Answer the question
In order to leave comments, you need to log in
Combine VoIP Node.js SIP client into a single service to connect the call to the required account?
There is a task to realize the following goals:
1 there are two parties A and B , which should not know about each other (do not own real contact details)
2 when party A calls party B - in fact the call gets to our service, the task is to combine this stake with the party B- find a connection in the DB and most likely combine it with a conference connection
3 the same situation applies to SMS
There is quite a bit of information in this bundle and from what I found I see two solutions:
1. install your own PBX - SIP-trunc + Asterisk- but here I didn’t fully understand the moment, we can sort of get a call on the PBX and, in theory, make a request to our service in order to get a contact with which you need to connect, and then the PBX will connect the same picture with SMS to the subscriber and this whole logic is described in the Asterisk
config
2. there are already services that have their own clients in t / h / for Nodejs that we can use: as I understand it, the call will fly directly to our service to the client and we can further use the API of this service to connect with the necessary contact
This should be implemented on the server side under Node.JS , UI is there but for other purposes.
The question is whether I correctly identified two possible directions for implementation?
Who had the experience of such a task, maybe you can tell how they implemented what you should pay attention to, maybe what resources you should familiarize yourself with?
Thank you in advance for the answers and perhaps for the criticism!
Answer the question
In order to leave comments, you need to log in
In order not to shoot number B, you can use a fictitious number stored in the database and tied to the subscriber.
In asterisk, any call can be processed in the dial plan file like this:
exten => _FAKENUMBER,1,Dial(SIP/trunkname/REAL_NUMBER)
i.e. calling the asterisk to the FARENUMBER number, the call will be transferred to REAL_NUMBER.
The dial plan file can be generated dynamically by a script.
You can put the lua module on the asterisk and add logic to Lua.
You can, like me, write in the AEL pseudo-language. Accordingly, the type of dial plan will determine how you will act - each time you will generate a new extensions.conf or access the database directly from the AEL / LUA script.
Didn't find what you were looking for?
Ask your questionAsk a Question
731 491 924 answers to any question