I am working on a project where we make a VOIP phone call to the end user, essentially, exactly like skype does.
The problem is, that all calls from VOIP need to be terminated as PSTN if it has to call a normal Landline phone. In the indian environment, the govt of India does not permit the call (from mobile to landline/landline to landline etc) if it is through a VOIP service, the call has to end in PSTN at the user end and national laws do not allow this, it only allows call from outside the country to end as PSTN but not from within the country.
Is there a work around solution for this? It is a bit hard to phrase this question but if you have any queries please do ask.
Thank you all so much for your help
Mohan
The obvious solution is to route the call to a PSTN outside India and then dial the Indian phone number from there.
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I am trying to build a system with a raspberry pi that allows clients access into a building depending on their membership status. Right now, it uses QR codes, but I want to know if it is possible to add a feature where it uses some technology like NFC or RFID or Bluetooth to detect their phone or RFID card from at least a foot away and confirm they have a membership.
Someone told me I could use RFID, but I am only aware of that being used in short-distance applications, like a card on a hotel door. I am not sure about Bluetooth either, because the phone would have to connect to the pi first, right? Maybe there is something I don't know about. So please offer any suggestions. Thanks
I think bluetooth does good work for tracking user. Since it's the best to handle large distances than NFC and RFID these two technologies are used for low range scenarios, check this link.
In addition, you can check distance(using Proximity and RSSI) and membership status as well. but you need to know how to handle bluetooth connectivity with raspberry pi check this link. as well create an app on that mobile phone to use Bluetooth (depending which OS you're using for Android, iOS).
Regards,
I want to write an application which will be a bridge between VoIP app and phone line.
E.G.:
- I am writing in Skype to user XXX "call to ******"
- User XXX call me back and by phone modem calling to ******
- So I can speak throw my VoIP and phone modem for free (except internet and phone fees)
I thought to use something like this.
The better description is here in Calling section.
But it is outdated and my server part is on Ubuntu
Could you please advice VoIP (e.g. Skype, Viber, WhatsApp, etc) which I can use for such purpose? It would be great to have a client on Android Phone and server on Ubuntu.
Thank a lot,
If I have well understood, the use case is:
A wants to call B through an application running in a mobile device
B has a phone land or mobile line, but not a VoIP one to receive the call.
Bridge between internet and phone lines is to be done at home (A's home) without specific subscription costs, that is to say, without the services of a VoIP provider (I should like here to suggest rethinking the use of a well stablished solution as costs to call phone lines from IP can be really cheap).
Well, there is a lot of solutions for this scenario. I am going to speak about one of them that I consider interesting because it opens the way to a lot of additional communication services.
First, the softphone. To make and receive calls, A will need an application in his or her device. Consider a softphone as Zoiper or Jitsi Meet.
Then, the gateway between VoIp and phone lines. Asterisk can do the work as a SIP server. It is a lightweight linux software with a lot of features. It can switch VoIP lines with land phone lines via FXS - FXO cards (if the phone lines are analogue ones), ISDN cards, VoIP interfaces, bluetooth using mobile devices, etc.
Last, but not least, the connection. Ok, you do not want to expose your gateway to the dangers of all those wicked people of internet, eager to stole your phone line minutes. Connection between mobile and server could be done using a VPN (e.g. OpenVPN), or through a web app (SIP on top of WebRTC).
Once you have the asterisk working at home, you could use it as an answering machine sending email messages with the received audio, as (if your local regulations allow it) a recorder, as an IVR or as a part of a security system, calling sequencially phone numbers in case of emergency.
I recently see a voip dialer for mobile which we can purchase a username and password. And make calls to telephones. I think thousands of people are using this service. My question is how they call to phones from the voip server? Will they take 1000+ telephone lines for calling out ??
From my limited knowledge, VOIP or voice over internet protocol works similar to a normal telephone network, using switching networks. There is probably some interface between internet and a telephone network at some point, from where the call is routed through a normal telephone exchange which sends the call to the desired receiver using switches.
BTW this is more of a question on Networking than Programming! You can try posting it on https://networkengineering.stackexchange.com/
We are working on a web phone application that can make sip calls to other devices and make PSTN calls as well. We use Asterisk 1.8 as our sip server. The SIP calls from the web phone is working fine.
We want to be able to provide SIP to PSTN calling service to our clients and thus require to connect to a PSTN VOIP Gateway. Only outgoing (SIP to PSTN) calls are required for our system. My question is, are there companies that provide such services which can provide us with connection to the gateway to route calls to PSTN without any limit for the number of simultaneous calls. All the companies I had contacted told me about SIP Trunking with a fixed number of ports. We plan to have multiple clients registered to our system and cannot be sure of the number of simultaneous calls required.
I know this is an old question, but I was searching for something and saw this hadnt been answered properly.
All the vendors telling you to get a SIP trunk are correct. Even though you dont know how many simultaneous calls you need, that doesnt matter. The number of ports if the number of simultaneous calls you May end up using. This is a number you need to provide them for ensuring they can dedicate those for you.
Sip Trunks are basically SIP lines that can call over the PSTN network.
Google just announced that they will add support for VoIP calls in its Gmail application.
Does someone know how this will work? Did they manage to write a web-based VoIP client, or will they require the user to have Google Talk installed and somehow (how?) call this app from the browser?
I'd also like to provide customers with a way to make/receive calls through their browser, so that they wouldn't have to install an SIP client.
Thank you.
Google don't use a VoIP client in a browser. Instead the browser is used to initiate a callback to a phone number you must have previously registered. Once you answer that call Google Voice will then ring the destination number you specified and then bridge the calls together.
I've just noticed that in my inbox. They ask you to accept their EULA to start installing Google Voice. So it's not really a browser solution.
There are several companies that have built VoIP clients as e.g. Java applets.
It is totally doable, although depending on the exact requirements it may be expensive and time consuming: for instance, echo cancellation is not exactly trivial when you need to deal with arbitrary audio drivers across any and all laptops, netbooks etc out there.
There are also consulting firms that can help with that.
Full disclaimer: I own one such company ;)