Astonishingly stupid question

[badbet]

Hi all, I have what may be the dumbest question ever asked on this forum.

I am very new to VoIP, super duper green, so please be gentle. Unfortunately/fortunately I've been tasked with standing up 3CX in our office and am struggling to follow along in the majority of these threads.

On to the meat of it: we're using a generic sip provider and I'm trying to understand the role of the 'Registrar/Server/Gateway Hostname or IP' field when creating trunks, as it seems to take any data I place there, with absolutely no effect on the calls being placed. For example, I was laboring under the delusion that I'd need to create a trunk for each location (where we have DCs) calls would be routed through (sort of a primary/secondary failover kind of thing). What I noticed was that, irrespective of which trunk had what IP address, so long as the DID existed on at least one trunk and at least one trunk was capable of receiving calls, the call would complete.

I'm assuming this is expected behavior and I've been thinking about this the wrong way the whole time? In reality, I only need one trunk with all our DIDs and that's it (for inbound). Our SIP provider gave me an IP address, but again, I can put literally anything in the 'registrar' field and it will still work.

I'm sure this is painfully obvious to most and that I've been needlessly overcomplicating things from the word GO. I would be eternally grateful if someone could confirm this for me. I'm feeling very lost and very stupid at the moment.

Thanks!

 

[Mark Phillips]

You certainly only need one trunk as a basic minimum (with x number of concurrent inbound/outbound channels). What do you mean by "DCs"?

You could certainly also have multiple trunks - for example - if you had a US office and a UK office, you'd want a US-based trunk and a UK-based trunk so that calls within each country are handled via the respective trunk. You would also have more than one trunk for redundancy.

 

[badbet]

Thanks for your reply. Can you clarify what you mean by 'concurrent inbound/outbound channels'? I'm still trying to pick up all the lingo. When I say DCs I mean data centers. I'm not entirely certain what equipment is located at the two sites in question, so I'm using that as a catchall.

I'm assuming (in your scenario) that I'd need multiple trunks for a US and UK office because presumably they'd require different SIP providers?

When you say 'more than one trunk for redundancy', how exactly does that operate in practice? That's what I thought I was building between the two trunks I put together, but I could never seem to get either the traffic to route to the alternate trunk, or to mirror the DIDs from trunk 1 onto trunk 2 successfully. It bears pointing out that I did not copy the DIDs exactly from trunk 1 to trunk 2, I tried using variations on the numbers themselves, excluding the area code and replacing it with a wildcard. Calls never completed in that scenario.

 

[leejor]

Depending on what your provider gives you to put in the settings, you may only need the Registrar/Server/Gateway Hostname or IP . This can be a URL, or an IP, some may also require that you use a proxy server as well. I'm not certain as to how it works for you if you put in "anything". The Registrar/Server/Gateway Hostname or IP field tells 3CX where to send calls. Did you put in two instances of the trunk?

https://www.3cx.com/docs/manual/sip-trunks/

 

[leejor]

Are you attempting to route the same DID numbers, incoming, from two different trunk groups?

You might want to read through.
..
https://www.3cx.com/community/threads/inbound-calls-for-multiple-trunks-to-the-same-ip.60545/

Note the comments on enabling Call Source Identification

 

[badbet]

Depending on what your provider gives you to put in the settings, you may only need the Registrar/Server/Gateway Hostname or IP . This can be a URL, or an IP, some may also require that you use a proxy server as well. I'm not certain as to how it works for you if you put in "anything". The Registrar/Server/Gateway Hostname or IP field tells 3CX where to send calls. Did you put in two instances of the trunk?

https://www.3cx.com/docs/manual/sip-trunks/

That's the thing, my provider gave me the IP address for our resources in those two different locations I mentioned, but in testing they didn't seem to make any difference as far as inbound calling was concerned. The first thing I tried was changing the IP address to another known-good IP we use and the calls completed just fine. Then I tested the same DID attached to another trunk (with yet another, different IP address) and the call completed just fine. I then changed the IP address to the word no and the call completed without issue.

You mentioned that registrar field tells 3CX where to send calls, does that mean that that field only applies to outbound calls? I guess I was working under the assumption that 3CX was responsible for some portion of the call path as calls come in but that doesn't seem to be the case for us.

I did not put in two instances of the same trunk, just two different trunks (with similar IP addresses, but not the same) that would ideally act as primary and backup, should the need arise.

Thanks for replying by the way.

 

[badbet]

Are you attempting to route the same DID numbers, incoming, from two different trunk groups?

You might want to read through.
..
https://www.3cx.com/community/threads/inbound-calls-for-multiple-trunks-to-the-same-ip.60545/

Note the comments on enabling Call Source Identification

Heya, thanks for getting back to me.

Sort of yeah, I had trunk 1 and trunk 2. Trunk 1 had all of our DIDs (formatted as *xxxXXXX) and trunk 2 had an almost copy of all those DIDs (formated *xxxxXXXX). The thinking was that, since each trunk had it's own IP address corresponding to where those calls would be coming from, if calls came from the IP address on trunk 1, they'd be connected to the proper extension. In the event that the resource associated with trunk 1 ever went down, 3cx could fall back to the trunk 2 and still connect those calls to the appropriate extensions.

Thanks for sharing that post, that's pretty helpful! As I mentioned I'm still really new to this so I've been trying to figure out what exactly all those options in all those drop downs actually do. My resources are pretty scarce and naturally we're using SIP providers that aren't supported by 3CX so I'm not even sure I can submit a support ticket.

 

[AWS2P]

You are right, no support ticket for anything is not supported by 3CX. Even if those solutions can be fully functional you need to support yourself in that case and with help of others on forum

 

[leejor]

You mentioned that registrar field tells 3CX where to send calls, does that mean that that field only applies to outbound calls? I guess I was working under the assumption that 3CX was responsible for some portion of the call path as calls come in but that doesn't seem to be the case for us.

You might want to have a read through this, to help understand the basics, of how SIP works.

https://en.wikipedia.org/wiki/Session_Initiation_Protocol

https://www.3cx.com/pbx/sip/

https://www.3cx.com/blog/voip-howto/sip-messages/

https://www.3cx.com/blog/voip-howto/sip-invite-header-fields/

https://www.3cx.com/pbx/sip-methods/

In most cases, your provider will give you their URL(or IP) to register with, sometimes a proxy address, a password, and a user name (this is sometimes your phone number, or billing number). When this has been datafilled correctly into 3CX, an attempt to register, with the provider will be made. At that time 3CX and the provider will exchange some information (SIP message), this includes where (IP and port) the provider should send calls to you. 3CX will re-register from time to time, this is normally determined by the provider. It lets them know that you are still available to receive calls
That is the bare-bones explanation.

 

[badbet]

You might want to have a read through this, to help understand the basics, of how SIP works.

https://en.wikipedia.org/wiki/Session_Initiation_Protocol

https://www.3cx.com/pbx/sip/

https://www.3cx.com/blog/voip-howto/sip-messages/

https://www.3cx.com/blog/voip-howto/sip-invite-header-fields/

https://www.3cx.com/pbx/sip-methods/

In most cases, your provider will give you their URL(or IP) to register with, sometimes a proxy address, a password, and a user name (this is sometimes your phone number, or billing number). When this has been datafilled correctly into 3CX, an attempt to register, with the provider will be made. At that time 3CX and the provider will exchange some information (SIP message), this includes where (IP and port) the provider should send calls to you. 3CX will re-register from time to time, this is normally determined by the provider. It lets them know that you are still available to receive calls
That is the bare-bones explanation.

Thanks a bunch! I'll definitely look through that this weekend.

That's the thing though, at least in the case of one of our SIP providers, I was given IP addresses and that's it. No proxy, credentials, nothing. None of my trunks register but that may be because I have it set to 'do not require - IP based'. To confuse matters further, I was told that there are no credentials for that provider (by the provider). To confuse them beyond that, calls complete regardless of what information I put in the registrar field. As I mentioned earlier, I experimented with moving a DID across three different trunks, disabling inbound calling and altering (using actual functioning IP address and then things that I know shouldn't work) the IP address and consistently completed calls (in and out). I'd be fine if this is expected behavior, and if I could say that to my boss(es) with some degree of confidence. But it feels almost too easy, I feel like I should need to give 3CX more information for this to work but once we pointed our DIDs at 3cx with our provider, they worked without issue.

The gist of it is it works but I don't really know why. If that makes sense.

 

[leejor]

There are different types of registration used by providers.

https://www.3cx.com/docs/sip-trunk-registration-authentication/

Yours may be "IP- Based"

 

[badbet]

Yeah that’s the way I have it set up, as the provider said there are no credentials, just the address

 

[usti]

May I ask why you're using a generic SIP trunk instead of one of the many fully supported SIP trunk vendors?

https://www.3cx.com/partners/sip-trunks/

Also, make sure you're using supported desk phones:

https://www.3cx.com/sip-phones/

We go exclusively with Yealink phones because they are solid and just work. I cannot vouch for other brands.

TL;DR
Try to stick with supported everything for your first foray into this. Otherwise you will have a tremendously difficult time achieving success. Feel free to PM me if you want additional assistance.

 

[badbet]

May I ask why you're using a generic SIP trunk instead of one of the many fully supported SIP trunk vendors?

https://www.3cx.com/partners/sip-trunks/

Also, make sure you're using supported desk phones:

https://www.3cx.com/sip-phones/

We go exclusively with Yealink phones because they are solid and just work. I cannot vouch for other brands.

TL;DR
Try to stick with supported everything for your first foray into this. Otherwise you will have a tremendously difficult time achieving success. Feel free to PM me if you want additional assistance.

We're using generic SIP trunks because it's what I was handed and it wasn't really up for discussion to change them, for what it's worth.

We're definitely using supported (albeit legacy/end-of-live) phones.

Ultimately we were able to get everything working, save a few oddities (hotdesking, conference phones, etc etc). Nothing life-threatening.

 

[badbet]

For the curious, I spoke with tech support and they confirmed that what we were seeing was expected behavior. Our providers only gave us IP addresses and so long as we pointed our DIDs at our 3CX install (with at least one trunk), the traffic comes through unimpeded.

Thanks for all the help, y'all!