Sangoma's doing a good job of consolidating the SMB Telephony market. They bought Schmooze Com Inc.[1] (owners of FreePBX, the largest Asterisk based Linux distro), Dialogic's hardware business, and VoIP Supply[3] (where most independent VARs order phones from).
Indeed, I ran a large-ish asterisk user group for years in Toronto and know a number of people at Digium and Sangoma, and there was always a bit of animosity between them. Glad to see them finally coming together.
Most of the talent I know took a step to the right (like myself) and are working with WebRTC. It's all the same stuff! Just more signalling layers on top of RTP channel management. I quite often end up doing SIP over WebRTC with things like sip.js (still backing onto Asterisk and FreeSWITCH) and XMPP over WebRTC with jitsi-meet.
I fear more 3CX as a monopoly, the bought elastix and ditched it.
3CX CEO bought lots of SANGOMA shares back 1n 2016 (and later sold a few of them in 2017)
Why your competitor's CEO would buy shares of your company?
tips tinfoil hat
As someone who previously worked for a Dialogic customer with a single-digit customer number, I didn’t realize Dialogic had sold its hardware business off (again; after the IBM acquisition and subsequent spin-off). About time.
That said, I never really saw Dialogic as in the SMB space, certainly not their hardware division.
We must have had over 300 D480SC-2T1 boards in service at peak. Absolute workhorses. Their cPCI offerings were never as stable but 16 T1s or 1000 SIP ports in a single slot was extremely attractive. I recall them having issues with their supply chain for the high-density SIP boards in conjunction with field failures that pushed us towards their software product, HMP.
Dialogic’s HMP product was more stable and ultimately cheaper at scale (for SIP anyway; never had good luck with their T1 interface boards on HMP). Funny they didn’t sell that division off. Perhaps it’s still profitable or maybe Sangoma was simply uninterested as it isn’t really aligned with the Asterisk/FreeSWITCH ecosystem.
Wonder how much of that stuff is still running. You left a year after me so I suppose you wouldn’t know ;)
I have little faith in Sangoma's leadership when it comes to open source, it's not in their nature. Products and marketing is not the same as software. This is an old strange beast, and it's being put to sleep.
Digium used to host the local Linux Users of North Alabama meetings back in 1999 when they were Linux Support Services. I remember being blown away seeing Asterisk running on commodity hardware. They had all kinds of cool hardware hacks in their office. Kudos to them for revolutionizing the PBX space.
absolutely not true!! and other competitors like 3cx have been eating their lunch for years. the fact of the matter is FreePBX, even for the most knowledgeable audience is still a PITA to work with and configure.
> FreePBX, even for the most knowledgeable audience is still a PITA to work with and configure.
Agree to disagree. I cut my teeth on freepbx; love it, miss it, constantly yell at my customers digium appliances in disappointment that they don't live up to their sibling. I rolled my old MSP over to it without any hassle and afaik they're still running it a few years later and I know none of them have touched it for admin. Rolled it on my VPS for my own consulting needs and it's pain free.
Digium was mostly a hardware company for a long time, which doesn't fetch the valuation of recurring SaaS revenue though they have hosted SaaS services now.
But they were bleeding money, and just sold $30M total last year. Lots of people use Asterisk (it's basically everywhere - whatever you call - dentist, doctor, plumber... - they have the default Asterisk music-on-hold), but not all of them put a coin in the box. :)
They're everywhere because they're already there, but doesn't mean new boxes are being installed.
Personally I wouldn't use it; I'd just write some quick code with Twilio (using Twilio functions or a companion app running on Heroku for the business logic) and let them handle it. Less maintenance overhead.
I'm not really old enough to have experienced Centrex, but for small businesses its promises makes sense from a business point of view. Less maintenance/support overhead is a huge plus.
sounds good. until you run conferencing services or whatever. Then the pricing is pretty simple: buying wholesale SIP and operating on your own is substantially cheaper.
irl maintenance is pretty straightforward. I do it as a side thing to my main job.
Somewhat off-topic, but what wholesale SIP provider would you recommend? Looking for something small-scale for now, pay as you go as opposed to $X,000/month.
Twilio, voip.ms, or DIDlogic. Twilio may be for large users, but it works great for tiny ones as well. No minimums, simple wholesale pricing, and they still have free support even if you don't want to spend thousands a month for 24/7/365 phone support with an expert.
The state you tend to store is in the application side, so it still backs onto things like SQL databases in the end. You'd put another service in front of your Asterisk instances like OpenSIPS or Kamailio to distribute load.
Seems like a huge hack. What we really need is some kind of etcd for telecoms, where you just configure it and they replicate and share state automatically.
It's not a hack. Think of Asterisk as if it was an http server + backend code. How do you scale?
You scale horizontally using proxies (kamailio dispatchers), (what you would call "reverse proxies in the http world), and use external storage / database / logic.
Asterisk is just another daemon. You could use etcd for it or Puppet or whatever to create config on the fly. On FreePBX you generally use MySQL/MariaDB.
Lots of ways. Usually externally, through a database. An Asterisk box processes and forwards calls - you can hook into a "master controller" that decides what is to be done.
And they also have a very decent line or hardware. Digium cards used to be the de-facto standard to build quickly a small PBX system, and I really appreciated their quality.
Interesting. I built some VOIP systems on Asterisk about 12 or 13 years ago. It saved a few small companies a ton of money compared to an older proprietary system.
I remember the FreeSWITCH from 10 years ago. They didn't even cut releases, you were just expected to always run at HEAD - and the whole community refused to talk to you until you showed your commit hash. I got cores constantly from all the null derefs littered throughout the code. And at this point they considered themselves a stable, production-ready project.
I tried _hard_ to make FreeSWITCH work for me in an SMB environment, but I ended up with such a bad taste from the poor QA that I eventually had to call it all a false start and go back to Asterisk.
I worked at Digium from 2008-2011. Digium employed some really great programmers and I obtained a solid starting foundation for my career as a developer hacking on Asterisk in C. Definitely enjoyed my time there.
Going back to around that time period, there was a project started called Asterisk SCF (internally called Hydra, I couldn't remember if it was made public but it's in the list archives: http://lists.digium.com/pipermail/asterisk-scf-dev/). Asterisk SCF was going to be new software (written in C++) that supported scalability as a primary feature. I still wonder if Digium had continued funding it what would have happened, but eventually development was halted: http://www.digium.com/blog/2012/09/14/asterisk-scf-pause/.
All that to say I'm happy for Digium's success, but still think an acquisition could have occurred far sooner with a product based on Asterisk SCF.
wow, pretty amazing they started kind of working on top of Asterisk, with some of their own hardware, building out their own fork of it/pbx software...and now this? Impressed. However I'm curious about Asterisk growth/Digium as it seems like VOIP service from major telcos, in North America anyways, has caught up to what was once a very viable open source alternative. I built asterisk systems for our own in-house multilocation system for years because it was cheaper and offered infinite more control and options. But we phased it out a few years ago because big telco had caught up and offered an easily managed system with alot of the same features, it had just taken them 5-7 years to catch up. The flexibility isn't on par but it's good enough.
> wow, pretty amazing they started kind of working on top of Asterisk, with some of their own hardware, building out their own fork of it/pbx software...and now this?
they were selling T1/T3 hardware for years before asterisk was a thing. does seem like they pivoted to the voice stuff after asterisk, though.
This is their business model that leads them to where they are now. Open source market is very tricky, you work for others and if you don't know how to earn money, you are done.
FreePBX (built on Asterisk, which is the basis for another dozen PBX platforms) is still far-and-away the most popular PBX platform for small businesses and most VARs. FreeSWITCH does exist and it is fantastic, but the two are quite similar in terms of what they can accomplish.
There is really no other option in the FOSS community for small-to-medium phone systems, as Asterisk-based systems are rivaled mainly by 3CX (proprietary) in the industry which is somewhat expensive and significantly less-configurable than the open source options.
Mildly hijacking this thread... A few months back I investigated SIP as a replacement for skype, discord, whatsapp etc. It seemed a very insecure protocol: If I have an account on server A, and someone else has an account on server B then a user on server B can call me but neither my client nor my server will verify with server B that the caller is who they say they are.
The email ecosystem had the same problem where anyone could send an email from any address but managed to solve it by adding new standards on top of old ones to verify senders. I hope something similar can happen for SIP.
If you are talking about SPF and DKIM, none of them verify the sender. The former indicates which IP adresses are allowed to send a email from for a specific domain name. The latter lets you verify that the email originated from the domain. But not from the sender itself.
If you were talking about PGP signatures, ignore my previous words :)
The issue you see in SIP predates voice over IP. PSTN suffers from the very same issue.
Well if you get an email from xxx@somemail.com and the SPF and DKIM check out then it means that the mail really came from somemail and they have had the chance to verify that xxx is authorized to send the email with e.g. a password. The system is not 100% foolproof but it's good enough when working with reputable or selfhosted email services. It's way, way better than "anyone can trivially pretend to be anyone"
You can use many of the same techniques used to drop email spam, for example forward and reverse DNS lookups and of course your favourite next gen firewall can do all sorts of fancy things to help decide "trustworthyness". You can use authenticated trunks between A and B. You can allow anon connections but pre-screen calls by asking the source to identify themselves and play that to the recipient who can choose to accept or drop the call.
You can do an awful lot of things with SIP that are unthinkable or plain unlikely with PSTN. SIP can be encrypted (OK the RTP streams can) You can use IAX2 for those times when NAT and SIPnRTP are too hard (hint: try Symmetric RTP - fixes many NAT related problems)
Don't confuse SIP with something it isn't! SIP is a comms mechanism and a damn good one, considering how old it is. When you deploy SIP, you also have the option of using all of the very latest funky security stuff around it to support it thanks to the fact that it runs over UDP/IP or TCP/IP.
The email ecosystem had the same problem where anyone could send an email from any address - I'm not sure that you can call this a problem. I could call myself Mr Donald Trump, in fact my real name could even be Mr Donald Trump but it wont really make me POTUS but you need some way to tell the difference. Comms is tricky and safe comms in the modern world is very tricky. How far should a comms protocol go in ensuring that the source is who the recipient thinks it is? Or is that really a job for another protocol/system perhaps with some hooks of some sort?
[1] https://www.freepbx.org/sangoma-completes-the-acquisition-of... [2] https://www.dialogic.com/2018-01-09-sangoma-announces-sixth-... [3] http://www.marketwired.com/press-release/sangoma-announces-t...