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Setup voip server : How to setup a voip phone system | Setup Asterisk with UBUNTU & AWS | SIP Server
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- Опубліковано 5 гру 2020
- VoIP is one of the most populer technologies in telephony industry. VoIP works on SIP or Session Initiation Protocol. SIP works on TCP or UDP. In this video we have set up an AWS free tire ubuntu ec2 instance, and installed asterisk SIP server. And, then added users to established phone calls, between two users.
So, here you will get to learn:
1. What is VoIP?
2. What is SIP & how does it work?
3. How to launch an ubuntu ec2 instance in AWS.
4. How to install asterisk SIP servr in ubuntu?
5. Edit asterisk configuration files and add users in asterisk server.
6. Set up sip phone in android and desktop.
7. Establish a call between two phones.
After watching this video, you will be able to set up your own telephony network.
GitHub Link: github.com/kou...
Asterisk Commands:
===================
asterisk -vvvr
module load chan_sip.so
reload
sip show peers
#sip #voip #asterisk #aws #channelcodeboard
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excellent in that we can get the call up and running after watching this 10 min video; rather than 50+ video tutorials from the german technician with aweful english.
Thanks 😊
@@CodeboardClub hi can u help me create one pls
That's very low of you to speak of someone who has taken time to create useful content. The so called 'german' technician you are referring to takes time to explain each and every action while building a VOIP solution, that is what learning is about... understanding. Not just about having a system work without knowing why you have to make certain decisions, and in the process making it easy for you to troubleshoot. If you build something you are unable to troubleshoot then you end up being a half baked technician creating unsecure systems that can cost an organization a lot of money.
I feel like it's no exaggeration to say you are a godsend. I've been making my own soft since i was 14ish and since then it's beco my
Very good information. Subscribed to the channel already. thanks man
eventually it all snapped into place and I started learning how to add all the effects, titles, motion text. It was pretty cool to see my
hey in this video you used text to speech synthesis to narrate it ,... can you tell me which voice is this and where to get it from ?
Would be interessting to see interfacing with other SIP Gateways, like if it would be possible to have this setup you showed and have the Asterix connected to public phone line (e.g. Telecom of India). So when I call a regular phone number, and the line is avialable to the outside, than i can call through. Kinda using Asterix as Proxy. Calls from the outside go to all internal clients, first to pick up wins (^^).
Nice video. was able to set up the same setup without any issues. Thanks!
if you are going to use this then set Tenancy must be "Dedicated" except "shared server" .
Great tut. Can you make a video on how to connect to the PSTN, so that one can actually call real-life phones?
is it possible to use mobile device to terminates call via sip server or voipswitch?
excellent waiting for other videos for this topic
its working, i use it for my personal server, 😊
I am unable to upload sip.conf files in directory as its showing sftp permission denied. Did u get any errors like that?
wowo I like this video, I'll try out myself and comment again... Thank you.
@Codeboard club Hi!
your video was very meaning full. I likes your video. It was very valuable information. but I have question that ( like One office that have 20-30 of total employees and they couple of floor's, everyone have internet connection on their Desk. Now how can they communicate each others via CISCO Or Grandstream IP Phone without Any physical server.)
@Codeboard Club I know you are Care's others emotions.
TNice tutorials was very helpful thankyou.
Thank you for this video. It is really great explanation. 🙏
Can you only call numbers on the sip network, or can you call any phone number in the world outside of the sip server e.g. just a random phone number?
Please make video how to test these calls and take trace on tcp dump and then analysis on wireshark
Very informative, I am trying to build one that works only on my LAN. How do I modify the sip.conf file because soft phones are failing to register. The Sip Server resides in my Ubuntu PC
Video for ASTREKS outbound calling
What if you put this behind load balancer?
Question, are you only able to call the numbers that you've set up within the sip server or can you call them from a mobile or landline from any network and if so, will it work vice versa?
With VOIP, you can typically call any phone number, regardless of whether it's set up within the SIP server or not. SIP (Session Initiation Protocol) is just one of the protocols used for VOIP communication.
If you have a VOIP service configured on your mobile or landline device, you can use it to call any phone number, whether it's on a VOIP network or a traditional landline or mobile network. Similarly, if someone has a VOIP service, you can call their VOIP number from your mobile or landline device.
In essence, VOIP allows for interoperability between different networks, so calls can be made and received between VOIP networks and traditional telephone networks.
The best video about voip server config. i have tried to load the link to the sip code for the clients but it does not register on asterisk server. when you run the cmd sip show peers, it says ''no host registered'' Do you know another way to do this?
really great video and excellent presentation!
Thanks
Need more video on Asterisk.
how u change the CLI command to ubuntu@ipaddress? @ 5:45 Also,it showed no devices online after "sip show peers" command? Note: I have done changes in sip.conf (externip)?Pls resolve this
Sir it is possible to set a system in a way that I am able to receive and make online calls from iphone but the person who I call receives it on a mobile sim .?
Hi, it's really a great video and clear guidelines.
I have one issue here. When I call from 7001 to 7002. Call received by 7002. Issue here when I talk from 7001 , 7002 able to listen my voice but when 7002 speaks 7001 not able to listen. How to fix this issue. Fyi I'm using android linphone client in both devices.
My Ubuntu server running at Google cloud platforms
how did you solve this issue
Thanks, this video is very helpful for beginner. Do you have a guide to dynamic add sip clients and save CDR to MySQL?
Nice video, thanks! What are you using for narration?
Great video, excellent explaination. How do we enable video calls and billing?
I tried to send HEP of asterisk to Homer, but seems not data capture via Homer. Do you have tutorial how the hep can be send from asterisk to Homer? Thank you
I've tried folowinbg this and was good to the point I tried to load module chan_sip.so at which point I got the error "Error loading module 'chan_sip-so": /usr/lib/asterisk/module/chan_sip.so cannot be open or shared object: No such file or directory.
while overwriting the sip.conf, extensions.conf, voicecall.conf file, i am getting error "Open request has failed with SFTP error PermissionDenied: Permission denied." can anyone help me out
Having the same error. Did u resolve it?
please help, how to set up the linphone application so that it can be used to send messages/chat? please help i really need it.
Thank you so much, great tutorial.
Hi, this is a great video with excellent explanation. It is possible to have another video for adding number dynamically and allow api to create acc, add number and call and call details ?
Good idea. I will try to make one video on that.
@@CodeboardClub i am looking for a custom software for voip and sip would you be looking to work as lead on it?
@@CodeboardClub hello bro place halp me
Nice video.Thanks!👍
Thank you for informative video. Is this setup workable for Video Call or different config required, if yes could you pls help setup video call.
Can you make a video how to use smtp for another computer ..
Very Nice information, Thanks
Thanks
Very good video
Smart video
this is a great video
thanks you your easy explanation
Do you have a tutorial on how to do the same in version 20, got an error on trying to load the module?
Where you got this nice voice to your video? What software you are using to edit it? Your answer would be aprecciated.
really superb thanks
Thanks
Also can u elaborate how u created profile 1 in bitvise SSH ? like I'm creating the first time host key ...directions needed and whats the connection between this and sip pem file?
Just couple of steps. Pls check Google how to create public key profile in bitvise ssh. Thanks.
@@CodeboardClub ok thanks
Hey, did you able to create the key? If not let me know.
@@CodeboardClub No...can u help me up?
Oh okay !
I've detected your real voice 😀
Which software do you use for the voice??
Can we run this on local LAN instead of public IP
It works. Thank you!
Is it possible to interconnect GSM sim using switch then make international calls but call rate will cost will domestic?
Good Exam
Great tutorial...thanks
Thanks
@@CodeboardClub Hi...from sip phone got video and IM...just to check if video and messaging features also works with the current setup? Any additional setting required? Thanks
U earned a follow for this ❤
Thanks for this video, really helpful, do you have any contacts?
Sir I had A Doubt... Can't we call to Normal Mobile Numbers
Please...respond to Me Sir..🙏. I am waiting for the above Video... When i was in need of voip... Sir.. Please Respond to Me Sir.
You can, but setup will be a bit different in that case.
@@CodeboardClub Can you Please Guide me through it Sir...Please
@@CodeboardClub when you are going to made video on this topic it will fetch more views and subs to your channel
hi am unable to connect from linphone to Asterisk voip server on ubuntu 20 on oracle cloud infrastructure.any settings to be made on oracle cloud
Excellent 👌
Thanks a lot 😊
Thanks!
Thumbs Up bro, but show us how to setup and call landline numbers.
Great ! Can you make a video on how to connect to windows pc.
Hello, Thanks for this video, works fine. It is possible to make a video call ???
Indians' tech are so danger :))
😊
How long did it take for you to compile the aestrix , I used sudo make -j2
and now my cpu is 100% for 1 hour ....and frozen
You are using AWS free tire Ubuntu? It should take 10 - 15 mins max. Never more than that.
@@CodeboardClub yeah found out just on small instance 1gb memory not enough needed 2gb
Hey man. It's a really nice video. Could you briefly mention the server costs associated with the calls. Like whats would be the cost if we have 1000 users or 10000 users or say a million users.
Hi, great explanation. How can we use this for actual phone numbers ?
Hi bro is that video true we can call for free just with internet?
Türkçe altyazıyı koyan kişinin eline sağlık ö-ö-ö-ö-ö-ö-ö-ö-öptüm bayy, gö-gö-gö-gö-gö-gö-gö-gö-gömmdüm say
Thank you!
Hey! I'm new to FreePBX. I've set up Raspbx on a Raspberry Pi 4 and would like to make calls using a 4G SIM router (TP Link - TL-MR6500v/ VoLTE, VoIP, VoiceMail). The router works fine on the RJ11 port with a standard phone.
I previously attempted to use the chan_dongle module, but it appears that the 4G SIM doesn't work with it. I don't believe the mobile carrier provides any SIP settings for 4G SIMs. Is it possible to connect the VoIP router with Raspbx over Ethernet?
bro can u suggest any cheap product for 3g calling. my isp don't support volte. i want to make calls on 3g. openwrt + with device can we used?
i need a vidoe on soip with xml call
Hello,
Which software do you use for the voice?
So if we need to connect PSTN, we need an asterisk card?
the extension files are failling mate...it's say sftp no have permission to overwrite
Were u able to resolve it
connectoin failed
problem
how can create client key on Bitvise to connect with EC2?
Can I able to call from my VoIP number to external phone number like jio
Thanks.
Can I integrate asteriks with node js
Thank you.
How about pure PC to PC. Do you have a video about that? Or I have to install an android emulator on my PC
I think you can use microsim. or linphone on PC.
Can anyone let me know, whether this method works with Mac?
I execute all the commands properly but after I use sudo systemctl start asterisk it shows "system has not been booted with systemd as init system (PID 1). Can't operate. Failed to connect to bus: Host is down"
can you please help me with these?
What server are you using?
@@CodeboardClub windows. I’m a totally a newbie please bare with me
Hello sir
Can I connect 8 VOIP phone to one Server (via Ethernet Cable) and use this system locally??
Yes, you can
@@CodeboardClub thanks
Thanks for this video
😊
@@CodeboardClubwaiting for more existing videos bro😊
Are you only able to place calls to external numbers? Or is that only for numbers connected to the server?
No
I got permission denier error when uploading the configuration files using SFTP.So I have tried "sudo chmod 777 *".
Now i am getting another ubuntu error like below:
sudo: /etc/sudoers is owned by uid 1000, should be 0
sudo: no valid sudoers sources found, quitting
sudo: unable to initialize policy plugin
can you help me
Please send me error screenshot via mail
You need to add the account to the /etc/sudoers file or just use root.
Ah! Congrats.
But I feel stupid. I thought Asterisk would be able to have a phone number so people can call to?
thank you
hello ...anyone thr ...i am having problem with bitvise login
How do i find aws public ip address, answer plzz.
Can I have a discussion with you ?
How video call can be made ?
Hi I mangged to install it and set it up exactly like your tutorial i can call from 7001 to 7002 and from 7002 to 7001 but there is no sound .....
Ok. Send me all config files via mail
@@CodeboardClub Hello I am facing same issue.. I used same config files which you provide
How can we do voip to local sim call
I manage to have everything running but when I’m uploading configuration file it says permission denied
Please run "sudo chmod 777 *" in the server (for testing purpose only), to solve permission issue.
@@CodeboardClub i got the same issue and tried this solution. but i am getting another error like below
sudo: /etc/sudoers is owned by uid 1000, should be 0
sudo: no valid sudoers sources found, quitting
sudo: unable to initialize policy plugin
mine is not working, can someone help me
And how do I add this on my app?
I use PBXnSIP using windows
And a like 😂
ok can we make free calls to anymobile numbers in India? with this method
For that you need to pay money to the operator, with which you want to connect (airtel, jio etc.)