This is very useful screen , it show you some statistics for last 24 hours like ( ACD , ASR , Connected calls count , and failed calls count) , also the calls traffic will appear in "call live monitoring" section , in this section the system will show the last 100 calls with their details , also during running this page , all incoming calls will appear directly "in real time" , the calls are sorted descending based on call date and time , the new calls will be appear at the top

You can click on any call to see call details , call flow , and errors if any

Each node color mean something :

• The call is trying to connect
• The call is not answered
• The call is answered , and still connected
• The call is answered , and ended
• The call is rejected for some reason , you can click/expand call node to see the details

To allow your clients to send calls traffic to your SIP-Electron server , you have to create record on the system for each one of them , clients are two types , [1] Retail Client - SIP Account , [2] Wholesale Client - IP Based , in this section we are talking about retail clients

Retail Client - SIP Account : your client here will use SIP account info to send calls to your server , authentication here will be done using username and password

To add retail client , please follow the steps as shown in the videos below :

Step #1 Add tariff for your retail client for billing and revenue calculation purposes [Mandatory Step]

Step #2 Add your retail client information [Mandatory Step] , the fields are explained below

Client Name : your retail client name , nickname , or any name , this field will not be used in authentication process

SIP Username : username for your retail client , this field will be used in authentication process

SIP Password : password for your retail client , this field will be used in authentication process

Calls Limit : default 0 unlimited , or change it to any other number as needed

Tariff : choose the created tariff in step #1 , for billing and revenue calculation purposes

Filter Traffic Rule : you can create filter rule to filter your client calls based on dialed number , caller id , or based on other factors

Active : this field is to activate or deactivate your client , if you deactivate your client , he will not be able to send calls

Remaining Balance : credit limit that you want to allow your client to use it

Codecs : you can enable disable codecs as needed

Calculate Costs : If you want to send your calls traffic to your service provider through SIP account you can activate this option to calculate your provider costs and your net profit

To allow your clients to send calls traffic to your SIP-Electron server , you have to create record on the system for each one of them , clients are two types , [1] Retail Client - SIP Account , [2] Wholesale Client - IP Based , in this section we are talking about wholesale clients

Wholesale Client - IP Based : your client here will send traffic from his IP - to your server IP directly , authentication here will be done using your client IP address

To add wholesale client , please follow the steps as shown in the videos below :

Step #1 Add tariff for your wholesale client for billing and revenue calculation purposes [Mandatory Step]

Step #2 Add your wholesale client information [Mandatory Step] , the fields are explained below

Client Name : your retail client name , nickname , or any name , this field will not be used in authentication process

Calls Limit : default 0 unlimited , or change it to any other number as needed

Tariff : choose the created tariff in step #1 , for billing and revenue calculation purposes

Filter Traffic Rule : you can create filter rule to filter your client calls based on dialed number , caller id , or based on other factors

Active : this field is to activate or deactivate your client , if you deactivate your client , he will not be able to send calls

Remaining Balance : credit limit that you want to allow your client to use it

IP Address : your client IP address , if your client send calls from another IP address the calls will be rejected

Codecs : you can enable disable codecs as needed

Destination is where you want to send the calls to , destinations are two types :

1) Destination IP : SIP-Electron will send calls to static IP address

2) Retail Client - SIP Account : this way is used when the destination does not have static ip , here SIP-Electron can send calls to registered SIP account , for more information about creating retail account , you can check retail clients section.

The video below demonstrate how to create Destination IP on SIP-Electron server , the fields are explained below

Destination Name : your provider name , gateway name , or any name

IP Address : your provider or gateway IP address

Port : port number used in SIP communications , default port usually used is 5060

Calls Limit : default 0 unlimited , or change it to any other number as needed

Active : this field is to activate or deactivate your gateway or provider , if you deactivate your gateway , SIP-Electron will not send the calls to this destination

Username : the username field is optional , it's mandatory if your gateway or provider use username and password for authentication process with your IP address , usually authentication process is done only using your IP address

Password : the password field is optional , it's mandatory if your gateway or provider use username and password for authentication process with your IP address , usually authentication process is done only using your IP address

Codecs : you can enable disable codecs as needed

Calculate Costs : You can activate this option to calculate your provider costs , and your net profit

After you create clients , and destinations , you need to create a dialing plan , the dialing plan is needed to route incoming calls traffic from your client to specific destination by using call prefix

One dialing plan can be used by many clients , if you have multiple clients sending calls to the same prefix and destination , then one dialing plan is needed only

You can create many dialing plans for same prefix , the system will use them based on their priority , if the call rejected for any reason on the first dialing plan the system will re-route the call to next dialing plan

Example : if your clients are sending traffic with prefix 1122xxxxxx , you have to create a dialing plan to let the soft-switch to route the incoming calls that starting with prefix 1122 to the assigned destination

The video below demonstrate how to create a dialing plan on SIP-Electron server , the fields are explained below

Prefix : your client prefix that you want to link it with destination

Priority : if you want to create many dialing plans for the same prefix , you need to set priority for each dialing plan , the system will use the created dialing plans for the same prefix based on their priorities

Calls Limit : default 0 unlimited , or change it to any other number as needed

Route Type : if you want to route calls to static IP address , then choose destination IP , if your destination doesn't have static IP address then choose retail client

Destination Name : choose any destination you want , to create a destination you can check how to create destination IP section

Remove Prefix : if you want to strip prefix and pass the call to your gateway or to your provider without prefix , you can here add prefix to remove it from dialed number

Add Prefix : if you want to append prefix for dialed numbers you add it here

Http Lookup URL : you can here use external web service (API) to validate your dialed numbers before pass it to your destination , to use this feature , just add web service URL , the service should return number as plain text , then SIP-Electron will pass the returned number to your destination

Example : let suppose that you have API provider (http://api.numbers.com/) to check and validate numbers before sending them to your destination , you can put your api provider url in lookup field , the api provider will receive the dialed number and other call info in [query string] like this http://api.numbers.com?dialednumber=1122334455 , he will check the number and return the same number if the number is valid and will return any other message if the number is not valid to reject the call

Client calls traffic sometimes is not clean 100% , and sometimes the calls are not generated by human , so you need "Filter Traffic Rules" feature to filter junk/spam calls , this feature increase your destination/gateway performance and protect your gateways and SIM cards from suspicious numbers

This feature allow you to pass/reject incoming calls from client to your destination using conditions based on some factors

Examples :

- You can reject calls that repeated quickly to same dialed number

- Pass calls based on static whitelist or reject calls based on static blacklist

- Reject short duration calls

- Reject numbers that have incorrect number of digits (length)

- Caller Id conditions

The video below demonstrate how to create "Filter Traffic Rules" on SIP-Electron server , the fields are explained below

Check dialed number length : If you enable this option , the server will pass the calls if the dialed numbers length within the set range

Check caller id length : If you enable this option , the server will pass the calls if the caller id length within the set range

Accept (dialed number) specific prefixes only : you can set some prefixes to pass them , you can add many prefixes here , one prefix on each line

Block (dialed number) specific prefixes only : you can set some prefixes to reject them , you can add many prefixes here , one prefix on each line

Caller Id format : you can choose caller id format that you want to accept calls from it

Accept dialed number if it's exists on whitelist : if you enable this option , you can upload whitelist , and calls will be passed only if dialed number exists in this list

Block dialed number if it's exists on blacklist : if you enable this option , you can upload blacklist , and calls will be rejected only if dialed number exists in this list

Accept caller id if it's exists on whitelist : if you enable this option , you can upload whitelist , and calls will be passed only if caller id exists in this list

Block caller id if it's exists on blacklist : if you enable this option , you can upload blacklist , and calls will be rejected only if caller id exists in this list

Block caller id by attempts in time : This is dynamic filtering technique , you can reject calls in real time if the caller id is repeated multiple times during a certain time

Block caller id by attempts and call duration : This is a dynamic filtering technique , you can reject calls in real time if the caller id is repeated many times and the duration of the calls is less than the set value

Block dialed number by attempts in time : This is dynamic filtering technique , you can reject calls in real time if the dialed number is repeated multiple times during a certain time

Block dialed number by attempts and duration : This is a dynamic filtering technique , you can reject calls in real time if the dialed number is repeated many times and the duration of the calls is less than the set value

Go to "Connected Calls (CDR)" page , insert date and time from , date and time to , you can select specific route or all , also you can select specific client or all , then click fetch or export

Go to "Failed Calls" page , insert date and time from , date and time to , you can select specific route or all , also you can select specific client or all , then click fetch

Go to "Statistics" page , insert date and time from , date and time to , you can select specific route or all , also you can select specific client or all , then click fetch