StarTrinity VoIP Softswitch Tutorial

• Database architecture diagram
  • Runtime operations and commands, dynamically calculated fields
• Database entities and system concepts
  • Originators
  • Invoices
  • Service plans
  • Tariffs / Destination sets
    • Tariff rates / Routes
  • Suppliers
  • Terminators
  • Routing groups
    • Routing group items
  • Audio files
  • Number lists
• Web interface
  • Web pages to configure database entities
  • Status
  • CDR
    • Call details
    • CDR statistics
  • Destination set details
  • CDR to number lists
  • System status and maintenance
  • Settings
• Default call processing steps
• Installation
  • Setting up the VoIP softswitch on your own Windows server
  • Setting up the VoIP softswitch for the first call - basic steps
  • Setting up connection to GoIP GSM gateway in single server mode
• Web API
• Dialer campaigns (call generation module) - step by step tutorial
• Software architecture (modules)
• Change log
• Blog
• Video tutorials/presentations
• CallXML scripts for the softswitch

Database architecture diagram

Runtime operations and commands, dynamically calculated fields

Database entities and system concepts

Web interface

Web pages to configure database entities

The softswitch web interface contains web pages to configure all entities listed above - add new record, view existing records, edit record, delete record. Here is screenshot for originators:

Status

The status screen displays live information about current calls, originators, terminators

CDR

The CDR screen displays call history: list of calls with CDR fields and links to call details page. Both A and B (incoming and outgoing direction) call legs are displayed in the CDR. The calls are sorted by destroyed time, descending (newest to oldest). There are various filters for the displayed calls. CDR fields are described directly in web interface and in SIP tester tutorial.

Call details

The call details page is accessible from CDR screen, it contains routing/billing log (used for troubleshooting), SIP trace, CDR fields, links to download WAV recordings (see screenshot below)

CDR statistics

The web page is used to calculate CDR statistics for a specified period of time: number of minutes billed, total sales, total cost, total profit, ACD, ASR, statistics about actual number of channels

Destination set details

The page is used to manage termination-side rates. You can download rate sheet as CSV, upload from CSV, apply filter to displayed rates, make bulk modifications (add margin).

The CSV upload procedure works in this way:

• Upload new CSV file, parse lines, for every line do this:
  • If the uploaded prefix does not exist in current list - add it into the list
  • If the uploaded prefix already exists in list - overwrite it. If the new uploaded line has non-empty "effective date" field, and the effective date is in future - keep current rate and use field "next rate" for new uploaded rate
• Remove prefixes which do not exist in new uploaded file (in case if the supplier is no longer able to terminate the removed prefixes)

Similar web page is used to manage tariffs.

CDR to number lists

The screen is used to analyse numbers in CDR and build blacklists/whitelists to filter VoIP traffic and reduce SIM blocking. For example you can analyse CDR for previous 2 years and build list of B numbers that have been dialed with no answer, or very low ASR or ACD. Such B numbers can be used in originator or terminator settings as blacklist, to avoid spam calls. Another example: you can analyse your previous CDRs to select pairs of A and B numbers with high ASR and high ACD, it would mean combinations of valid caller and called parties. The combinations can be used as whitelist for originator or terminator to filter traffic and pass to NCLI route. The softswitch is able to process both its own CDRs or imported CDRs from another sofswitch (see "Imported CDRs"). Data flow: CDR ---> group by something (A, B numbers) ---> get ASR and ACD for group ---> filter by ASR, ACD, Ncalls in group ---> for the groups build black lists/white lists/grey lists ---> filters for old SIMs, for new SIMs
Example: Find out numbers that called more than 10 times and connected 0, 1 or 2 times (ASR less than 20%) with low ACD (less than 30 seconds). B number in CDR contains tech. prefix, last 11 digits are used.

System status and maintenance

On this page you can check status of CPU, memory, monitor system overloads (large delays in threads), restart the softswitch, update software to latest version

Settings

The page contains general softswitch settings with descriptions; originators and suppliers self-care web portal settings; PayPal API settings for automated settings from VoIP retail users

Default call processing steps

Installation

Setting up the VoIP softswitch on your own Windows server

• Install Google Chrome browser if it is not already installed (it works better than Internet Explorer for the softswitch WebUI)
• Install StarTrinity Softswitch from this link
  • Download ZIP file to the server, open it, run web installer
  • Make sure that default SIP port (5060) and HTTP port (19019) are not busy by some other application on the server
  • This will install Windows Service "StarTrinity Softswitch" and open web interface (web UI) on default http port 19019
• Optionally you can use settings from another installation of the softswitch (if you set up a redundant server with same configuration):
  • Go to Control Panel - System and Security - Administrative tools - Services
  • Stop service "StarTrinity Softswitch"
    
  • Replace XML files ("StarTrinity.Softswitch.settings.xml" and "StarTrinity.Softswitch.settings_rates*.xml") with from another installation of the softswitch. Default folder with softswitch program files is C:\Program Files (x86)\StarTrinity\Softswitch
    
  • Start service "StarTrinity Softswitch"
  • Access web UI and check the restored configuration settings
• If you set up softswitch with clean new settings file, enter administrator's password in web UI. Please don't use simple passwords
• The softswitch is ready to for further configuration
• Optionally you can open TCP port 19019 in Windows Firewall to allow remote HTTP connections to the softswitch via internet. You can also set up HTTPS certificate for better security (default HTTPS port is 19020)
• If you have anti-virus software installed with its own embedded firewall, configure it to allow access for the softswitch

Setting up the VoIP softswitch for the first call - basic steps

• Setup termination (supplier) side
  • Terminators -> Add new terminator into previously created supplier (a way to connect to the supplier, SIP trunk). Enter SIP connection parameters.
  • Destination sets -> Add new destination set (tariff for supplier, list of available country codes with prices, routes)
    -> Add new route into the new destination set or import routes from supplier's pricelist file
  • Routing groups -> Add new routing group -> Add new item that links the routing group with previously created terminator and destination set
• Setup origination (customer) side
  • Tariffs -> Add new tariff (list of country codes and prices to charge clients) ->Add new rate into the new tariff
  • Service plans -> Add new service plan (service options like billing cycle), select previously created tariff
  • Originators -> Add new originator (company or user which sends VoIP traffic to you), select service plan and routing group. Enter SIP connection parameters
• Make the first test call from the originator
  • CDRs -> look at the call history, see if the new call is there. Check status field
  • If there is a problem, go to CDRs -> "..." (details) -> SIP trace. Also check system log

Web API methods

• GET /API/MainViewModel/Originators/CreateOriginator?originatorId=xxx& parentOriginatorId=xxx&maxChannelsCount=xxx&webUiUserId=xxx&webUiPassword=xxx&authUser=xxx&authPassword=xxx& balance=xxx&maxCredit=xxx&customCallerId=xxx&email=xxx&address=xxx&customerName=xxx - creates originator, is used for new user sign up at VoIP originator's website
• GET /API/MainViewModel/Originators/GetBalance?originatorId=xxx - returns originator's balance
• GET /API/MainViewModel/Originators/AddBalance?originatorId=xxx&value=10 - add some value to originator's balance
• GET /API/MainViewModel/CreateCall?url=http%3A%2F%2Fyour_server%2Fget_script.aspx&user=usr&password=pass - downloads CallXML script from your web server with specified auth. user and password, creates an outgoing call with the downloaded script
• POST /API/MainViewModel/CreateCall_Post - creates an outgoing call with the uploaded script. Returns Call-ID SIP header of the created SIP call in JSON format: {'status': 'OK', 'sipCallId': 'the_new_call_id'}. CURL example: curl http://X.X.X.X:19019/API/MainViewModel/CreateCall_Post -uadmin:admin --digest -X POST -d "@my_file.xml" -H "Content-Type:text/plain;charset=UTF-8"
• GET /API/MainViewModel/CreateSingleCallCommand - creates an outgoing call using currently pre-configured script
• GET /API/MainViewModel/CurrentCallExists?callerId=XXX&calledId=YYY - checks existence of current call, returns 'true' or 'false'
• GET /API/MainViewModel/DestroyCall?[sipCallId=XXX][&calledId=YYYY][&calledIdSuffix=ZZZZ] - destroys current SIP call(s) with specified parameters:
  • sipCallId - SIP Call-ID header of the destroyed call(s)
  • calledId - CLD (B number) of the destroyed call(s)
  • calledIdSuffix - CLD (B number) suffix of the destroyed call(s) - this parameter omits tech. prefix
• GET /API/MainViewModel/GetCurrentCallsCount?optionalCallXmlVariable1=XXX&optionalCallXmlVariable2=YYY - returns number of current calls, optionally filtered by CallXML variables
• GET /API/MainViewModel/GetVersion - returns software version information - compilation time (UTC)
• GET /API/MainViewModel/SetCallXmlVariables?callerId=XXX&calledId=YYY&direction=ZZZ&var1=value1&var2=value2 - sets CallXML variables for current SIP call, returns 'true' if call was found or 'false' if call was not found. The direction parameter of URL query is optional, it can be "in" or "out"
• POST /API/MainViewModel/UploadNumberList?id=xyz.txt - uploads new numbers into number list with ID = xyz.txt, one number per line in HTTP POST body. The uploaded numbers can be used with "readnumberlist" CallXML element. Uploaded numbers are stored in .TXT files in program folder
• POST /API/MainViewModel/CallcenterAgents/Upload - uploads call center agents data, one item per line. Line format: u=agentSipPhoneUserId;p=agentSipPhonePassword;g=group1 or n=agentPhoneNumber;g=group1. The uploaded numbers can be used with "getcallcenteragent" CallXML element, see CallXML documentation. Uploaded call center agents are stored in .TXT files in program folder
• GET /API/MainViewModel/CallcenterAgents/Download - downloads call center agents data, one item per line
• GET /API/MainViewModel/CallcenterAgents/IsAgentRegistered?u=userId - returns "true" if agent with "userId" is registered; if not registered - returns "false"

Video tutorials/presentations