I’m new to building my PBX and I recently converted 5 ISDN lines to VOIP accounts
I don’t have a SIP trunk, just five VOIP accounts with username/password.
Hopefully this is supported?
The reason for the PBX is that my SIP provider doesn’t allow two phones to register with the same number. But I need two phones to ring if a number is dialed. In my understanding this should work with FreePBX.
I’ll figure that out later.
First I need to register the five VOIP accounts. I tried two so far, setting each up as a new trunk. I suppose I should use pjsip.
It looks like both trunks register with the same settings with the SIP provider:
When using SIP phones to register directly I get:
When I registered a second trunk it registered with the same settings (same port), which probably will not work.
What do I need to change so that at the SIP provider end all five accounts have a different registration?
If you want go SIP instead of ISDN, you don’t need 5 VoIP accounts, nor 5 trunks.
One would be enough. The type of plan with your provider will then determine how many concurrent calls you can make and receive.
When your PBX registers to your VoIP provider, then you have a SIP trunk basically.
Some providers do no support PJSIP and even if they do, that can be problematic. CHANSIP is still preferred, at least for trunks.
thanks for your reply.
Let me explain the requirements:
I need at least three numbers:
- One published in the directory and mostly ignored because of junk calls
- One for friends and family
- One for a FAX ( I know, old school)
For the first two numbers I need two phones ringing if receiving a call. I’m fine with the FAX going only to one ATA
So in order to get two phones ringing on the first two numbers I suppose I can create a ring group.
But the question still remain:
How do I register five VOIP accounts (no SIP trunk) so that they all have a distinct registration URI?
What you are describing is a single “SIP Trunk” with multiple channels (if you want concurrent calls going on). So if you want to be on the phone talking to someone and still receive a Fax at the same time then you need two channels. If you need to talk to someone while your wife/kid/friend also talks to someone and receive a Fax at the same time you would need three channels etc etc etc. This is by far a better setup than trying to cobble together FreePBX to send receive the traffic on a SIP account instead of a Trunk and far less expensive too! A single DID can be dialed by 20 people and have 20 people in your house/business talking at the same time.
When I first started working with SIP, PBX’s and so forth, I was stuck in my head wrapping my brain around this concept. I could only ever associate a phone number with a line you can pick up and dial and talk and hear. In the world of VoIP things are so much more than what you see. All of a sudden it is realized that a numbers (DID) only points the person dialing it to someplace. If that place that is being dialed is a copper pair then you can only have one call going or possibly two if you have call waiting. But if that place the number is pointed to is a Trunk then the sky is the limit. One DID can pointed at a trunk and all of a sudden you can have hundreds of calls inbound and no one dialing is any wiser.
You would simply get 3 DID’s (Direct Inward Dial) phone numbers. When someone dials the DID it would flow into your machine via a channel on the trunk. If you have a provider that bills by the minute then you dont need to pay exorbitant rates for an unlimited channel for each concurrent call. $20 - $40 - $60 per month if you only make/receive a couple thousand minutes a month.
I personally use Flowroute due to the fact I pay $1.25 a month per DID and I have such low volume of calls I generally pay about $3-7 a month for my usage. They allow unlimited inbound/outbound channels! So you could have a dozen people making calls at the same time. I pay about a penny a minute for inbound and outbound calls thats many thousands of minutes before a few $20 flat rate trunks would be justified.
I do not work for Flowroute nor represent them in any way. This is solely my opinion based on my research and my business needs. Please look around and do your research as to what you find will work best for you!
I think I understand the concept of a SIP trunk.
The problem is that I’m looking for a low cost solution (phone is rarely used). Getting a real SIP trunk would triple the cost. I’m living in Switzerland where everything is expensive
That’s why I have 5 simple VOIP accounts at the moment.
If I register my SIP phones I can see that each of them talks to the provider back end at a different port. Therefore inbound calls are getting routed to the phone registered under one of the numbers.
My Yealink phones support registering two phones under the same number (account), but my provider doesn’t support that feature.
Therefore I need some “magic system” which would register all five accounts under different outgoing ports and then would allow internally distributing the calls as I define it. One of the rules would be something like a ring group, so if one number is called then two phones would ring, allowing the call to be picked up from any phone.
I’ll have a look at Flowroute if this can do something like the above.
Thinking about that, I probably would need something between my provider and FreePBX.
With the provider it would register the 5 standard SIP accounts and to FreePBX it would look like a SIP trunk with 5 DIDs.
Okay, with this being the case, I think you should be able to register your PBX individually to all five accounts and treat them like five separate trunks. This is a highly unusual trunk arrangement, so expect some difficulties getting the initial setup correct. You would have to ensure the register string for each trunk has the correct port. You also probably want to your trunk type set to ‘peer’. I would start with chan_sip, but if you can’t get that to work, you can try pjsip. Also know that this is all Asterisk not FreePBX so you might wish to seek help elsewhere as well.
You will have trouble if you use SIP with 5 VOIP accounts to the SAME ITSP because of the the way asterisk is matching the SIP connections (ipaddress;port) only. I have similar setup with 3 voip accounts to the same itsp and I had to use PJSIP with the “line” operative. What it does is it handles the different accounts within PJSIP and returns as different trunks to asterisk.
See this thread: on how to implement it in Freepbx 13.
you are the man.
That’s exactly what was missing and I could now register several standard VOIP accounts to the same ITSP.
Thanks a lot