Hi all, First I need to tell you that I have almost zero experience with SIP servers. I am familiar with VoiP networks, DHCP options, and the like, but after they are on the network and functional I am not longer involved. Until the following situation came up, that is… I figure I need to dive into this as some point so may as well be now! On an aside - please feel free to share any resources that would help me get my head wrapped around SIP servers and VoIP technology.
Project advice sought - I need find/program a sip server (I think) for an installation that had one running on a RPi. Somebody removed it and the system stopped working - imagine that. They are asking me as the IT guy (me) to fix it… These are GS phones (5 or 6) with large display and are about 3 years old… (edit: the phone model is GXV-3275) running Android? There is no phone service per say - this was setup to answer calls strictly from 2 Axis outdoor door stations with IP cameras - video door that would ring the phones inside when somebody pushed the button.
I am good with Raspberry Pis, and the networking part of this. Any advice on how to get his back up and running. Should I figure out Freepbx and figure out how to program it? I only know the name perhaps there are better solutions. Any other way I should do this?
Thanks in advance for any/all help and sorry for the long post!
It is an Axis door station - can’t recall the model. Thanks for the help. That may be the way to go on the UCM6202. Reasonable price - appears to be less hassle than a Pi.
There is a very active Grandstream community forum at forums dot grandstream dot com
Another great resource is the DSLReports VOIP forum at dslreports dot com/forum/voip
If you take advantage of all those resources, in very short order you could become Grandstream/VoIP wizard. Once you start down this rabbit hole of discovery you may find yourself wanting to enable more and more features.
If your company is currently running an analog pbx. You may want to evaluate exactly what they are running. What pbx are you currently running, how many fax lines, how many DID’s / DOD’s, peak concurrent calls, current price per month they are paying, how many minutes per month incoming / outgoing, how many extensions, how many physical desk phones, what model are the current phones, how many concurrent conference bridges, etc…
You may want to use this info to perhaps buy an IP pbx that you can grown into, and can support a complete migration. Instead of buying the least powerful ip pbx to only support the camera system.
You would also need to take a look at your network infrastructure as well (I can go into greater detail if you like). If you decide to go down the VOIP rabbit hole just know their is tons of resources available to help you.
Also find out the model camera you using and make sure it is compatible… depending on what you are trying to do, you may be able to connect the camera to the phones directly and not even need pbx (peer to peer).
I did a quick google search for “axis door station grandstream” and found people discussing various installs of the Axis cameras on Grandstream equipment.
From those couple discussions I read, you should be able to do exactly what you are wanting… Plus you have the added benefit of joing those forums and communicating directly with the users that have installed the same setup.
I would air some caution running a PBX on a pi since you might get a bit of jitter running it on one. although you mentioned you had it already running on a PI so I presume you had no issues. I also ran into jitter issues running FreePBX on a VM a few years ago too. Only had 5 phones and 2 SIP trunks on it too and the performance was terrible.
I would always advise running a PBX on a dedicated piece of hardware with a decent number of CPU cores for codec processing.
You’ll probably get away with using GSM codecs since they are small, but use any of the uLAW or aLAW codecs at 64Kb and your asking for trouble on a PI or VM, in my honest opinion.
Tom’s right with FreePBX. I’ve used it for over 10 years with Cisco phones running the SIP OS on them and it works great. I have my PBX running on an IBM x3250 server with a single CPU and 8GB of memory, so the hardware doesn;t have to be top notch to run it.
The new PI’s might be ok to run it on, but my experience with the Pi 3b led to me realize it’s probably not man enough to run a PBX. I’m also not sure how easy it would be to install FreePBX on a PI and possibly the VM route might be the easiest option to give it a try and explore it’s potential.
If you run into any issues with setting it up then I should be able to guide you through it as i’ve set a few up myself. The trickiest bit is getting the SIP trunks to work properly through the firewall with NAT. That’s always been where i’ve had the most issues, but as far as setting up the extensions, it’s a piece of cake since the Web GUI on FreePBX makes things very easy and it’s fairly configurable too.
@Bionic Thanks for this - great stuff. I think wizard’s apprentice would be aiming high for me! The documentation does take a great deal of the mystery away.
Network infrastructure - can you give me a 50K overview of this. Are you speaking about VLANS, LLDP, DHPC options and the like?
I may pick your brain on more resources if I need them, but looks like once I know where to look for quality info it is not too hard to find. This appears to be a pretty mature category.
Thanks again for your help and candor - I really appreciate it!
@ad4m1 Thanks for the info Adam. Yes - the Pi3 was in there for quite a while with no issues. I would hope the 4 would be as good or better. I appreciate your feedback!
I primarily use Mikroitk and some Fort for routing. I use Unifi, Cisco and some Mikrotik for switches. I do a great deal of Wifi so am good with WMM, PoE and the like. I use Unifi, Rukus and Aruba for APs. most of the systems will be a single WAN connection - sometimes dynamic sometimes not. I have just never done any SIP stuff I think I just need to get my hands dirty and figure it out. Thanks for all the help - you have given me some things to think about. I appreciate your insight.
You totally could figure out the sip stuff… it is not that complicated… Also it is very inexpensive to get a sip trunk to play with…
I use AnveoDirect.com for DID .
For 1 DID(a phone number that people can call to reach me) that can support 10 channels (10 diffrent people can call me at that same number at the same time and the calls will go through, if a 11th call is made into my system and all of the original 10 calls were still going on, the 11th caller will receive a “all circuits busy” error.)
Costs $1.00 per month + $0.004 per incoming minute… And to be clear that is not $1 per channel, that is $1 for all 10 channels.
I also use AnveoDirect.com for the call termination. (That is when a call originate from me going out into the public… For instance, if were to pickup my desk phone and call you… Prices per minute vary depending on the routes you choose for your outbound calls… For example, I can choose that my outbound calls be routed on the ATT network, but that route would cost more than if I choose Bandwidth.com to route it… depending on how you set it up it could cost $0.0001 to $0.01 per minute… just all depends.
So you could literally put $25 on a Anveo Direct account and that could take 1 year or more to spend that much if you are only setting it up for self training purposes and testing.
So as you see the, the incoming calls and the outgoing calls is 2 totally different things and can be configured totally separate from one another… You can have Anveo Direct handling the incoming calls… then have an entirely diffrent company handling the outgoing calls… Your PBX is the brain’s directing all the traffic.
There are tons of DID Sip trunk providers and outbound termination providers.