VoIP call generator (dialer campaigns) in StarTrinity Softswitch - step by step tutorial
Key features
- Scripts (call flow scenarios)
- Simplest IVR - generate calls and play an audio file when the call is answered. Cancel call if it does not connect in specified time
- Looped generated traffic -
Generate call, end the call if it goes to another route (does not get back to this server). Link call to originator by B number (CLD) in real time.
If call does not go to back to this server in 7 seconds after '183 session progress', cancels the call.
If call gets connected by some other provider, ends call immediately.
This script must be combined with originator's script type = 'receive looped traffic'. B numbers must be unique for live (current) calls.
Plays one audio file or random file from folder. It could be configured to play random pair of A-side and B-side recordings from real calls, together with originator's script.
Number of concurrent calls is not limited by number of recordings - 10, 100 or 1000 in one folder
- Custom CallXML scripts
- Import of A and B number lists from TXT or CSV files
- Display of current calls, SIP trace
- CDR display
- CDR statistics
Pricing
Monthly fee, based on number of concurrent calls (CC) and number of calls per second (CPS). Note that actual number of CC and CPS depends on hardware capabilities.
- 1 server, our hosting, 100 CC, 10 CPS: 100USD/month
- 1 server, our hosting, 300 CC, 30 CPS: 150USD/month
- 1 server, your hosting, 600 CC, 100 CPS: 200USD/month
- 1 server, your hosting, unlimited CC, unlimited CPS: 300USD/month
- Download user agreement
- Regular technical support: 300USD/month
- High-priority technical support: 100USD/hour
Sign up to use the VoIP dialer (call generator) on our servers
Demo with live calls
You can take a look at live demo set up here:
netherlands softswitch server. Use
user=dc_demo, password=123456 for login.
Step by step tutorial
- Install softswitch on your Windows server using this tutorial, or register on our server here. Free trial is available.
- Open softswitch web UI in browser, log in
- Go to menu - Configuration - Terminators - add terminator (VoIP provider). Enter unique ID for the new terminator, number of channels allowed by the destination server (capacity)
, enter IP address or host name of destination server, where you need to send VoIP calls.
- Configure A and B numbers for generated VoIP traffic
- If you have lists of numbers: go to menu - Configuration - Number lists - add number lists with A (CLI) and B (CLD) numbers. Upload CSV or TXT files with numbers, 1 number per line.
After uploading click "details..." to check the uploaded numbers
- If you want to generate numbers randomly: enter CLI and CLD pattern further, when configuring dialer campaign
- Go to menu - Configuration - Dialer campaigns - add dialer campaign. Select type of script, terminator, enter max channels count and max calls per second for the new dialer campaign.
Generated traffic is always limited either by "max channels count" limit or by "max calls per second" limit.
Click "Add", system redirects you to more detailed configuration screen. Here you configure source of A and B numbers,
max. channels deviation (for randomly fluctuating number of channels).
- For "Simplest IVR" script: enter max. connected time (billed duration), max ringing time, audio file for playback
- For "Looped generated traffic" script: configure audio file:
Configure receiving generated traffic back to this server:
- Go to menu - Configuration - Originators - add originator. Enter unique ID, max channels allowed to receive from originator (source server).
- Configure authentication: enter source IP address, set script type = "looped traffic receiver", set ASR, configure audio file playback:
- Optionally make a single test call (menu - configuration - dialer campaigns - dialer campaign details - state - make 1 call)
- Start dialer campaign (menu - configuration - dialer campaigns - dialer campaign details - state - start)
- Monitor current calls:
- menu - Status - Status:
- menu - Status - Current Calls:
- Check call history (CDR), see details of connected and/or failed calls
- If calls failed: find call in CDR, click on "status code", see details in SIP trace
- Check CDR statistics: total connected time for a period of time and specific dialer campaign
- In order to make a pause in generated VoIP traffic, go to menu - configuration - dialer campaigns - dialer campaign details - state - click "pause"
