I did research and could not find any information about how hunt group or busy signal work with VOIP. I understand the traditional hunt group philosophy with POTS lines, you put multiple lines in a hunt group and then if all lines are used, it give a busy signal. How does this work with VOIP? We are using bandwidth.com and I asked them to move one of the trunks to point to a backup asterisk server so we can do testing. I also told him to take it out of the hunt group and he told me they do not do the hunting. He said my asterisk server does the hunting. On my backup asterisk server, I decided to test the concept and called from 2 different cell phones concurrently and both were answered by my backup asterisk server. The backup asterisk server only had one trunk going to it. How is this possible? I thought when you order a trunk from bandwidth, it is for one concurrent call. What would prevent someone from ordering just one trunk and using multiple concurrent calls?
By definition a trunk is a facility to transport multiple lines. A t1 carrier is a trunk circuit.
Circuits are “trunked” together.
A SIP trunk actually carries no traffic, it is used to setup and tear down the call. The actual transport of the media or “bearer” traffic is provided by a protocol called RTP.
The amount of concurrent RTP streams is only limited by your bandwidth and other Layer 2/3 concerns.
The actual limit of number of calls per trunk is simply defined by your carrier based on your rate plan.
Thanks for responding so quickly. This is great information. My confusion is when someone calls my main number, how does it know when to rollover to the next available line as opposed to producing a busy signal. In Bandwidth.com there is an option to Configure Endpoint Ip Address. So I took one of our numbers and changed the Configure Endpoint Ip Address to point to our backup asterisk server. I then called from 2 separate cell phones to that number and both calls went through. It did not produce a busy signal on the second call. How is that possible? Bandwidth could not explain this to me so I am confused how this works.
I don’t know what plan you are on with Bandwidth, the concept of rollover seems to be confusing you.
The carrier simply keeps sending you calls until you hit the maximum number of channels they have configured for you. They are not physical channels, it’s just an administrative limitation.
Here is my confusion:
1)Is a phone number associated with multiple channels?
2)Lets say you had 2 asterisk servers(Main and backup). You have a main number 444-555-6666 and 7 other phone numbers that nobody dials directly. When you log into Bandwidth, you configure the endpoint ip to point to either the main or backup asterisk server. Lets say I configure the main number’s ip to point to the main server. I configure 3 of the other numbers ip to also point to the main server. The other 4 of the numbers ip, I point to the backup server. When first person dials main number, it goes to main server since the endpoint is pointing to the main ip. When second person dials main number, how does it which server to goto?
I think you are missing the point. You could have 100 different callers dial your main number and they would all ring your Primary Server (because that’s where you’ve pointed your main DID). The other numbers are irrelevant in this discussion. Call to you main number do not “roll over” to the other DID’s - they continue to ring your server on the Main Number DID.
Atcom is right, why would you have numbers that nobody dials.
Again, the calls do not roll over, they simply arrive on your trunk. The Dialed Number field is populated with the DID number and is used by inbound routes.
In the non-IP world this is the same way a PRI circuit works.
The converse is also true, you can have 1000 DID’s and only 10 channels on your trunk. Once the 11th user tries to dial in they will receive the busy treatment from your carrier.
So what if you have a trunk that your provider says will only support a maximum of 1 channel? Is a hunt group an option then? I would think not.
In order for this to work I would have to setup a trunk to represent each channel on the PBX and then the provider would somehow need to configure the hunt functionality on their end.