnusBilling will send an email to the client informing the lack of funds.
The email can be edited in the menu, Email models , type, plan_unpaid, Expiry of Monthly Plan Notice subject.
To send emails it’s necessary the configuration of SMTP in the SMTP menu.
To learn how free packages works: https://wiki.magnusbilling.org/en/source/offer.html.
Initial block¶
Minimum time in seconds to sell. This value will subscribe the tariffs of the client’s plan.
Billing block¶
This defines how the time is incremented after the minimum. This value will subscribe the tariffs of the client’s plan.
Minimum time to charge¶
Minimun time to tariff. If it’s set to 3, will only tariff calls when the time is equal or more than 3 seconds.
Offer CDR¶
This menu has the following fields
Offer Use¶
This menu has the following fields
Menu Reports¶
This is the list of all fields with their description of the menu Reports
CDR¶
This menu has the following fields
Date¶
Start time of the call
Sip user¶
SIP user that made the call
CallerID¶
Number sent to the trunk as the identifier of the call.
If the trunk accepts the sent CallerID, then this number will be used as the identifier.
In order to this work its going to be necessary to have the Fromuser field in the trunk empty.
Number¶
Number dialed by the client.
Destination¶
Name of the destination, this name is a relation to the prefix menu.
Username¶
User that made the call, the one who the call cost was taken from.
Trunk¶
Trunk that was used to complete the call.
Duration¶
Duration of the call in seconds.
Buy price¶
Buy cost. You can see more details at the link https://wiki.magnusbilling.org/en/source/price_calculation.html..
Sell price¶
Sell price, the value that was taken from the client. You can see more details at the link https://wiki.magnusbilling.org/en/source/price_calculation.html..
Sell price¶
Sell price, the value that was taken from the client. You can see more details at the link https://wiki.magnusbilling.org/en/source/price_calculation.html..
Uniqueid¶
Unique ID generated by Asterisk, this field is also the start time of the call in Epoch Unix.
CDR Failed¶
This menu has the following fields
Summary per Day¶
This menu has the following fields
Summary Day User¶
This menu has the following fields
Summary Day Trunk¶
This menu has the following fields
Summary Day Agent¶
This menu has the following fields
Summary per Month¶
This menu has the following fields
Summary Month User¶
This menu has the following fields
Summary Month Trunk¶
This menu has the following fields
Summary per User¶
This menu has the following fields
Summary per Trunk¶
This menu has the following fields
Call Archive¶
This menu has the following fields
Send Credit Summary¶
This menu has the following fields
Summary Month DID¶
This menu has the following fields
Menu Routes¶
This is the list of all fields with their description of the menu Routes
Providers¶
This menu has the following fields
Name¶
Provider name
Credit¶
The amount of credit you have in your provider’s account. This field is optional.
Credit control¶
If you set to YES and your provider credit is < 0, all trunks from this provider will be deactivated.
Description¶
Description to the calendar, only for self control.
Trunks¶
This menu has the following fields
Provider¶
Provider which the trunk belongs.
Name¶
Trunk name, must be unique.
Username¶
Only used if the authentication is via username and password.
Password¶
Only used if the authentication is via username and password.
Host¶
IP or Trunk domain.
Add prefix¶
Add a prefix to send to your trunk.
Remove prefix¶
Remove a prefix to send to your trunk.
Codec¶
Select the codecs that are allowed in this trunk.
Provider tech¶
You need install appropriate drive to use card like dgv extra Dongle.
Status¶
If the trunk is inactive, Magnusbilling will sent the call to the backup trunk.
Go to backup if 404¶
Send call to the next trunk if receive error 404.
Register trunk¶
Only active this if the trunk is authenticated via username and password.
Register string¶
<user>:<password>@<host>/contact.
“user” is the user ID for this SIP server (ex 2345).
“password” is the user password
“host” is the SIP server domain or host name.
“port” send an solicitation of the register to this host port. Standard for 5060
“contact” is the extension of Asterisk contact. Example 1234 is set in the contact header of the SIP register message. The contact ramal is used by the SIP server remotely when it’s needed to send one call to Asterisk.
Fromuser¶
Several providers demand this option to authenticate, primarly when it’s authenticated via user and paswword. Let it blank to send the CallerID of the SIP user of From.
Fromdomain¶
Defines the FROM domain: in the SIP messages when act like a UAC SIP (client).
Language¶
Default launguage used in any Playback()/Background().
Context¶
Only change this if you know what you are doing.
Dtmfmode¶
DMTF type. You can see more details at the link https://www.voip-info.org/asterisk-dtmf/..
Insecure¶
Insecure. You can see more details at the link https://www.voip-info.org/asterisk-sip-insecure/..
Max use¶
Maximum simultaneous calls for this trunk.
NAT¶
Is the trunk behind NAT. You can see more details at the link https://www.voip-info.org/asterisk-sip-nat/..
Directmedia¶
If activated, Asterisk will try to send the RTP media directly between your client and provider. It’s necessary to active on the trunk as well. You can see more details at the link https://www.voip-info.org/asterisk-sip-canreinvite/..
Qualify¶
Sent the “OPTION” package to verify if the user is online.
Sintax:
qualify = xxx | no | yes
Where the XXX is the number of milliseconds used. If “yes”, the time configurated in sip.conf is used, 2 seconds is the standard.
If you activate “qualify”, the Asterisk will sent the command “OPTION” to SIP peer regulary to verify if the device is still online.
If the device don’t answer the “OPTION” in the set period of time, Asterisk will consider the device offline for future calls.
This status can be verified with the funcion “sip show peer XXXX”, this funcion will only provide informations of status for the SIP peer that possess “qualify = yes.
Type¶
Default type is “friend”, in other words they can make and receive calls. You can see more details at the link https://www.voip-info.org/asterisk-sip-type/..
Disallow¶
In this option is possible to deactivate codecs. Use “Use all” to deactive all codects and make it avaible to the user only what you selected below.
Sendrpid¶
Defines if one Remote-Party-ID SIP header task to be send.
The default is “no”.
This field is frequently used by VoIP wholesalers providers to supply the callers identity, independently of the privacy settings (From SIP header).
Addparameter¶
These parameters will be added in the final AGI command - Dial command, where is in the ajust settings menu.
By default the DIAL command is:
,60,L(%timeout%:61000:30000)
Let’s say that you wanted to add an MACRO in the trunk, therefore in this field you will add the parameter, set it up M(macro_name) and create your MACRO in the Asterisk extensions.
Port¶
If you want to use a different port than 5060, you will need open the IPTABLES port.
Link SMS¶
URL to send SMS. Replace the number variable to %number% and text per %text%. EXAMPLE. Your SMS URL is http://trunkWebSite.com/sendsms.php?user=magnus&pass=billing&number=XXXXXX&sms_text=SSSSSSSSSSS. replace XXXXXX per %number and SSSSSSSSSSS per %text%
SMS match result¶
Leave it blank to not wait the provider answer. Or write the text that needs to consist in the providers answer to be considered SENT.
Parameters¶
Valid format of Asterisk sip.conf, one option per line.
Example, let’s say that you need to put the useragent parameter, so put it in this field:
useragent=my agent
.
Enable CNL¶
We did not write the description to this field.
Trunk Groups¶
This menu has the following fields
Provider Rates¶
This menu has the following fields
Servers¶
This menu has the following fields
Name¶
Server name.
Server IP¶
Server IP. You can see more details at the link https://magnussolution.com/br/servicos/auto-desempenho/servidor-slave.html..
Public IP¶
Public IP.
Username¶
User to connect to the server.
Password¶
Password to connect to the server.
Port¶
Port to connect to the server.
SIPport¶
SIP port that the server will use.
Type¶
Server type.
Weight¶
This option is to balance the calls by weight.
Example.
Let’s say there’s 1 MagnusBilling server and 3 slave servers, and you want to send the double of calls to each slave, proporcionaly to the MagnusBilling server. Then just set the MagnusBilling server to weight 1, and for the slave servers weight 2.
Status¶
The proxy will only send calls to active servers and with weight higher than 0.
Description¶
Used for internal control.
Trunk Errors¶
This menu has the following fields
Provider CNL¶
This menu has the following fields
Menu Settings¶
This is the list of all fields with their description of the menu Settings
Menus¶
This menu has the following fields
Text¶
Menu name
IconCls¶
Icon, default font “awesome V4”.
Main menu¶
Menu which this menu belongs. In case the menu is blank, it’s a main menu
Order¶
Order that the menu will be shown in the menu
Group Users¶
This menu has the following fields
Configuration¶
This menu has the following fields
Value¶
Value. You can see more details at the link https://wiki.magnusbilling.org/en/source/config.html..
Description¶
Description. You can see more details at the link https://wiki.magnusbilling.org/en/source/config.html.
Emails Templates¶
This menu has the following fields
Log Users¶
This menu has the following fields
SMTP¶
This menu has the following fields
Host¶
SMST domain
You need to verify if the datacenter where the server will be hosted don’t block the ports used by SMTP.
Username¶
Username used to authenticate the SMTP server.
Password¶
Password used to authenticate the SMTP server.
Port¶
Port used by the SMTP server.
Encryption¶
Encryption type.
Fail2ban¶
This menu has the following fields
IP¶
IP Address.
Perm ban¶
With this option marked on YES, the IP will be placed on the ip-blacklist list of fail2ban and will be blocked forever.
The option will NOT block the IP momentarily according the parameters of the file /etc/fail2ba/jail.local.
By default the IP is going to stay blocked for 10 minutes
Description¶
These informations are captured from the log file /var/log/fail2ban.log
It’s possible to track this LOG with the command
tail -f /var/log/fail2ban.log
API¶
This menu has the following fields
Username¶
You need use the MagnusBilling API from https://github.com/magnussolution/magnusbilling-api-php. The username owner this API
Api key¶
This apy key will be necessary to execute the API
Api secret¶
This apy secret will be necessary to execute the API
Status¶
You can activete or inactivate this API
Permissions¶
Which action the user will have execute
Restriction IPs¶
What IPS you want allow access this API. Leave blank to allow any IP. It is very recomended set the IPS
Dashboard¶
This menu has the following fields
Call per Minutes¶
This menu has the following fields
Extra2¶
This menu has the following fields
Group to Admins¶
This menu has the following fields
Extra3¶
This menu has the following fields
Backup¶
This menu has the following fields
Alarms¶
This menu has the following fields
Type¶
We did not write the description to this field.
Period¶
We did not write the description to this field.
Condition¶
We did not write the description to this field.
Amount¶
We did not write the description to this field.
Email¶
We did not write the description to this field.
Status¶
We did not write the description to this field.
Subject¶
We did not write the description to this field.
Message¶
We did not write the description to this field.
Extra¶
This menu has the following fields
Menu Voice Broadcasting¶
This is the list of all fields with their description of the menu Voice Broadcasting
Campaigns¶
This menu has the following fields
Username¶
User that owns the campaign.
Plan¶
What plan do you want to use to bill this campaign?
Name¶
Name of the campaign.
CallerID¶
We did not write the description to this field.
Status¶
Status of the campaign.
Starting date¶
The campaign will start from this date.
Expiration date¶
The campaign will stop in this date.
Type¶
Choose Voice or SMS. If you choose Voice you will need to import audio. If you choose SMS you will need to set the text in the SMS tab.
Audio¶
Available to massive calling. The audio needs to be compatible with Asterisk. The recomended format is GSM or WAV(8k hz mono).
Audio 2¶
If you use TTS, the name will be executed between Audio and Audio2.
Restrict phone¶
Activating this option, MagnusBilling will check if the number that will be sent the call is registered in the Restrict Phone menu, if it has, the system will change the status of the number to blocked and will not send the call.
Auto reprocess¶
If there are no active numbers in this campaign phone book, reactivates all pending numbers.
Select one or more phonebooks to to be used.
Number to forward¶
Do you want to forward the call after the audio? E.g, if the callee presses 1, he gets sent to SIP user XXXX. Set Number to Forward = 1, Forward Type = SIP and select the SIP user to send the callee to. Set -1 to disable.
Forward type¶
Choose the type of redirect. This will send the call to the chosen destination.
IVR¶
Choose a IVR to send the call to. The IVR needs to belong to the owner of the campaign.
Queue¶
Choose a Queue to send the call to. The Queue needs to belong to the owner of the campaign.
Sip user¶
Choose a SIP user to send the call to. The SIP user needs to belong to the owner of the campaign.
Destination¶
Record call¶
Record the calls of the campaign. They only will be recorded if the call is transferred.
Daily start time¶
Time that the campaign will start sending.
Daily stop time¶
Time that the campaign will stop sending.
Monday¶
Activating this option the system will send calls on Mondays.
Tuesday¶
Activating this option the system will send calls on Tuesdays.
Wednesday¶
Activating this option the system will send calls on Wednesdays.
Thursday¶
Activating this option the system will send calls on Thursdays.
Friday¶
Activating this option the system will send calls on Fridays.
Saturday¶
Activating this option the system will send calls on Saturdays.
Sunday¶
Activating this option the system will send calls on Sundays.
Call limit¶
How many numbers will be processed per minute?
This value will be divided by 60 seconds and the calls will be sent every minute at the same time.
Maximum call limit¶
This is the maximum value that the client will be able to set. If you set it to 50 the user will be able to change to any value that is 50 or less than 50.
Audio duration¶
Used to control the max completed calls.
Toggle max completed calls¶
If activated MagnusBilling will check how many calls were already made and have a duration total bigger than the audios. If the quantity is equal or bigger than the value set in the field, the campaign will be deactivated.
Max completed calls¶
Maximum amount of complete calls. You need to activate the field above to use this.
Description or SMS Text¶
This field has different uses if the campaign is sending Voice or SMS.
Uses:
* Voice: This field is simply a description of the campaign.
* SMS: The text in here is going to be sent to the numbers. You may use the var %name% where you want to use the name of the customer. Example:
Hello %name%
Audio 1 TTS¶
With this setting the system will generate the audio 1 for the campaign via TTS.
In order for this to work, you will need to set the TTS URL under Settings, Configuration, TTS URL.
Audio 2 TTS¶
Same setting as the previous field but for audio 2. Keep in mind that in between audio 1 and 2, the TTS executes the name imported with the number.
Phonebooks¶
This menu has the following fields
Phonenumbers¶
This menu has the following fields
Polls¶
This menu has the following fields
Polls Reports¶
This menu has the following fields
Restrict Phone¶
This menu has the following fields
SMS¶
This menu has the following fields
Username¶
User that sent/received the SMS.
Number¶
Number in the E164 format.
SMS¶
SMS text.
From¶
If your SMS provider accepts the submission of FROM, put it here. This value will be replaced for the variable %from% in the trunk URL.
Provider result¶
We did not write the description to this field.
Quick Campaign¶
This menu has the following fields
Campaigns DashBoard¶
This menu has the following fields
Campaign Report¶
This menu has the following fields
Menu CallShop¶
This is the list of all fields with their description of the menu CallShop
Booths¶
This menu has the following fields
Booths Report¶
This menu has the following fields
Booths Tariffs¶
This menu has the following fields
Summary per Day¶
This menu has the following fields
Menu Summary Month Trunk¶
This is the list of all fields with their description of the menu Summary Month Trunk