Help needed for a use case: A very-small travel agency. First steps? [newbie]

Hi guys, this is my first post and I’m unsure where to ask, even if maybe I should ask in other non-asterisk forums.

I need help in understanding super-basic things. You’ll find maybe those are silly questions, but I’m not sure where to start.

Here I go with the problem description: I am running a travel agency startup. We sell trips. As we still don’t have investors, we don’t have an office with employees, and we are a bunch of collaborators working from remote locations.

I have the following needs:

  • I want my collaborators to be able to call the travelers from their home with their mobile phone (Android) and that the traveler sees exactly the same “source” or “from” number (sorry, IDK if that’s the terminology) so we have 2 effects with this: a) No matter what agent is calling the traveler, the traveler feels a coherent image, b) This hides the agent’s real number so the don’t see their home-life compromised by traveler’s calls.

  • I have a deal with a call-center. This call center already uses VoIp solutions. I’d like that each time a traveler calls to the number that appeared on the first paragraph, it is routed to the call-center.

  • This call-center is not 24x7, only operates mon-fri x 12h/day. I want that any entering call outside the agreed time-table gets into a virtual PBX so I can configure if I want a robot to answer or I want to route it to me.

  • I want all incoming and outgoing calls to be recorded, with no exception.

What I have learned up to now:

  • There’s something called SIP address.
  • There are phone-number providers that create a number for my company and this can be routed to any SIP address. I think this is that is called “DID” or something like this.
  • The same number can be used as a “display number” when calling going out. This is what I think it’s DOD.
  • There’s somehing called asterisk which is a software that Mark Spencer created and got to be the most popular and community-maintained core for the VoIp infrastructures.
  • I think asterisk acts “as a server” so it does not have any fronted “per se” and there are some cooler or not-so-cool frontends one can “connect” to it.
  • I feel asterisk is “everything you need” if you are able to deal with files, configurations, and you are a server-guy, most likely if you configure an apache or a mysql.
  • There are images (like “Box” I think) that are like linux distributions with all you need out of the box.
  • I think that any connection from SIP to SIP and any VoIP routing is always free until you reach a termination point to go to the “outside”.
  • I feel there are no free numbers and there’s always a cost for “having” a number.
  • I feel there’s no cost for “incoming calls” from the physical telcos and there’s always a cost for “outgoing calls” to the physical-telcos.

Having said this, I have all my infrastructure for other things (websites, databases, document storage, emails, etc) in Amazon AWS. I’d like to setup my asterisk in AWS too.

So here we go with the questions:

I want to play around with a dummy infrastructure to get used. I’d love to have this setup:

  • My mobile phone will simulate the call-center.
  • My son’s phone will simulate the traveler.
  • My daughter’s phone will simulate an agent.

I want to have “something” in AWS so that

  1. When my son (the traveler) calls a certain number (let’s say the company’s number), It’ll ring into my phone (the call center).
  2. When my daughter (the agent) calls my son (the traveler), he sees the company’s number.
  3. When I dial out from my phone (the call center) to my son’s phone (the traveler), I use my flat-rate in the telco I have, and I don’t mind if my son sees my real number, but the call still gets recorded.


  • Are my learnings above correct?
  • Are there any places I can get “numbers” from a kind of pool for testing purposes, but paying “as you go” instead of “monthly fee”?
  • What’s the simplest way to have an asterisk in AWS configured to do so?

I will have a dedicated t2.micro instance only to those testing purposes, so I can start, stop, terminate and playaround with the full server if needed.

If you guys feel this is not the appropiate community, please help me find what forum should I ask in.

Thanks in advance!!

There are too many questions to attempt a complete answer, and you should go to for a more complete overview.

Asterisk is not limited to VoIP and, in fact, was originally not created for VoIP.

Some of the issues you mention relate to commercial policy of service providers.

I hate the misuse of the term DID by the VoIP industry, but yes, what you describe is now marketed as DID in that industry. In the general telecomms industry, it stands for Direct In Dialling, which means that some of your local extension number digits are forwarded to you and on set of incoming lines can directly call any of a much larger number of internal extensions, without the caller needing to talk to an operator, or key the number in in two s stages.

I’ve never heard of DOD, but the default position of most service providers is that they will ignore any caller ID you try to provide and will use the one associated with your connection to them.

Thanks david551 for the pointers.

At this moment, what I’ve done (just in case another newbie interested in):

  • Install a raw ubuntu 18.04 LTS in Amazon AWS with UDP port 5060 open.
  • apt update
  • apt install asterisk => to install and run the server
  • asterisk -r -vvvvvvvvv => to connect to the running server (for example I did not know asterisk had an own CLI!! this has been a good finding!)
  • help in the asterisk CLI and start playing around.

Then I started to follow these two documents in parallel and try my own initial config:

  • Particularly I edited only the sip.conf to add users and extensions.conf to add extensions and with that I started to have something.

At this moment I got 3 soft-phones to register in the server. Still having some trouble with the voice-channel…

But I have stuff to play around a while.

I think we can close this thread. I’ll ask more newbie questions as I try to do something :wink:

Thanks guys.