I have a job soon to replace a mitel system with sip pbx in a hotel, we are using gateways for the rooms old phones, well actually using 6 of them.
how will they talk to freepbx? OK I know you tell it which SIP server and extensions etc, so they talk to freepbx, but where I am at a loss is, and dr google doesnt help explain it, some youtubers who are well known with sangoma - or so they claim, say trunks need to be setup as well, so I think is this crud? is it gateway specific, or is freepbx lacking?
a sip handset registers with pbx, for incoming and outgoing calls, the pbx knows where to send my calls to a did, and handles my outgoing.
the gateways are supposed to just replicate this right? So why should a trunk be setup of freepbx to say go-to-gateways if they are already registered by ext/authid/pass in each extension in the gateways?
Which brings another question, how does freepbx know which gateway to send the call to if it relies on trunks and not the sip registration, does it send it to all the gateways until the appropriate one responds? Even though it should see the endpoint sip logged in through sip regisitration of that extension
Are the gateways dumb? or is this the fault of asterisk/freepbx?
Or have I been watching wrong or outdated youtube videos, and trunks are not needed (apart from the main upstream trunk provider) - which is my way of thinking, but im new to using gateways. Most my experience is setting up small standalone sip networks for small businesses
Seems like you are very confused with FXO and FXS.
FXO connections (usually POTS lines) will need a Trunk, FXS connections (Usually Analog Phones or Fax Machines) will authenticate as any SIP Phone/Device with a regular extension.
So in your case, you can have a ATA for each analog phone and register them with their extensions to the PBX. Or you can buy something like a Sangoma Vega Gateway.
No. I believe that the OP understands that he’s buying FXS gateways with one port for each guest room. Although ATAs and smaller gateways with up to 8 ports appear as one extension per port (an 8-port gateway would have 8 active registrations), some larger units including those from Sangoma have an option to use a single registration and act as a trunk.
Setting up the gateway as a trunk reduces the load on the system, but is considerably more complex. Most hotel systems have very little traffic (guests mostly use their own mobiles) and I would recommend the extension-per-port scheme because of its simplicity.
@nickzed , which gateway(s) are you considering for this project?
The gateways should be 1:1 (port/account to phone), never the trunking option. As well do your gateways currently support 50-pin connections? In every hotel I’ve done this in I just use a 16 or 32 port FXS gateway (depending on it’s wired) because each 66-Block has 16 lines per wired to the analog devices. Be it a room, pool, lobby, etc.
Yes, in that the connection method for the gateway and your phones will determine how to connect it to the PBX.
If the phones are SIP phones, you can connect them to a standard data network (10/100, for example) with POE (to power the phones) and define each phone in the system.
If the phones are ‘standard POTS’ phones, you need a gateway or channel bank to run them. These will connect to FreePBX through a T1/E1 interface, or will connect as a SIP interface. If the former, you’ll get 24 channels of phones per back and each will have a DAHDI channel assignment that you set up to talk to the extension definition. If it’s a SIP connection, you’ll identify the phones through the syntax that the gateway wants you to use.
An important programming note that might not be completely obvious is that your outbound and inbound “outside” connections (from your ITSP, VSP, or from a couple of DAHDI interfaces) are handled distinctly from the phones. It really doesn’t pay to confuse them or even talk about them in the same breath. Do one, then the other. Don’t try to get both working at the same time - one thing at a time will make your troubleshooting go easier.
You are not doing yourself any favors with this question. The extensions talk to the PBX, and only the PBX. When you make a call, you aren’t connecting “out”, you are connecting to the PBX with a request that IT call out for you.
Next, DID is a specific term. It is the number that people call to contact your PBX. You don’t call a DID - the “I” even stands for “INBOUND”. When you are talking about your DID, you are talking about a VSP sending a call to your PBX. Your PBX will do with it what you instruct.
There is no much mish-mash in this question, I don’t really know where to start.
When a call comes in to your PBX from the outside to one of your Trunks, it is passed to your Inbound Routes. These process the call and look for a local “service” (think extension as an example, but there are lots of other things it can be) based on the information in the Inbound Route.
When someone wants to place a call, their extension sends a call to the PBX, which then (based on various conditions internal to the PBX AND your Outbound Routes) sends the call to the destination. In the case of a call to another location, the Outbound Route will code the phone call to be processed by the Trunk that is associated with it. Outbound Routes are handled in order presented, and if the call “matches” with the conditions in the Outbound Route, the call will be processed.
There is a smidge of “cagieness” in there, because some calls from extensions go to places like “Voice Mail”, which obviously is an internal process within the PBX.
Sometimes, gateways are necessary. They act as a bridge from your PBX to another technology, giving you flexibility to do a lot of really interesting things to the mix. How your gateways are connected and what they are doing for you will determine if they are useful in your new implementation.
You’re correct, pure FXS gateways,we woud prefer not to go down trunk route for simplicity, and again correct, very little call usage, but they want to retain it for calling kitchen, reception housekeeping and so on, but with little usage rewirring for ethernet is not cost effective for them.
We are looking at the sangoma vega 3050, and a grandstream 48 port unit, from what I’ve read the Vega has more REN output so likely go that path.
So we do little more than tell the vega ext/authid/pass and have it log into freepbx to matching ext, is this down to FXS port, or gateway setting on Vega? seems to be little info on that?
Thanks (I did think it was overkill having trunks)
I qgree on 1:1 which is why I was sure I was watching wrong information, yes they ones we are replacing existing with have telco50’s , but we are not in USA, we dont use 66’s, we use krone’s
Thanks, it seems you did not understand my question, where others did and made it much clearer, I am also no newbie to sip. only a newbie when comes to configung FXS gateways, I know all about normal trunks, routing, DID and so on and no i wasnt getting it wrong, some of the rooms are long term leased to minning, they pay to have their own DID to their extensions so does not go via reception.