I create a custom ivr in freeswitch. caller calls 11111 and routs to custom ivr. How can i test this ivr using sipp tester? What does do for config freeswitch if requied and what is the sipp command line and scenario xml file for testing this scenario?
I think you can get that info in this article.
https://mojolingo.com/blog/2013/load-testing-voice-applications-with-sipp/
Related
I'm trying to make load testing of my Telegram bot. So I tried to use HTTP(s) Test Script Recorder for that in JMeter and ran Proxy in Firefox for that, while did manipulations in my Telegram bot, but all that I had was just some random *.png files recorded in JMeter, which say me nothing and moreover, they came just ones and after records nothing, when I continue to work in the bot. When I record another sites request, everything is fine, but not Telegram. Read the following question and Rubycon says that there is only single Http Api request, but don't know which one and how to find that.
Telegram Bot load testig
Can somebody help with that?
If you want to use JMeter's HTTP(S) Test Script Recorder for recording Telegram network traffic you need to set Telegram to use JMeter as the proxy:
additionally you may need to install a special loopback adapter so Telegram wouldn't ignore local proxies and would be able to route traffic through them.
Also you will need to add JMeter's MITM certificate to Trusted Root Certification Authorities
More information: How to Run Performance Tests of Desktop Applications Using JMeter
My scenario is given as below :
I login into application and upload a doc file then request send to server where thrid party tool (doc to pdf)is running and convert doc file to pdf and send file conversion status pass\fail on application. Can i Record scenarion in JMETER that access microsoft remote desktop protocol and check third party tool activity and come back to application?
Thanks,
Raman
If you want to "record" the RDP protocol the answer is "no", JMeter's HTTP(S) Test Script Recorder can only record HTTP or HTTPS protocols, other traffic will not be recognized.
If you want to take some screenshots of the remote Windows machine you can use i.e. JavaRDP library from JSR223 Test Elements
If you need to read the conversion log and add it to JMeter test results it might make more sense to consider connecting to the machine using PowerShell Remoting from the OS Process Sampler.
Check out How to Run External Commands and Programs Locally and Remotely from JMeter article for more information on the concept and example configurations.
I am setting up a performance testing using Jmeter for Chat bot application, How to start with it?
I have tried with JMeter WebSocket Samplers by Peter Doornbosch but not succeeded
To start, you need to find out how the chat client communicates with the server, i.e. what websocket messages it is sending to the server. To find out, ask the developers of the chat bot, or use a tool like WireShark to capture the communication and analyse it. Once you know which messages are exchanged, you can start setting up a test plan that simulates the client.
For doing perf testing with jmeter I am using this plugin :-
https://github.com/damadei/BotServiceStressToolkit/blob/master/docs/creating.md
This will provide you with the correct options to perform chatbot testing .
I developed a IVR and sms system using twilio.com. I have used it's api to develop the system. But now I want to have my own setup which does the same features. I did some home work and read things about VOIP and SIP. All those talks about internet calls. But this is something that I need to call a cell phone or land line and expect calls from those phones. Is this possible without using any hardware and also not relying on services like Twilio.
You can setup Asterisk or Freeswitch project in AWS or Digital Ocean. Asterisk/Freeswitch offer IVR services, take a look at this guide and setup via Twilio documentation:
https://www.twilio.com/docs/api/sip-trunking/sample-configuration#freeswitch
For PSTN integration, you can use Twilio or any other SIP trunk provider.
This will be the flow:
Cellphone > PSTN > Twilio/SIP Trunk > PBX -> IVR
You can have your own setup by having FreeSWITCH or Asterisk. these are the soft switch. you can generate your own logic for ivr. You can control the switch as per your guidance.
I suggest you to go with FreeSWITCH.
I wonder if there is a way to send sms messages out from freeswitch box via a gateway just like bridge api in mod sofia for calls.
I can originate and receive calls from a sip provider via an external gateway on freeswitch. Now i need to do the same philosophy for chat using the same gateway.
If not, I would be greateful if you recommend an open source http to sip gateway as an alternative.
Thank you for any help.
You need to check what messaging protocol (e.g. SIP SIMPLE) is accepted by your remote sms gateway. SIP SIMPLE is supported by Freeswitch out-of-the-box. Then, you can either execute "chat" API command from your dialplan:
<action application="set" data="api_result=${chat(sip|from#sender|to#receiver|${your_text_msg})}"/>
or create a chatplan (see mod_sms for latter) if you require more sophisticated message routing.
The answer is indeed using chat application but a follows:
Assuming that I have an iptel gateway registered on my FreeSWITCH server "8123456789#iptel.org" and that I have the following user registered on ipetl "sip:811111111#iptel.org"
To send an SMS to this user out from FreeSWITCH via the iptel gateway, I can do it from FreeSWITCH console as follows:
chat sip|noreply#mydomain|external/sip:811111111#iptel.org|Hello It's working !
Or the api (with ESL for e.g.):
api chat sip|noreply#mydomain|external/sip:811111111#iptel.org|Hello It's working !
Of course you can replace the iptel gateway in this example with the gateway that your SIP provider gave you. Make sure that the Gateway supports chat using the SIP SIMPLE protocol.