Noyce! One of the rare times that I follow a tutorial that is kinda long (but extremely well made) and that I don't run into a single problem. Thanks a lot!
You're welcome, man. Looking forward to seeing some of your videos where you use Asterisk. Also, a lot of those numbers you've tried in the past where you get "this number is currently out of service" are because the scammer call center has configured *their* Asterisk to play that "out of service" message when calling from a Skype caller ID. I've found that if I get that message using Skype, calling from Asterisk gets me through. ;)
look, idk if itd be 100% legal, but could either of you two tell me a way to track an ip address off of instagram? I mean no harm and its a life and death situation. My friend may be dead but i dont know, i just want a way to find out about her. Hannah needs help, and in case shes alive i want to be able to send help next time she does something like this.
+Dante Rosales Well, that might not be possible as Instagram does not have any point to point technology implemented like Skype, which by a little work, you can find out someone's IP by their Skype ID. Instagram is, AFAIK, only server-based and getting an IP from an Instagram account probably means accessing Instagram's servers. As I said, Skype is your best bet if you want to find out someone's IP address.
So, GoTrunk went the lame route and configured their servers to completely ignore the "callerid=" command from Asterisk. This was most likely a direct result of so many people flooding their service, or because the scamming madarchods saw this tutorial and complained to GoTrunk. I did find another SIP trunk that has a free trial, and does not block the caller ID command from Asterisk. However, I'm not too sure if I should be posting it publicly now that they might do exactly what GoTrunk did. If I updated the tutorial to explain how to do it with another company, then for all I know the scammers would see it and complain to the new company. Then I'd have to waste another whole half a day searching for another replacement. My only suggestion for now is to get this tutorial working, and then search popular SIP trunks that provide free trials. Paste their Asterisk config examples into sip.conf, edit ext-template.conf from this tutorial to say "@[name of new peer from sip.conf]" instead of "@trunk", and try it out. If the caller ID that you put in comes up, then you've found one. If the caller ID comes up, but then when you call a scammer the call doesn't go through, then that particular SIP trunk is reporting to the scammer's SIP trunk that your DID is unverified. You just move on to the next SIP trunk trial until you find one that lets you through to the scammer's phone number with a spoofed number. So, basically I've found one that does not report DID verified status, so the scammer's SIP trunk just has no clue it's a fake number and accepts the call. Some other trunks I've tried did spoof the number when calling my cell phone, but the scammer's server denied the call because the SIP trunk sent out that it was an unverified DID. So, trust me, there are SIP trunks out there that work like GoTrunk used to, but the one I'm using now is just buried in tons of BS ad websites and shitty SIP trunk services. Now to make "trolling scammers with caller ID spoofing - part 2"... ;)
Hello dear one, hope you're fine. 7 years is a lot of time. I tried to check the Internet for a connection with more and updated tutorials made by you. No luck. If someone has news about your new projects or if there's a blog with your contributions, I would be happy to know. Thank you
Thanks for the inspiration. Didn't get this method to work because of the whole caller ID issue. But I did come up with a different method that is able to call from a random number X amount of times.
guys does it work even now.. i spent more than 9 hours to set this up bcz all the given links are changed. i successfully set the asterisk server but nothing works after that🙁
When you test your set up with test number 12120001234 do NOT use the "+" prefix! Ran in the same issue and took me around half an hour to figure this out. Hope this helps someone.
Awesome tutorial. I don't know how you stay so serious while talking to them. And your Hindi... don't know how/where you learned it, but you really get to them when you use it. Brilliant! What do you do when the "free trial" is over on GoTrunk? Just set up another with a fake email?
Thanks, man. But yeah, if you don't have an inbound number, then you don't have to pay anything except to top off your call credit. And toll-free numbers don't use any call credit. ;)
@@Asdfghjklmnbvcxer No its not, the page he shows in the tutorial its shutted down I think. You can still use Asterisk but you need to find another Sip tho
when i run asterisk -rx "core restart now" unable to open master control file '/etc/asterisk/..asterisk.conf but there is no such file in /etc/asterisk help please
Around 3 years later: Does this still work? Lets figure it out! But first things first: VERY GREAT TUTORIAL! Thank you very much! As expected some parts changed in the meanwhile. Like "OpenSSH server" installation, "Bria solo" instead of "X-Lite", etc. Finally I was able to change the outgoing caller ID with the "./new_number" script, saw the change in "/etc/asterisk/extensions.conf" and was able to make outbound calls. But "Anonymous" was shown in the phone I called. A phone number (and not "Anonymous") was only shown on the called phone in case I uses the number that was assigned to the GoTrunk account (and which is displayed in the GoTrunk wizard step #6, region "Test Inbound Call"). So finally: It does not work any longer! Any ideas?
lol ... RTF comments! "FakeTechScammers" answered this already 3 years ago here in the comments. ... lol ... GoTrunk ignores the asterisk callerID entry ... Search another provider like "GoTrunk" that works like GoTrunk does ... But do NOT publish the name, etc. of this "provider" here. Because otherwise would the same again happen: The "new" provider would change their API ... But search first here for "FakeTechScammers" comment. He explained it well in HIS comment.
@@v.gedace1519 I've setup everything, and it's working now, but I'm not able to guess the name of the new sip trunk provider which still works with callerid option of asterisk, Can you give me a hint or any clue or something? I can give you my email id or any other temp contact information where you can give me an idea, I really wanted to piss one scammer, now this scammers also started to rob money in India by fooling old peoples.
@@v.gedace1519 I've setup everything, and it's working now, but I'm not able to guess the name of the new sip trunk provider which still works with callerid option of asterisk, Can you give me a hint or any clue or something? I can give you my email id or any other temp contact information where you can give me an idea, I really wanted to piss one scammer, now this scammers also started to rob money in India by fooling old peoples.
@@kalindpatel8787 lol ... What didn't you understand? I said "No, sorry". I also don't know. No. I found NO replacement for GoTrunk. Not working. No email. No info. No whatever. Sorry.
If you're on windows change the network connection to a bridged connection then reboot the vm. After you do that run ifconfig again and use the address you get there. This is for PuTTY.
i have a problem when i ave it on bridged addapters i get a 192.168.1.119 ip and can connet but i can not do apt-get updater or any thing it won't connect to the sites how can i fix this?
I can follow this step by step and it works flawlessly up to the point where X-Lite is supposed to become available, but I get a SIP 408 or 503 error. The only difference is I am running Windows as my host OS w/Putty and installed PJSIP(not native Asterisk) library. I know the errors are either the username/PW don't work or can't reach the Asterisk server. I have verified that the UN/PW are correct and the domain(IP) is correct through Putty. Any ideas?
GoTrunk has now changed their server to not obey the "CALLERID=" command from Asterisk, effectively disabling caller ID spoofing. You now need to find a different SIP trunk service to spoof the caller ID. And so the search begins... those damn madarchods just had to complain to GoTrunk.
***** Do you have a twitter or something for that ? The MP system is juste a lot more better than on TH-cam, and links are automatically detected as spamms :/
can someone help please. on xlite after all the configuration, when I try to make a call to any number I get "failed to establish call. forbidden number"
im getting the same issue, i think it has something to do with internet provider not allowing call spoofing but im not sure, im still stuck on that step figuring it out.
Hello, i have a problem, I have made same as you in the video, but when I test the call with xlite, it does not work (Call failed to connect) :( I think it's because I chose dynamic IP address , but I have a fixed IP ... Can you Help me? Thank's
Hi! Is it possible to set it up inside a docker container? And can I use x-lite on linux (if not what is a good alternative) Thanks for the amazing tutorial!
When i tried to make an inbound call. It gave me the busy tone and when I click on the verify inbound call it says "No Inbound calls found to your SIP Endpoint."
For some reason I can call local numbers (haven't tried out of country numbers yet) but when I go to call Samsung support for example, it gives me a 'connection could not be established'...
i follow the tutorial and everything seems to be going fine right up until i try to make a test call and it cant dial out, i cant figure out what the hell im doing wrong
I'm actually trying to figure out HOW call spoofing works. Is it injecting encoded FSK caller ID audio in to the phone line as the call is being placed? I haven't been able to find out if this is it actually works. It seems strange to me the telephone company would allow audio to go out to the receiving caller from the source, before the call is answered. I also want to mention this method won't work on any SIP trunk providers that have CNI enabled.
I do see what you mean. I did all of the things that he did and everything works (be doing all the updated stuff) however i cannot spoof it anymore, as his notation says. Which is sad, cannot find any sip providers that actually has CNi off nor anyone that doesnt take painment instantly.
If the test call works, then everything is set up correctly. That means the number you entered isn't valid. If it's a US number it has to be 1(area code)(number). No spaces, no parenthesis, and you have to include the "1".
bro..I m facing some problems when I m trying to Installing dependencies(apt-get install build-essential).Its saying I have to get root permission and cant access to the locked directory.What can I do now?
@FakeTechSchammers i cant use x Lite in linux and i installed the asterisk in my kali mechine but i cant find any alternative of xlite can you please help brother?
at min 7:40 i fail when i hit enter it says(ssh is not recognized as an internal or external comand,operable program or batch file) ihave windows 10 ,command prompt.
Hopefully someone can help me here. I set up everything correctly and it's working fine but the audio seems very low when everything is maxed out, any ideas what I could do, I've almost tried everything I can..
I re-watched the video this morning and have a couple of questions. 1) Isn't providing false details to Go-trunk breaching some TOS or could it be classed as fraudulent 2) Does this setup mean you need to have two virtual machines running to perform this?
Yeah, you probably shouldn't do that. I deleted my account right after filming the tutorial. And yes, you'd have to have Ubuntu Server running while you are trolling the scammer with a VM, or else X-Lite won't work.
Yeah I tried it on a VM first then installed it on my actual Ubuntu server, it runs with no issue, plus I installed csipsimple on my android phone which can connect to it from anywhere :-)
@@weirdscix bro .. I have a issue. Server runs perfect in VM ubuntu server. But on a actual cloud server from amazon aws .. it calls but no voice, then hangs up by it self.
noticed here the SIP trunk no longer allows for caller ID spoofing. this is absolute bullcrap. How am i supposed to know how to set it up with other services or which service offers what we need? 3/4 into the video and we cant even use gotrunk. really sucks. can I please get something off you that works?
Probably people freak out when they see the list of the files loading. Also I suppose he set things up so every time the computer opens, it opens in safe mode, so if you don't know how to switch it back to normal mode then you will think something is wrong.
dude everything is set up but the problem is i am unable to spoof the number. even after removing the variable and assigning the mobile number manually it isn't working. please help me dude ...
hELP! I get this error after starting my asterisk server : Peer 'trunk' is now UNREACHABLE! And then I get call to '12120001234' rejected because extension not found in context 'from-internal' and DNS error for registration to 27900@trunk
I have some problems I am setting up asterisk server on my Amazon vps and also on Microsoft azure vps but the problem is I'm getting only one way Audio I thought maybe you could help me please
someone please help i am on windows and i get this error Account failed to enable, Acout:Test could not be enabled Problem at server(SIP error 408). Try agian Later
Had it working, but i made a new go trunk account and all outbounds calls from the system now come up anonymous/unknown, as if caller id was blocked. Checked cdr in go trunk, and even there it comes up blocked
Please tell me how can I put my own VoIP service that I already bought on the sip.conf file ,I spent hours following your tutorial so please don't leave me hanging
@@faketechscammers9454 could you also tell me a good free alternative from gotrunk that allows change the idcaller? [my email is necropower123@outlook.com]
raspberry pi* yes you can do everything on the raspberry pi, silent p in raspberri and its pi not pie reason = pi mean an infinitaly large number short version being 3.141
I couldn't build the source on my VPS (kept getting terminal size error, because I was SSHed in I guess?) but you CAN do this from the release (apt-get install asterisk) and the setup is exactly the same.
Oh wow, you must not have a 64bit CPU or operating system. Try downloading the Ubuntu Server 32bit version... not the 64bit. Also, in my steps when creating the virtual machine, you need to select "Linux 32bit" instead. Here's a link to a Ubuntu Server version that might work for you: releases.ubuntu.com/trusty/ubuntu-14.04.5-server-i386.iso
Im having an issue where the calls just say "Failed to establish call". You got any ideas, ive followed the tutorial to the letter and its not working :(
I actually never did technically, the test call number was ringing but it never picked up to the automated system. But i still cant call anybody. The test number only seems to ring and not give the error if I put a 9 infront of the number which is used in businesses if they want to call an outside number rather than another internal extension.
I seem to be getting various errors if i trace the connection on the asterisk cli. Things like "SIP/2.0 403 Forbidden" and "SIP/2.0 100 Trying" & "SIP/2.0 404 Not Found". Any help would be appreciated. I get the same as everybody else i think where it just says call failed to establish
I got this error when i did the "make" command. Please help! collect2: error: ld returned 1 exit status /usr/src/asterisk-certified-13.1-cert8/Makefile.rules:138: recipe for target 'chan_iax2.so' failed make[1]: *** [chan_iax2.so] Error 1 Makefile:386: recipe for target 'channels' failed make: *** [channels] Error 2
also, I know the method no longer works, but for those who are willing to purchase a trunking sip, and wish to do so over firertc (such as myself), could you make a tutorial on doing this with Elastix?
not for free. You see, you can still call, but you'll come up as unlisted, and honestly, most scammers avoid unlisted calls anyway. This is because Gotrunk no longer accepts the "set callerid" command shown in the video. You'd have to find a service that does allow it, but I have yet to find one that also has a free trial. However, if you are willing to pay for an SIP service, the setup on your end is basically the same, just with a different upstream trunk provider. If I got anything wrong, please correct me
You know, I'm not really sure. I think outbound calls use "credits," which means as long as you're calling toll free numbers it's unlimited (and you can top off your credits with payments). The trial and monthly payment would only be for incoming calls. Don't quote me on this, but it seems like that might be it.
![กี่หมื่นฟ้า | Your Sky Series EP.1 [1/4]](http://i.ytimg.com/vi/vLGjxFL6U_I/mqdefault.jpg)
Noyce! One of the rare times that I follow a tutorial that is kinda long (but extremely well made) and that I don't run into a single problem. Thanks a lot!
Quality tutorial, works great! thanks :)
You're welcome, man. Looking forward to seeing some of your videos where you use Asterisk.
Also, a lot of those numbers you've tried in the past where you get "this number is currently out of service" are because the scammer call center has configured *their* Asterisk to play that "out of service" message when calling from a Skype caller ID. I've found that if I get that message using Skype, calling from Asterisk gets me through. ;)
+FakeTechScammers Interesting to know that. I'll keep that in mind next time I do some scammer calling
look, idk if itd be 100% legal, but could either of you two tell me a way to track an ip address off of instagram? I mean no harm and its a life and death situation. My friend may be dead but i dont know, i just want a way to find out about her. Hannah needs help, and in case shes alive i want to be able to send help next time she does something like this.
+Dante Rosales Well, that might not be possible as Instagram does not have any point to point technology implemented like Skype, which by a little work, you can find out someone's IP by their Skype ID. Instagram is, AFAIK, only server-based and getting an IP from an Instagram account probably means accessing Instagram's servers.
As I said, Skype is your best bet if you want to find out someone's IP address.
+Ciprian Ionescu and how would i go about it the skype way?
Hello from 2024.
Hope you guys are doing well.
Is this thing still working?
So, GoTrunk went the lame route and configured their servers to completely ignore the "callerid=" command from Asterisk. This was most likely a direct result of so many people flooding their service, or because the scamming madarchods saw this tutorial and complained to GoTrunk.
I did find another SIP trunk that has a free trial, and does not block the caller ID command from Asterisk. However, I'm not too sure if I should be posting it publicly now that they might do exactly what GoTrunk did. If I updated the tutorial to explain how to do it with another company, then for all I know the scammers would see it and complain to the new company. Then I'd have to waste another whole half a day searching for another replacement.
My only suggestion for now is to get this tutorial working, and then search popular SIP trunks that provide free trials. Paste their Asterisk config examples into sip.conf, edit ext-template.conf from this tutorial to say "@[name of new peer from sip.conf]" instead of "@trunk", and try it out.
If the caller ID that you put in comes up, then you've found one. If the caller ID comes up, but then when you call a scammer the call doesn't go through, then that particular SIP trunk is reporting to the scammer's SIP trunk that your DID is unverified. You just move on to the next SIP trunk trial until you find one that lets you through to the scammer's phone number with a spoofed number.
So, basically I've found one that does not report DID verified status, so the scammer's SIP trunk just has no clue it's a fake number and accepts the call. Some other trunks I've tried did spoof the number when calling my cell phone, but the scammer's server denied the call because the SIP trunk sent out that it was an unverified DID. So, trust me, there are SIP trunks out there that work like GoTrunk used to, but the one I'm using now is just buried in tons of BS ad websites and shitty SIP trunk services.
Now to make "trolling scammers with caller ID spoofing - part 2"... ;)
Can i get a PM?
Please PM if you can:)
I wait for another SIP trunk because I did not find one like gotrunk. If you can give one I will be happy. ..
You should share the SIP trunk only to the persons who ask it :)
Hey could you PM me the new SIP Trunk? Thanks.
Hello dear one, hope you're fine. 7 years is a lot of time. I tried to check the Internet for a connection with more and updated tutorials made by you.
No luck.
If someone has news about your new projects or if there's a blog with your contributions, I would be happy to know.
Thank you
literally took me 9 times to input that first one with my windows computer. I can tell this will be very fun.
I'm a little upset that I spent 8 hours doing this only to find out it doesn't work
For Linux you need more knowledge. Like programming a bit hacking. With no dumb ass knowledge you going no where
I'm unironically impressed you spend 8 hours on it. That's dedication.
Me too spent a lot time ...
Got to know many things..
Same here xd
i honestly barely remember spending 8 hours on something like this but i wouldn't doubt that i did.
Thanks for the inspiration. Didn't get this method to work because of the whole caller ID issue. But I did come up with a different method that is able to call from a random number X amount of times.
please share....
share pls
asterisk tutorial, start off with half hour setup of virtualbox!
does it still work?
Did try this yet but as an Indian I really enjoyed the end encounter. Omg it was hilarious!! Good tutorial though dude
guys does it work even now.. i spent more than 9 hours to set this up bcz all the given links are changed. i successfully set the asterisk server but nothing works after that🙁
i had issues configuring ubuntu with asterisk
btw, you can create a snapshot without shutting the machine down.
Awsome video I'm studying this at a university. May get this set up in advanced...
"Failed to establish call"
But I can connect to the account with X-Lite!
And I entered the SIP information into the sip.conf file, please HELP!
and I can't reach the domain eu.st.ssl7.net & 109.233.115.107!
use vpn to connect to canada or europe
When you test your set up with test number 12120001234 do NOT use the "+" prefix! Ran in the same issue and took me around half an hour to figure this out. Hope this helps someone.
Awesome tutorial. I don't know how you stay so serious while talking to them. And your Hindi... don't know how/where you learned it, but you really get to them when you use it. Brilliant! What do you do when the "free trial" is over on GoTrunk? Just set up another with a fake email?
Thanks, man. But yeah, if you don't have an inbound number, then you don't have to pay anything except to top off your call credit. And toll-free numbers don't use any call credit. ;)
@@faketechscammers9454 Hlo Bro This method is still working or Not ?
@@Asdfghjklmnbvcxer No its not, the page he shows in the tutorial its shutted down I think. You can still use Asterisk but you need to find another Sip tho
Thanks for the information . I'm working on this right now. Wish me luck
good luck
Tell me if this worked
@@mirsaiyed7854 I gave up as I exams
when i run
asterisk -rx "core restart now"
unable to open master control file '/etc/asterisk/..asterisk.conf
but there is no such file in /etc/asterisk
help please
Find the file online and create it yourself or download it. Pretty simple
Around 3 years later: Does this still work? Lets figure it out! But first things first: VERY GREAT TUTORIAL! Thank you very much!
As expected some parts changed in the meanwhile. Like "OpenSSH server" installation, "Bria solo" instead of "X-Lite", etc.
Finally I was able to change the outgoing caller ID with the "./new_number" script, saw the change in "/etc/asterisk/extensions.conf" and was able to make outbound calls.
But "Anonymous" was shown in the phone I called. A phone number (and not "Anonymous") was only shown on the called phone in case I uses the number that was assigned to the GoTrunk account (and which is displayed in the GoTrunk wizard step #6, region "Test Inbound Call").
So finally: It does not work any longer! Any ideas?
lol ... RTF comments!
"FakeTechScammers" answered this already 3 years ago here in the comments.
... lol ...
GoTrunk ignores the asterisk callerID entry ...
Search another provider like "GoTrunk" that works like GoTrunk does ...
But do NOT publish the name, etc. of this "provider" here.
Because otherwise would the same again happen: The "new" provider would change their API ...
But search first here for "FakeTechScammers" comment. He explained it well in HIS comment.
@@v.gedace1519 I've setup everything, and it's working now,
but I'm not able to guess the name of the new sip trunk provider which still works with callerid option of asterisk,
Can you give me a hint or any clue or something?
I can give you my email id or any other temp contact information where you can give me an idea,
I really wanted to piss one scammer, now this scammers also started to rob money in India by fooling old peoples.
@@v.gedace1519 I've setup everything, and it's working now,
but I'm not able to guess the name of the new sip trunk provider which still works with callerid option of asterisk,
Can you give me a hint or any clue or something?
I can give you my email id or any other temp contact information where you can give me an idea,
I really wanted to piss one scammer, now this scammers also started to rob money in India by fooling old peoples.
@@kalindpatel8787 No sorry, figured it also not out.
@@kalindpatel8787 lol ... What didn't you understand?
I said "No, sorry".
I also don't know.
No.
I found NO replacement for GoTrunk.
Not working.
No email.
No info.
No whatever.
Sorry.
At first I thought he said his name was Uncle Gelati XD
If you're on windows change the network connection to a bridged connection then reboot the vm. After you do that run ifconfig again and use the address you get there. This is for PuTTY.
Awesome! Great tutorial, would have been a little more awesome if the video was based on windows 10 j/p.
i have a problem when i ave it on bridged addapters i get a 192.168.1.119 ip and can connet but i can not do apt-get updater or any thing it won't connect to the sites how can i fix this?
Same
Hello it deosnt work anymore its to old
I can follow this step by step and it works flawlessly up to the point where X-Lite is supposed to become available, but I get a SIP 408 or 503 error. The only difference is I am running Windows as my host OS w/Putty and installed PJSIP(not native Asterisk) library. I know the errors are either the username/PW don't work or can't reach the Asterisk server. I have verified that the UN/PW are correct and the domain(IP) is correct through Putty. Any ideas?
use you have to run native astriks
I’m having the same issue
you're the greatest. None of this shit ever works for me, but it is now! Thankyou sir.
Have you still got it working?
can you get in contact with me personally mine doesnt work x-lite connects to asterisk fine but it fails to establish call
actually doesnt even connect with sip
+chrisheeley You followed all steps to edit your sip.conf file and replace with the username and password from GoTrunk?
yep
+chrisheeley and you ran the "core restart now" line?
+FakeTechScammers yep did everything on the vid
Can you make a updated version
Nice tutorial. Wonder if you are able to use ENV var in the config file, that way you dont need to rewrite the config just the env var.
So there is no alternative or solution to this? I'm looking up how to make my own SIP trunk but it doesn't seem like it would be able to call phones.
is this working on different countries from europe or only usa?
Hey, one question: All works, but if I call someone, there is no number. There is anonymous
when i try to connect with x lite on 10.0.2.15 i get that is was not suceded and i have configured everything right as in the video :(
GoTrunk has now changed their server to not obey the "CALLERID=" command from Asterisk, effectively disabling caller ID spoofing. You now need to find a different SIP trunk service to spoof the caller ID. And so the search begins... those damn madarchods just had to complain to GoTrunk.
Have you found a new sip trunk service yet?
Fuck! No wonder it wasn't working for me.
Is there any other SIP trunk service?
fck it is legal ... i do not understand why they bloked it...
ow. Nice. Can you give me the name of the SIP Trunk ? :)
***** Do you have a twitter or something for that ? The MP system is juste a lot more better than on TH-cam, and links are automatically detected as spamms :/
thanks man!, ill give this a try soon and let you know how it went
can someone help please. on xlite after all the configuration, when I try to make a call to any number I get "failed to establish call. forbidden number"
im getting the same issue, i think it has something to do with internet provider not allowing call spoofing but im not sure, im still stuck on that step figuring it out.
@@megazuccc same here! anyone help?
It’s go trunk find another trunk check for creator comment
I need this configuration. How can I contact you, I will pay you
Hello, i have a problem, I have made same as you in the video, but when I test the call with xlite, it does not work (Call failed to connect) :(
I think it's because I chose dynamic IP address , but I have a fixed IP ...
Can you Help me?
Thank's
If calling 12120001234 doesn't work and you've followed all these steps exactly, maybe you forgot to run asterisk -rx "core restart now"
no I have not forgotten, then I think it is on the site gotrunk I've done something wrong :/
I will be very thankful for lifetime man this has to do with my family bro
I thank you very much in advance
+FakeTechScammers I was wondering if you thought about running STIG on your server? or S-CAP?
Same problem
Hi! Is it possible to set it up inside a docker container? And can I use x-lite on linux (if not what is a good alternative) Thanks for the amazing tutorial!
and also help me about the sip trunk service as you told that go trunk is not working, so suggest something like that..
Hey... can I email you about something? Or how can I get in contact with you?
Buddy pleas give a alternative for x lite and that go trunk is not letting register on us ip or singapore neither indian
When i tried to make an inbound call. It gave me the busy tone and when I click on the verify inbound call it says "No Inbound calls found to your SIP Endpoint."
th-cam.com/video/pAOgJX5hsqc/w-d-xo.html
Does someone know why when I call the other person, they receives the call as "unknown"? How can this be fixed. Thx
For some reason I can call local numbers (haven't tried out of country numbers yet) but when I go to call Samsung support for example, it gives me a 'connection could not be established'...
th-cam.com/video/pAOgJX5hsqc/w-d-xo.html
is there anyway to generate random number using RND() func call in the script and put it in the loop? with a command line dialer ?
Hello i am having trouble working with the sip.conf it does now look the same the one in my machine. anyone know why??
It work to europe too ? because there is alot of europian scammers also it work on cell phones ? or just toll free numbers ?
Can we use for example "Mother" in caller id line, or there must be numbers?
I'm getting this error, "No RTP engine was found. Do you have one loaded?", I'm using ubuntu 20.04 and asterisk 18
I'm getting the same error aswell. I read this forum it helped me a lot community.asterisk.org/t/no-rtp-engine-was-found-do-you-have-one-loaded/42610
How I cat set up in asterisk that he should show my caller ID?
i follow the tutorial and everything seems to be going fine right up until i try to make a test call and it cant dial out, i cant figure out what the hell im doing wrong
same
i tried but cudent get it working cos of all that sip.conf stuff as the website he used dosent do it anymore
I'm actually trying to figure out HOW call spoofing works. Is it injecting encoded FSK caller ID audio in to the phone line as the call is being placed? I haven't been able to find out if this is it actually works. It seems strange to me the telephone company would allow audio to go out to the receiving caller from the source, before the call is answered. I also want to mention this method won't work on any SIP trunk providers that have CNI enabled.
I do see what you mean. I did all of the things that he did and everything works (be doing all the updated stuff) however i cannot spoof it anymore, as his notation says. Which is sad, cannot find any sip providers that actually has CNi off nor anyone that doesnt take painment instantly.
Please help mine is saying failed to establish call
When I call the test number it works, but doesn't work when I call another number
If the test call works, then everything is set up correctly. That means the number you entered isn't valid. If it's a US number it has to be 1(area code)(number). No spaces, no parenthesis, and you have to include the "1".
19 scammers didn't like the video..
Can i use this for Mobile Phones ?
my inbound calls are not coming in or is it because i called via skype?
Awesome tutorial and explanation is so deep and detailed
It works for you? I tried 3cx to host it, but I can't get it working
scammer detected
bro..I m facing some problems when I m trying to Installing dependencies(apt-get install build-essential).Its saying I have to get root permission and cant access to the locked directory.What can I do now?
th-cam.com/video/pAOgJX5hsqc/w-d-xo.html
Bro i need your help to setup i am facing a problem
@FakeTechSchammers i cant use x Lite in linux and i installed the asterisk in my kali mechine but i cant find any alternative of xlite can you please help brother?
at min 7:40 i fail when i hit enter it says(ssh is not recognized as an internal or external comand,operable program or batch file) ihave windows 10 ,command prompt.
or go into to windows developer mode and just install the Linux bash , through the back door lol , worked for me ;)
can i use as private call in the x lite
lol the title of the video specifically tell what i want to do XD
Have you got it working?
do you have another sip that's work today?
Hopefully someone can help me here.
I set up everything correctly and it's working fine but the audio seems very low when everything is maxed out, any ideas what I could do, I've almost tried everything I can..
th-cam.com/video/pAOgJX5hsqc/w-d-xo.html
I re-watched the video this morning and have a couple of questions.
1) Isn't providing false details to Go-trunk breaching some TOS or could it be classed as fraudulent
2) Does this setup mean you need to have two virtual machines running to perform this?
Yeah, you probably shouldn't do that. I deleted my account right after filming the tutorial.
And yes, you'd have to have Ubuntu Server running while you are trolling the scammer with a VM, or else X-Lite won't work.
+FakeTechScammers ah OK thanks, I'm going to give this a try.
Yeah I tried it on a VM first then installed it on my actual Ubuntu server, it runs with no issue, plus I installed csipsimple on my android phone which can connect to it from anywhere :-)
@@weirdscix bro .. I have a issue. Server runs perfect in VM ubuntu server. But on a actual cloud server from amazon aws .. it calls but no voice, then hangs up by it self.
why do you make two virtual machines why not just use the ubuntu one as your windows
Can you make updated video?
I need a little help getting the VM connected to my pc, its not allowing me to SSH nor connect to X-Lite
You need to set the network setting to bridged, and install openssh-server.
Thanks man!
Seems like a lot of work for a small payout. Any simpler way to do this that is just not as good?
is this still working in 2021?
Unable to open specified master config file '/etc/asterisk/asterisk.conf', using built-in defaults
th-cam.com/video/pAOgJX5hsqc/w-d-xo.html
noticed here the SIP trunk no longer allows for caller ID spoofing. this is absolute bullcrap. How am i supposed to know how to set it up with other services or which service offers what we need? 3/4 into the video and we cant even use gotrunk. really sucks. can I please get something off you that works?
Same problem here!!
xlite no longer free what is another free alternative that works?
What I want to know is: why do the scammers boot into Safe Mode? What do they think that will accomplish? :/
Probably people freak out when they see the list of the files loading. Also I suppose he set things up so every time the computer opens, it opens in safe mode, so if you don't know how to switch it back to normal mode then you will think something is wrong.
Is Caller ID Spoofing from Android smartphone possible through this Custom-Rom without any paid app by a third party ?
Buddhist (Anhänger des Dalai Lama) have you found an answer?
dude everything is set up but the problem is i am unable to spoof the number. even after removing the variable and assigning the mobile number manually it isn't working. please help me dude ...
th-cam.com/video/pAOgJX5hsqc/w-d-xo.html
th-cam.com/video/pAOgJX5hsqc/w-d-xo.html
Does this still work on Ubuntu 21?
Hey the ssh (Username)@Ip isnt doing anything for me plz help
!
hELP! I get this error after starting my asterisk server : Peer 'trunk' is now UNREACHABLE! And then I get call to '12120001234' rejected because extension not found in context 'from-internal' and DNS error for registration to 27900@trunk
th-cam.com/video/pAOgJX5hsqc/w-d-xo.html
I have some problems I am setting up asterisk server on my Amazon vps and also on Microsoft azure vps but the problem is I'm getting only one way Audio I thought maybe you could help me please
th-cam.com/video/pAOgJX5hsqc/w-d-xo.html
@@VideosPoc doesn't work sip service not made public
Can't connect x-lite to asterisk server (sip error 503 & 480) Any solution?
th-cam.com/video/pAOgJX5hsqc/w-d-xo.html
Hi I've got Asterisk, x-lite, Go Trunk all setup, the problem is the script you used for changing the caller ID is not linked in the description
Nevermind I managed to copy it off the video, works perfectly, brilliant video and tutorial :-)
It actually IS in the description.
It's the pastebin link.
Yeah that's my pastebin link
it is possible to make it with bria 4 ?
someone please help i am on windows and i get this error
Account failed to enable,
Acout:Test could not be enabled
Problem at server(SIP error 408). Try agian Later
fixed it but i cant make calls?
Had it working, but i made a new go trunk account and all outbounds calls from the system now come up anonymous/unknown, as if caller id was blocked. Checked cdr in go trunk, and even there it comes up blocked
same here. Got any idea on how to fix it?
nothing yet
It turns out that go trunk changed its rules on the calling (read in a comment below) so we have to find a different sip service :(
really? how so?
This is also happening to me, can someone help plz!
Please tell me how can I put my own VoIP service that I already bought on the sip.conf file ,I spent hours following your tutorial so please don't leave me hanging
Do you have to pay to do the calls?
Thank you!! I was totally lost in the world of voip. Your vim skills need work ;-)
No problem. ;) But yeah, hahaha... I don't normally use vim.
@@faketechscammers9454 could you also tell me a good free alternative from gotrunk that allows change the idcaller? [my email is necropower123@outlook.com]
Why do you smash your keyboard like that, Im watching this before bed and the exhaustion is making this particularly agitating
Does this work with the rasberry pie? Like can i do everything u do on ubuntu on a rasberry pie?
raspberry pi* yes you can do everything on the raspberry pi, silent p in raspberri and its pi not pie reason = pi mean an infinitaly large number short version being 3.141
I couldn't build the source on my VPS (kept getting terminal size error, because I was SSHed in I guess?) but you CAN do this from the release (apt-get install asterisk) and the setup is exactly the same.
have you fixed it?
Nirethia I literally provided the solution in the comment, mate.
Help me Please!!! it said This Kermel Requires an x86-64 CPU, but only detected an i686 CPU Etc.... Please help me ;(
Oh wow, you must not have a 64bit CPU or operating system. Try downloading the Ubuntu Server 32bit version... not the 64bit. Also, in my steps when creating the virtual machine, you need to select "Linux 32bit" instead. Here's a link to a Ubuntu Server version that might work for you: releases.ubuntu.com/trusty/ubuntu-14.04.5-server-i386.iso
+FakeTechScammers THANKS!! :)
Im having an issue where the calls just say "Failed to establish call". You got any ideas, ive followed the tutorial to the letter and its not working :(
Ive now got it to call the test number, but the test number never picks up, it just rings?
I actually never did technically, the test call number was ringing but it never picked up to the automated system. But i still cant call anybody. The test number only seems to ring and not give the error if I put a 9 infront of the number which is used in businesses if they want to call an outside number rather than another internal extension.
I seem to be getting various errors if i trace the connection on the asterisk cli. Things like
"SIP/2.0 403 Forbidden" and "SIP/2.0 100 Trying" & "SIP/2.0 404 Not Found". Any help would be appreciated. I get the same as everybody else i think where it just says call failed to establish
***** same here :/
I got this error when i did the "make" command. Please help!
collect2: error: ld returned 1 exit status
/usr/src/asterisk-certified-13.1-cert8/Makefile.rules:138: recipe for target 'chan_iax2.so' failed
make[1]: *** [chan_iax2.so] Error 1
Makefile:386: recipe for target 'channels' failed
make: *** [channels] Error 2
i have tried running the make command 4-5 time and it's still giving me the error
It looks like you downloaded the wrong version. You downloaded 13.1 instead of 13.8. Perhaps there was a bug in their code that got fixed in 13.8.
so i have to redo this whole thing?
i was able to fix it. now i can't make outbound calls or inbound calls
me too ...
also, I know the method no longer works, but for those who are willing to purchase a trunking sip, and wish to do so over firertc (such as myself), could you make a tutorial on doing this with Elastix?
wait.. this does'nt work anymore?
not for free. You see, you can still call, but you'll come up as unlisted, and honestly, most scammers avoid unlisted calls anyway. This is because Gotrunk no longer accepts the "set callerid" command shown in the video. You'd have to find a service that does allow it, but I have yet to find one that also has a free trial. However, if you are willing to pay for an SIP service, the setup on your end is basically the same, just with a different upstream trunk provider.
If I got anything wrong, please correct me
@@TurtleSauceGaming can u tell me one PAID sip trunk server who accepts the "set caller id"?
@@googledicas503 I don't know unfortunately
So because of gotrunk, will this only work for 30 days?
You know, I'm not really sure. I think outbound calls use "credits," which means as long as you're calling toll free numbers it's unlimited (and you can top off your credits with payments). The trial and monthly payment would only be for incoming calls. Don't quote me on this, but it seems like that might be it.
+FakeTechScammers Oh shit, that means I can prank call my friends within the U.S and its totally free? Damn son, thanks!