[HN Gopher] Building a Personal VoIP System
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Building a Personal VoIP System
Author : matthews2
Score : 277 points
Date : 2023-05-27 07:18 UTC (15 hours ago)
(HTM) web link (www.sacredheartsc.com)
(TXT) w3m dump (www.sacredheartsc.com)
| gamedna wrote:
| Having gone down this road many times with freepbx, asterisk,
| etc.. I ultimately settled on just using voip.ms and connecting
| phones/sip clients directly to their internal sub-accounts with
| voicemail. They have enough features for most users so you don't
| need to worry about running your own PBX.
| [deleted]
| jonpurdy wrote:
| I have been using voip.ms since 2015 for my phone service.
| Multiple numbers (DIDs) pointing to an IVR where callers must
| press 1 to connect to me (totally avoids robocalls). Plus
| voicemail (transcribed and emailed to me).
|
| One issue with voip on mobile (iOS in my case) is that I would
| often miss calls due to late push notifications and timing out.
| So recently I set up a calling queue that rings/pushes my phone
| a few times instead of just once (queue timeout to 30 seconds
| before hitting voicemail).
|
| Basically, if you want the control FreeSwitch and Asterisk
| offer but don't want to self-host, voip.ms is the best way to
| go.
| ivr-eric wrote:
| Same here. 3 businesses, only 1 problem in 10 years.
|
| The only thing I have to add: if you need to make telephone
| calls, the call quality using a SIP phone is much better than
| using a softphone.
| carlhjerpe wrote:
| This isn't necessarily true, a computer can speak SIP too,
| with a good headset it's just as good.
| stavros wrote:
| I disagree. Yes, in theory, an agent is an agent, but in
| practice I have never in my life at any point seen computer
| hardware that comes anywhere close to the usability of a
| Polycom device.
| zerox7felf wrote:
| Yeah, I would second this. In SIP a UA is a UA. So long as
| your softphone is good and your microphone is as well,
| there shouldn't be any difference. Although I would suspect
| the general experience may be that people with softphones
| more often will have terrible microphones for their PC...
| sydney6 wrote:
| Has someone good experience with a VoIP-Provider (not OVH or
| Sipgate) in Europe?
| folmar wrote:
| Nfon works fine but the panel is absymal at best.
| MattJ100 wrote:
| I'm eagerly awaiting jmp.chat to branch out into Europe (they
| have plans, but no concrete timeline).
|
| I'm curious why not sipgate, which I currently use, or Twilio
| which I half-use (it mostly forwarded to my own Asterisk
| server, which needs some attention).
| dominikmauritz wrote:
| Shameless plug. I'm co-founder and CEO of vio:networks -
| https://www.vionetworks.de
|
| We offer a virtual PBX based on Asterisk, Kamailio and
| Matrix/Element in Germany.
| progx wrote:
| Use fonial since 2 years without any problem. It has a good
| configuration options with a simple backend interface.
| linker3000 wrote:
| Do you have any specific issues with Sipgate? I only have a
| couple of numbers with them and don't use them much but am
| curious what their reputation is like.
| sydney6 wrote:
| I can't say anything about sipgate's reputation, as i haven't
| used their service. It just has been an issue with limited
| service availability in my country (BE)..
| gingerlime wrote:
| I just saw that they might discontinue their starter plan[0]
| as they focus on business. I can understand that residential
| voip is pretty much dead.
|
| [0] https://help.sipgate.de/hc/de/articles/4407798852113-sipg
| ate...
| zajio1am wrote:
| I have very good experience with Odorik (
| https://www.odorik.cz/?jazyk=en ).
| ttul wrote:
| If only there was an equivalent for building your own home
| GSM/LTE network. When around the house, I would love to connect
| to my own private cellular network. Not only is coverage poor
| where I live, but this would allow me to route calls in
| sophisticated and useful ways, because the backplane of modern
| cellular networks is VoIP.
| singpolyma3 wrote:
| I expect you would need a spectrum license for that :)
| zokier wrote:
| People still actually make phone calls? I mean its neat that you
| can run your own voip, but I struggle to imagine it getting used
| much or being worth even $5/month
| rjsw wrote:
| I'm just about to ask for my landline to be disconnected, no
| sense in paying for something I hardly ever use, won't bother
| porting the number to VoIP. Will either just keep the copper
| pair for VDSL or switch to FTTP.
|
| The only person that I did call using the landline is my
| father, but his VoIP system is unreliable.
| PopAlongKid wrote:
| When I'm conversing with family or friends by phone using my
| VoIP, and the connection starts getting bad, I always volunteer
| that the problem is not on my end, since my connection is
| wired.
| forgotmypw17 wrote:
| Yes, there are many people for whom voice phone is the
| preferred method of contact, and I accommodate them because
| they're important to me.
| hermannj314 wrote:
| People still read newspapers, ride horses, plant crops by hand,
| and dress up for renaissance faires.
| ocdtrekkie wrote:
| I want to set something like this up myself, probably less for
| external calling than as an in-home intercom. Public dialing
| might be handy just so it can easily include cell phones.
| rglullis wrote:
| So, maybe one of the VoIP experts that showed up on this thread
| can help me with one doubt...
|
| Let's say that I have a LDAP server where I manage user accounts,
| and I want people to be able to call each other with any SIP-
| enabled phone. I am not interested in voicemail/IVRs/any type of
| "voice application" on top of that. Do I really need
| Asterisk/FreeSWITCH or can I just go buy by setting up something
| like Kamailio?
| codeslinger wrote:
| Kamailio is what is called a Session Border Controller. Its
| primary purpose is to provide protection and some lightweight
| filtering for the media servers/PBXs behind it. Once you want
| "advanced" features like voicemail, parking, hunt groups,
| three-way calling, etc, you will need to use a PBX like
| FreeSWITCH (recommended) or Asterisk (not) behind it anyway. If
| you're only running a single machine, an SBC isn't really worth
| the trouble.
| rglullis wrote:
| > Once you want "advanced" features like voicemail, parking,
| hunt groups, three-way calling.
|
| Right, but what I am saying is that I _don 't_ want any of
| those features. At least not yet.
|
| I just want to give my customers a sip address ( _username_
| @communick.com) where they can call other SIP addresses. In
| that case, is that a PBX still needed?
| jjrh wrote:
| You certainly can use Kamailio but it will be much
| easier/faster to get something going in Asterisk.
|
| You don't need to run any of the advanced features in
| Asterisk (and can completely unload the modules if you
| want).
| michael_michael wrote:
| I've built my own business VoIP system -- a few times over. I've
| used Asterisk and FreePBX (the free-ish GUI built on Asterisk),
| but settled on Asterisk for stability. At one point FreePBX
| pushed out an update that took down my system for a couple of
| days and baffled me until I read the FreePBX forums and saw
| similar complaints from other users.
|
| If you want a week-ish long project, go for the full mid-2000s,
| boingboing and slashdot experience: buy the dead tree version of
| the O'Reilly Asterisk book, which has been kept up to date and is
| still an wonderful resource. Follow it until you have the
| Enterprise-grade phone system of your dreams.
|
| I even hired Allison "The Voice of Asterisk" [0] to do our phone
| tree voice prompts. Her prices were very reasonable for a small
| batch of prompts, and it's surreal hearing the same lady that
| does the IVR for your bank and cable company do your own voice
| prompts.
|
| I use Twilio for my SIP trunking, and it has nice fallback
| features in case the Asterisk system needs to go down for
| maintenance or the like. Costs about $20/month for a dozen or so
| users and fairly frequent calls/SMS.
|
| [0]: https://www.theivrvoice.com/
| TacticalCoder wrote:
| Ah good memories! Even though I knew nothing about VoIP I
| installed, years ago, one at my wife's little SME using "RasPBX"
| (a distro made of Raspbian + FreePBX + Asterisk) running on a...
| Raspberry Pi 1 (maybe a Pi 2 but I'm pretty sure it was a 1). The
| Pi was booting from the SD card but everything was running on an
| external HDD.
|
| I did put six Cisco VoIP phones and all was working fine as long
| as no more than four phones were used simultaneously (which never
| happened). It worked for years like that and wife ended up
| selling her SME with these VoIP phones still hooked to the RPi 1.
| We warned them that that thing was kinda a hack ; )
|
| For anyone hesitating: it's not hard to set up. Find a provider
| to get a SIP trunk, configure the thing, backup the config (I
| just imaged the entire drive), and you're good to go for a very
| long time.
|
| It's simple, reliable, stuff that usually won't move under your
| feet.
| revskill wrote:
| It's surprising that you could configure everything without
| hardware/software compatability issues.
| TacticalCoder wrote:
| Well I basically used a a Linux distribution made precisely
| for this: it was called "RasPBX" back then, IIRC. The
| hardware was simple: a Pi 1, an external HDD hooked through
| USB to the Pi and Cisco VoIP phones. That Linux distro
| already took care of picking software components/versions all
| working fine together and it was tailor made for the Pi. It
| was pretty plug and play from what I remember.
|
| I'm a software dev, not a sysadmin, but I can find my way
| around configure Linux machines.
|
| I don't remember it as being particularly complicated. What I
| do remember for sure though is that once it worked, it worked
| flawlessly for years.
| jamesmstone wrote:
| I have a tangentially related problem I have been battling with
| that you may be able to help me with. I'm moving overseas
| (Australia to Denmark) and would like to keep my current mobile
| phone number working for calls and SMS , but use it from afar.
| I would be able to leave a raspberry pi at a friend's house.
| Can you use this setup without a SIP provider? Would you need
| to buy a modem for the raspberry pi?what would you do?
| pseudostem wrote:
| I have setup a PRI to SIP channel before using elastix (now
| discontinued and bought by 3CX I think).
|
| This needs investigation, but - Over the top of my head, I
| think you could probably have some kind of a 4G/5G modem
| hooked on to an SBC (Pi, APU, etc.) and then forwarded to
| your SIP line which you can pick over IP (Internet), this
| should work.
|
| One of the child comments mentioned DID (Direct Inward
| Dialling). I am no expert, but if your Australian provider
| supports something similar, you will get the number (friend)
| who is dialling in, instead of your own Australian number
| dialling in to your receiver phone. This is useful for caller
| ID, otherwise it's just like a forwarded call.
|
| Hope this helps.
| singpolyma3 wrote:
| This is pretty much exactly the problem that we originally
| created https://jmp.chat to solve, no sip or complex setup
| needed these days for that use case.
| justsomehnguy wrote:
| I heard about you ages ago and wanted to use your service
| ever since. The only problem I don't need it for
| _anything_. But I 'm glad you you are still there.
| monkey26 wrote:
| I use Callcentric for this. Ported a land line over years
| ago. It now has SMS which I can do using the Callcentric app.
| And it forwards voicemails to me with email. Otherwise I
| don't use that line for actual calls anymore.
| sokoloff wrote:
| I've never done it, but look into chan_mobile and a bluetooth
| dongle to connect a mobile (with your SIM) to asterisk
| running on a Pi at your friend's place.
| sgc wrote:
| I would just port my number to twilio and use/write a really
| basic twilio mobile app. It's a bit risky to rely on a
| slightly cobbled hardware/software setup that is sitting
| across the world with no physical access.
| mindslight wrote:
| Note that one problem with voip providers is the
| numbers/lines often won't be recognized as valid by snake
| oil 2FA and the like. So porting to a voip provider only
| works if your goal is receiving SMS/calls from actual
| humans or just parking the number for later. But if you're
| doing it to avoid having to change your number for every
| service that has an SMS nagwall, it likely won't work.
|
| You can get a modem that will do SMS (eg a Sierra Wireless
| card), but I don't know if/how they do voice.
|
| Also, I'd shy away from the RPi based on the unreliability
| of SD cards. It would/will be pretty annoying for your host
| to go down and you have to travel back to fix it. I suppose
| if your friend is halfway handy you could keep an image of
| the machine as installed. Or even take new images remotely
| (rsync from the raw device a few times in a row?). If the
| root filesytem goes wonky, have your friend pull the SD
| card and reimage it.
| EVa5I7bHFq9mnYK wrote:
| I've been using a voip number for all my sms 2fa needs
| for the last 5 years while living abroad. I have a dozen
| banks and other financial institutions plus the irs
| happily accepting it. Some companies, like openai, wont
| accept it, for those I just buy single use SMS numbers.
| mindslight wrote:
| I've had problems with a bunch of places, so it's at
| least YMMV then. And I'd be wary of transferring an
| existing number in (as opposed to getting a new voip
| number and then setting it up).
|
| I'm sure specific services are hit or miss - there really
| _shouldn 't_ be a problem sending SMS to whatever number
| a user enters, but paternalistic snake oil salesmen gonna
| be paternalistic.
|
| I've had the least difficulty with Google Voice (I
| believe it's impossible to tell "Google Voice" usage from
| bona fide Fi usage where it's really the users' only
| phone number). Voip.ms has worked very little for me, to
| the point of that I don't even try it. Heck I even used a
| voip.ms number for some online classifieds, and another
| person using Comcast/Xfinity mobile couldn't text me. I'm
| not saying this as a slight against Voip.ms itself, from
| what I've gathered most voip providers will be treated
| similarly.
|
| One of these days when I get around to it I plan on
| setting up a wireless modem on a $3/mo paygo plan for my
| SMS nag needs. With a script that automatically brings
| the number as close to my paste buffer as possible.
|
| What single use SMS service do you use? I haven't really
| investigated those.
| Spooky23 wrote:
| Some services can definitely detect Google Voice. Chase
| bank won't send SMS to voice.
| singpolyma3 wrote:
| > I believe it's impossible to tell "Google Voice" usage
| from bona fide Fi usage where it's really the users' only
| phone number
|
| These are definitely distinguishable, and some auth
| service treat them differently. Google Voice numbers come
| from a different carrier (the one named Bandwidth) than
| most Fi numbers.
| InvaderFizz wrote:
| I use a RedPocket (GSMA Flavor) $60/year for my resume
| and 2FA number as an eSIM.
|
| I turn it on when needed, which is almost never. When not
| job searching, the voicemail plays the classic "this line
| has been disconnected" tones and message on repeat for
| two minutes. WiFi calling means I can use it anywhere I
| have WiFi.
|
| I get incredibly few spam calls using this strategy.
| rsync wrote:
| "One of these days when I get around to it I plan on
| setting up a wireless modem on a $3/mo paygo plan for my
| SMS nag needs. With a script that automatically brings
| the number as close to my paste buffer as possible."
|
| This is called a "2FA Mule":
|
| https://kozubik.com/items/2famule/
|
| "A 2FA Mule is a mobile phone configured to forward SMS
| 2FA codes via email."
| agwa wrote:
| There are two different reasons companies ask for your
| phone number, and it's worth distinguishing between them:
|
| The first is for sending a verification code during
| signup to prevent spammy/abusive signups. In my
| experience, this is the least likely to work with VoIP
| numbers because companies often intentionally block VoIP
| numbers.
|
| The second is for sending 2FA codes during login. My
| experience with this has been much better. AFAICT,
| companies do not intentionally block VoIP numbers for
| this use case. When SMS does fail, there is almost always
| an option to send the code by voice call, and this is
| always 100% reliable.
|
| I also have some experience with using a modem with a SIM
| card to receive SMS. I've used two different models of
| modem (not Sierra Wireless) and both have been very
| flaky, often locking up and requiring a power cycle, or
| having hours-long delays when receiving SMS. I would not
| call this approach a panacea.
| agwa wrote:
| Unfortunately, Twilio is no longer a viable option for
| sending SMS for personal use. If you want to send SMS you
| have to register a "campaign" and jump through a bunch of
| hoops that assume you're a company sending a large volume
| of application-generated messages (e.g. you must disclose
| samples of the types of messages you intend to send, and
| get explicit opt-in from recipients). Up until now, I've
| managed to avoid registering a campaign by instead paying a
| slightly higher per-message cost, but this is being phased
| out on July 5. I've been trying out https://jmp.chat (their
| founder is commenting elsewhere in this thread) and will
| likely port my number to them.
| rsync wrote:
| Hmmm ... I think the July 5 deadline is specifically for
| UK recipients ?
|
| Regardless, your overall impression is correct: Twilio is
| no longer a hacker/hobbyist/enthusiast option as you
| cannot (by the letter of the law, at least) send SMS
| without registering your business entity:
|
| https://twitter.com/rsyncnet/status/1593384850073214976?l
| ang...
|
| This is very troublesome to me because I have built my
| own personal telco out of twilio functions and twiml
| bins, etc., and am heavily reliant on all manner of SMS
| workflows.
|
| As of this writing (2023-05-27) everything - even SMS
| delivery to T-Mobile numbers - continues to "just work"
| but it sounds like I will just wake up one morning to
| have it all broken ...
| agwa wrote:
| The deadline is for US recipients:
|
| > * Effective July 5, 2023, all 10DLC phone numbers used
| to send SMS and MMS messages to U.S. phone numbers must
| be fully registered to an approved campaign under your
| brand. Messages sent using unregistered phone numbers
| will be subject to a gradual increase of message blocking
| by Twilio, beginning on July 5, 2023, ultimately leading
| to a full block of all unregistered U.S.-bound messages
| sent after August 31, 2023.
|
| Source: https://support.twilio.com/hc/en-
| us/articles/1260800720410-W...
| windexh8er wrote:
| I believe they're still related but the guys at Nerd Vittles
| [0] are still doing a PBX in a Box style deployment. Looks like
| it's called Incredible PBX [1] now. But the last time I ran it
| it took a lot of considerations around running SIP securely and
| brought a few of the pieces together. Probably worth a look if
| you're interested in self-hosting at home or for SMB.
|
| [0] https://nerdvittles.com/ [1]
| https://wiki.incrediblepbx.com/
| kunwon1 wrote:
| I've been working with bespoke VOIP/asterisk systems for a
| decade, I have one bit of advice to add - consider toll fraud.
| Especially for a system accessible from the internet. If an
| attacker can figure out how to make calls through your PBX,
| they can rack up tens of thousands of dollars in tolls over the
| course of a night.
|
| Most of the integrated FOSS solutions come with fail2ban
| already configured, it is essential. If you want more peace of
| mind, a prepaid trunk helps. That means you charge up your sip
| trunk account, and if someone drains it, it just stops working
| instead of continuing to drain your bank account.
|
| You can also limit international calling on your trunks, which
| effectively nullifies the financial drain of this kind of
| attack (though it's still obviously bad if an attacker can
| access your system in any way)
| pabs3 wrote:
| Does anyone have SIP setup on their domain? Are there many spam
| calls these days?
| deno wrote:
| Other way around, there are bots testing if they can use your
| systems for placing spam calls.
| linker3000 wrote:
| Yep, if I set 'Allow Incoming SIP Messages from SIP Proxy
| Only' to 'no' (allows direct SIP-SIP calls) on my home VoIP
| service, I'll get a silent call from '500' about every minute
| or so.
| Taniwha wrote:
| I've written about it elsewhere in this thread but I've found
| that a simple vopice menu that has a message (that might sound
| like a voicemail to a spammer's dial computer) and that
| requires people dialing in to press one key stops 99% of
| spammers
| singpolyma3 wrote:
| I run a public SIP server, and it gets constantly attacked just
| like any other public service. Mostly trying to use it to call
| expensive routes hoping it's unsecured though, not trying to
| spam local extensions.
| xnyanta wrote:
| The solution to the NAT issues is simply to use IPv6 and not
| worry about NAT.
| mnd999 wrote:
| Yes, that's what I do. Andrews and Arnold offer SIP over ipv6
| in the UK (and probably further afield).
| systems_glitch wrote:
| Yeah, it's amazing how many SIP providers don't support IPv6
| still :/ Total solution, no more headaches.
| deno wrote:
| Phones need to stay up and at this point in time NAT is more
| reliable than IPv6, which is probably not even an option most
| of the time anyway.
| fulafel wrote:
| Would be interesting to hear why they use v4+nat and face the
| described problem. First thought was maybe they bought some very
| old phones without v6 support - but the models he recommends
| don't seem to have this problem based on a googled datasheet.
| stonewall wrote:
| (Author here.) Sadly its a lot less interesting: my home ISP
| still doesn't support IPv6.
| jeroenhd wrote:
| For what it's worth, and if you're willing to tinker, you can
| get IPv6 for free through a tunnel as long as your router
| responds to ICMP: https://tunnelbroker.net/
|
| You can get a bunch of /64s and a /48 for free because HE
| really wants everyone to have IPv6 available already. Picking
| the right internet exchange to route from and making Netflix
| not throw a fit requires some minor experimentation but I've
| found it to work quite well.
|
| As an added bonus, because of the way IPv6 route
| advertisements work, you don't have to have a router with
| tunneling support. You can set up advertisements from any
| Raspberry Pi or other computer as long as it has outbound
| connectivity.
| remram wrote:
| > You can set up advertisements from any Raspberry Pi or
| other computer as long as it has outbound connectivity.
|
| You mean send the advertisement _and do the SIT tunneling_
| on that machine?
| jeroenhd wrote:
| Correct! It's relatively straightforward, actually:
| https://devzone.nordicsemi.com/nordic/nordic-
| blog/b/blog/pos...
|
| You use one of the /64 tunnels provided for you to route
| the /48 tunnel to the rest of your network. You advertise
| a subnet from your /48 to your local network and if
| you've got SLAAC enabled on your hosts that's all there
| is to it.
|
| You may need to mess with the default DNS server to get
| IPv6 results, though, that depends on whether or not your
| standard DNS server will respond to AAAA requests. It
| usually should, but some ISPs don't.
|
| This only works for a flat network, of course. If you've
| got different routers, you'll need to set up a more
| complicated setup.
| agwa wrote:
| The SIP provider would also need to support IPv6 for this
| to do any good. voip.ms does not: https://wiki.voip.ms/arti
| cle/FAQ#Do_you_Support_IPV6_with_SI...
| jeroenhd wrote:
| That's rather silly. Getting IPv6 connectivity is usually
| the difficult part, and servers are the easiest things to
| get IPv6 for. I wonder what part of their tech stack is
| still incompatible after all these years.
| agwa wrote:
| The quality of VoIP software tends to leave a lot to be
| desired, in my experience.
|
| None of the 4 VoIP providers I've worked with support
| IPv6 :-/
| da768 wrote:
| Last time I tried Tunnelbroker, it caused major performance
| issues. Not sure it's a good thing for VoIP calls.
| tempaccount1234 wrote:
| Any recomendable sip Clients for iPhones to Connect to such a
| system?
| supertrope wrote:
| Acrobits Groundwire. Counterpath's Bria Mobile. These use push
| notifications for incoming calls. That way there's no missed
| calls because the operating system killed the app or battery
| drain caused by keeping the app running.
| villgax wrote:
| I'd rather stab myself with WebRTC clients & a signalling+TURN
| server instead of enduring Asterisk
| mgbmtl wrote:
| Which clients do you use? And how do you connect to a SIP trunk
| / DID without Asterisk?
|
| I use WebRTC with Asterisk, and Browser Phone for the client
| (https://github.com/InnovateAsterisk/Browser-Phone). I don't
| use it much, but good enough for the rare times I have to use
| the phone.
| rcarmo wrote:
| This was sort of a thing in the mid-2000s, and I'm actually
| surprised to see it again. At the time I expected home routers to
| become SIP endpoints (and that came to pass with fiber), but they
| all expose FXO interfaces (jacks) rather than act as proxies for
| soft phones, so there's a missed opportunity there. But the truth
| of the matter is that just shipping a DECT phone in a bundle is
| much easier for the carrier to troubleshoot.
| forgotusername6 wrote:
| We could have easily been calling eachother by our email (SIP
| URIs) addresses. It is a shame that instead we got a dozen
| walled gardens when it comes to making voice/video calls over
| the internet.
| rsolva wrote:
| Can you share some resources explaining how this could be set
| up?
| gormandizer wrote:
| TRIP/ITAD is/was a framework that I believe was designed to
| facilitate this. Unfortunately it never took off.
| https://www.rfc-editor.org/rfc/rfc2871
| brazzledazzle wrote:
| One way was SIP and SRV DNS records. Seemed like it had
| some of the problems email has except it's a phone ringing
| instead of spam.
| orev wrote:
| Great explanation of the SIP protocol. I've been looking for
| something that explains it this clearly forever, and this gets
| right to the point with the high level concept instead of getting
| immediately bogged down in technical specs.
| systems_glitch wrote:
| It's worth noting that Asterisk will run on very low-end
| hardware: for the last 7 or so years, I've been running a small
| Asterisk box on a VIA C3 Mini-ITX system. System idle power is
| around 10W. The only reason I chose the VIA C3 system over
| something newer was that a) I already had it and b) it had a PCI
| slot (specifically PCI, not PCIe).
|
| The PCI slot let me plug in a Digium TDM800P and add eight POTS
| lines, either FSX or FXO, for pretty cheap.
| jjrh wrote:
| You will probably have a harder time finding hardware that
| won't run asterisk.
| WesolyKubeczek wrote:
| Today you could have the same average power consumption with an
| Odroid H3, and probably juggle more quality codecs too. (Well,
| sans the Digium card; but it's just mindboggling how far we
| went with power consumption these days.)
|
| I'm wondering how necessary POTS lines actually are these days
| (and how many connect to VoIP on the telco side). Should depend
| on the country.
| tomatocracy wrote:
| The H3 has an M.2 slot and a separate emmc slot. So if you
| really wanted to use that card perhaps you could use an M.2
| to PCIe connector and then connect that in turn to a PCIe to
| PCI adapter/riser. Power might be an issue though.
| systems_glitch wrote:
| The POTS lines were largely for goofing around, though I did
| use a FXO port to bring in the local POTS line. Mostly I used
| the FXS ports to interface a 1A2 KSU to Asterisk to run my
| old WE 2500 series key station.
|
| 90% of the traffic was handled over SIP or IAX to desk sets
| or ATAs.
| extinctpotato wrote:
| These days POTS lines are usually only used for last mile
| communications so the calls get converted to VoIP on the
| telco side. Basically it's for backwards compatibility -- the
| phone lines are already there, a lot of people have phone
| wiring in their houses and no configuration is required on
| the consumer's end.
|
| In general the days of having direct electrical connections
| between two distant telephones are long gone. The telco
| companies scrapped it when they realized that they could
| trunk the phone calls from a local branch to the central
| office using PCM streams over a single cable.
| systems_glitch wrote:
| Metallic path between two stations that weren't terminated
| in the same CO has been dead for a _long_ time! I suspect
| nowadays you 're unlikely to have metallic path outside the
| frame you land on, if that, unless you're paying for dry
| pairs.
| irusensei wrote:
| I've used to live in an inter generational big house (Italians am
| I right... even if we are not actually born there we still have
| spaghetti dna). Anyway I've setup one very lightweight asterix
| instance of statically defined accounts. A few cheap voip phones
| were installed through the house plus soft phones on personal
| smartphones.
|
| It was supposed to make things more convenient. Supposed to,
| because we just kept yelling to convey messages. When I've moved
| out I've dismantled the system since I wouldn't be there to keep
| it maintained.
| wnolens wrote:
| Cool to read someone indulge in this. For others interested with
| no prior context, also consider FreeSWITCH.
|
| Personal experience:
|
| I spent about 15 months working for a local telecom company,
| supporting their asterisk servers and developing interactive
| voice response applications. I was given the opportunity to build
| their residential voip service (in 2010?) from scratch and
| despite asterisk being so dominant I prototyped and eventually
| completed the project using freeSWITCH. I found it to be so much
| more developer friendly to configure and extend. Being able to
| build dial plans and implement logic with JavaScript or Lua
| rather than asterisk's config files was worth it. I suspect this
| system is still running.
| jelly wrote:
| I took a look at freeSWITCH's site and it looks like they got
| bought out. Their documentation is a total mess because the
| acquiring company has imported lots of wiki pages without much
| care for how they fit together.
| psim1 wrote:
| Their wiki has always been a bit of a mess. The company
| Signalwire is largely the same crew as the original
| FreeSWITCH team.
| dmpanch wrote:
| Asterisk from 12th version supports Asterisk REST Interface, no
| longer need to write configs.
| [deleted]
| danogentili wrote:
| Asterisk dialplans can also be written in Lua, thanks to the
| Lua extension!
| singpolyma3 wrote:
| Yes, this is what I do. The documentation is sometimes a
| trick to map over, but it's so nice to use a real language
| for everything else.
| password4321 wrote:
| Any technology recommendations or examples for integrating VoIP
| with open source chat platforms like Jitsi, Mumble, etc.?
| singpolyma3 wrote:
| Jitsi has something built in, I havent used it but I expect
| it's similar to big blue button which I have done, it runs a
| freepbx so setting up some extra dialplan rules to get
| interconnect is possible.
| psim1 wrote:
| I am one of the few people from my generation to maintain a "land
| line" (VoIP) and I, too, run it with Asterisk and the FreePBX
| configuration GUI. FreePBX provides a ton of macros out-of-the-
| box so that adding unwanted callers to a blacklist or doing phone
| number lookups is simple, for example. Why bother? Well, I find
| it interesting and fun, but most of the stuff I like can also be
| done with Google Voice. So I don't recommend self-hosting a phone
| system unless you are really into the idea and want to spend a
| decent amount of time learning telecom domain knowledge.
| derefr wrote:
| > SIP was initially released in 1999, and was designed with the
| assumption that each device has its own globally routable public
| IP address. After all, the IPv6 standard was released back in
| 1995, and NAT would soon be a thing of the past...right?
| Unforunately, this did not end up being the case.
|
| AFAIK, most residential _and_ commercial ISPs these days do
| assign customers both a dynamically-DHCP-leased IPv4 _address_ ,
| and a _static_ , globally-routable IPv6 _prefix_ -- usually a
| /64, though some are nicer than that. If you put your ISP's
| gateway router into bridge mode, and then plugged your computer
| directly into it -- then your device would acquire both an IPv4
| and an IPv6 address.
|
| But routers -- including ISP gateway routers -- insist on doing
| NAT not only for IPv4, but also for IPv6 (using the fe80::
| prefix.) So on any regular home or office network, devices are
| going to acquire private-use IPv4 _and_ IPv6 addresses.
|
| Is there some reason that modern routers don't do NAT for IPv4,
| while just further splitting+assigning the received prefix for
| IPv6, such that every device on the network receives a private
| IPv4 addr, but a _public_ IPv6 prefix, e.g. a /72?
|
| I know that Internet-backbone network switches ignore the last 64
| bits of IPv6 in their routing tables; but those bits are still
| being _carried_ in the IPv6 packets, and once they reach your
| home router, _it_ can make use of them to route to the final
| destination (i.e. one of the devices behind it.) Wasn 't this
| supposed to be the idea?
| t0mas88 wrote:
| My quite basic free router from my ISP does exactly that. It
| plugs into the UTP port on the fiber termination box (no idea
| how to call that) and handles DHCP for ipv4, while allowing
| ipv6 auto configuration using a /64 for all devices behind.
| Seems to work out of the box for both Windows and Apple
| devices. Only thing is that it automatically firewalls all
| incoming connections on both v4 and v6, but I think that's a
| very good default for an ISP device for home use. Especially
| since everyone is so used to v4 being NATed.
| kevincox wrote:
| A lot of people expect a stateful firewall blocking incoming
| connections on their local network. Applying the same NAT
| system that is used for IPv4 to IPv6 is probably the best way
| to get this layer of security.
|
| Now in theory most devices should be safe to connect to the
| internet. But how many Internet of Shit devices are there in
| the average household that probably shouldn't be trusted?
| Crappy security cameras with 10 year old firmware written by
| the lowest bidder as well as "smart" thermostats that probably
| aren't much better.
|
| So maybe keeping the stateful firewall by default is the best
| option.
| jeroenhd wrote:
| > A lot of people expect a stateful firewall blocking
| incoming connections on their local network.
|
| Totally! That's why that's the default setting for almost
| every router out there.
|
| > Applying the same NAT system that is used for IPv4 to IPv6
| is probably the best way to get this layer of security.
|
| No? The default firewall rules will work just fine.
|
| > So maybe keeping the stateful firewall by default is the
| best option.
|
| Agreed. That's why routers ship with fully-closed firewalls
| for both IPv4 and IPv6. Incoming connections need firewall
| exceptions, either manually or through UPnP depending on how
| you've configured your network.
|
| In fact, because of NAT issues like NAT slipstreaming, an
| IPv6 firewall is even more closed off than any IPv4 firewall
| that needs to let through FTP(S), SIP, and many other
| protocols depending on both sides of the connection using
| IPv4 as designed.
| derefr wrote:
| > Applying the same NAT system that is used for IPv4 to IPv6
| is probably the best way to get this layer of security.
|
| ...why? Routers have (stateful) firewalls, entirely separate
| from their NAT-ing abilities. You can still have the firewall
| without the NAT. (And the protocols that IoT devices rely on,
| like UPnP, are technically protocols for manipulating
| _firewall rules_ , not NAT port-forwarding rules; so they
| still work fine without NAT in place.)
|
| > But how many Internet of Shit devices are there in the
| average household that probably shouldn't be trusted?
|
| I'm not sure about IPv4, but in IPv6 a given single logical
| interface can acquire multiple IPv6 addresses -- meaning that
| your laptop or phone will have _both_ a public global IPv6
| address, _and_ a link-local fe80:: IPv6 address. And things
| like multicast, DHCP discovery, etc. will only be attempted
| or accepted through that link-local address.
|
| Now imagine a router that combines that firewall rule, with a
| bit of logic to automatically assign devices that join the
| AP, to separate VLANs, depending on their MAC address vendor
| part. So laptops and phones go on the "home" VLAN, while IoT
| devices go on the "sandboxed" VLAN. Where these VLANs are
| peered, but with stateful firewall rules between them:
| "sandboxed" devices don't get to speak to "home" devices,
| unless the "home" device speaks to them first; and multicast
| packets from "home" devices won't reach "sandboxed" devices.
|
| In other words, a home-network gateway-router should have all
| the same defaults that an IaaS-tenant VPC does: nodes on the
| network have public-routable IPs; but all inbound ports to
| them are closed unless a device asks; and traffic flowing
| between those devices can only be seen by other devices
| intentionally put into that same VLAN, not by other "tenants"
| who happen to be sharing the same pipes.
| mgbmtl wrote:
| > But routers -- including ISP gateway routers -- insist on
| doing NAT not only for IPv4, but also for IPv6 (using the
| fe80:: prefix.) So on any regular home or office network,
| devices are going to acquire private-use IPv4 and IPv6
| addresses.
|
| Maybe I'm misunderstanding your comment, but fe80:: is a link-
| local address and used by devices to talk to each other on the
| network. It's there by default when IPv6 is enabled.
|
| Most ISPs that support IPv6 will provide a /56 to the router,
| and then the router will assign a /64 to wifi. Then the clients
| get an IPv6 address using DHCPv6 or a route announcement.
|
| Unfortunately most VoIP providers do not support IPv6. voip.ms,
| which I use, for example, does not.
|
| In my Asterisk pjsip configuration, I use:
|
| external_media_address = dyn.example.org
| external_signaling_address = dyn.example.org
|
| where dyn.example.org is a dyndns that I use that points to my
| home Asterisk server, on a dynamic IPv4 address. My ISP does
| change my IPv4 address rather often, and sometimes I have to
| restart asterisk for the change to be effective.
| supertrope wrote:
| You can install Acrobits Groundwire or Bria. Those support PUSH
| notification for incoming calls. Push is better than missing
| calls because the app got killed, or forcing the app to run 24/7
| and severely shortening battery runtime. But the call quality
| will never be as good as the native phone app as that gets QCI
| prioritization.
| Taniwha wrote:
| I put in an asterix system 20 years ago, it still runs great, all
| I've done has been to replace a couple of dying (RAIDed) drives.
| I live in NZ used to live in the Bay Area, NZ trunks come in from
| a local SIP provider and use an Ooma box for our old US phone
| number, in home we had 5 FXSs and a sip phone in my office -
| everyone had a phone in their bedroom, and a couple on public
| spaces.
|
| It's set up so that all incoming calls hit a voice prompt saying
| which 1-digit extension you should enter to get who (this stops
| 99% of phone spam) - everyone gets a voicemail on their extension
| which rings in their bedroom and rolls over to the common spaces
| - and everyone gets a unique ring cadence, my daughter who at one
| point got 80% of the phone calls got the 2 short rings.
|
| The kids have gone, we've moved to a smaller house, only 2
| extensions, but that same hardware soldiers on (and still stops
| 99% of the spam - I've had 1 call in the past year)
| francescovv wrote:
| Excellent article, and sections "NAT Problems" and "NAT
| Solutions" are a good starter on that topic.
|
| Except even third-choice solution is not always feasible.
| Reserving fixed RTP/UDP port range is not possible with carrier-
| grade NAT, which is quite common with residential ISPs and
| nearly-universal with cell ISPs.
|
| Fourth-choice would be to reserve port range on a personal server
| (which would run B2BUA, asterisk in OP's case; or an RTP proxy),
| and force calls, including media, from/to SIP handsets to go via
| that.
| astrobe_ wrote:
| One often sees the STUN, TURN or ICE protocols around SIP-based
| VoIP, I believed they were supposed to help solve those issues?
| numpad0 wrote:
| They don't always work...
|
| The idea is if you send UDP packets to destination so
| arranged by middleman(STUN) or to a proxy so arranged by
| middleman(TURN) as an outgoing traffic, your Wi-Fi should be
| smart enough to set up a temporary NAT entry to allow
| responses to reach your $LOCAL_IP:$PORT. In reality, the Wi-
| Fi may have short memory or may be dying behind a
| refrigerator covered in dust and not able to handle all
| necessary combinations and ranges of addresses and ports,
| resulting in various partial failures such as one-way audio
| or missing participant in a group call.
|
| Fifth-choice option is to just encapsulate everything into a
| VPN, preferably L2 VPN over HTTPS to a server on a global IP.
| If it isn't working, there must be no Internet.
| remram wrote:
| Why would that be more reliable than TURN? If your router
| "forgets" about established streams half-way, your VPN will
| not stay connected either.
| astrobe_ wrote:
| UDP is unreliable transport _by specification_ , so I
| guess that if a network equipment such as a router cannot
| cope with the general workload, it would probably
| sacrifice UDP first without a second thought.
| remram wrote:
| If you don't have any evidence, guessing that
| routers/modems prioritize IP packets based on the next
| protocol sounds like a conspiracy theory.
| astrobe_ wrote:
| Huh? It's an obvious thing to do. If you have to drop a
| packet because your queues are full, any engineer with an
| IQ over 50 will pick the victim from the UDP packets,
| because the sender expects it might happen, and also
| because it won't necessary cause a retransmission - e.g.
| an RTP packet.
| numpad0 wrote:
| Makes it boolean. It's connected, or it's not. "One of
| RTP media transports to one of destinations is failing to
| establish DTLS ciphering and I think it has to do with
| either RTC issue or Chrome bug" is a self inflicted pain.
| deno wrote:
| Yes, Asterisk can poke holes in NAT on its own just fine. I
| was surprised how pessimistic the article is on this. I have
| systems running for months and years behind NAT with no
| issue. You might have to disable direct media
| (endpoint/disable_direct_media_on_nat).
|
| Also, this is just uptime related tip not NAT, you must
| explicitly set registration/max_retries to a huge number
| otherwise Asterisk just gives up permanently at some point.
| It's a really weird default.
| singpolyma3 wrote:
| Are you doing calls to/from other sip URIs that are also
| behind NAT, or just using your trunk and internal
| extensions?
| deno wrote:
| Trunk and internal, and I usually put all the phones in
| their own VLAN w/o direct Internet access. I don't really
| see a use for dialing arbitrary SIP URIs. If I need to
| add a remote phone I'll just connect it directly with a
| network tunnel.
| jeroenhd wrote:
| All of the NAT problems would instantly to away with IPv6, but
| with adoption still at a meager 50% I suppose you'll need a PBX
| of some kind to receive at least half the calls.
|
| For those stuck behind CGNAT, there are guides online for how
| to set up a VPN to a cheap VPS and forward all network traffic
| to your network so you can have almost-real connectivity at
| home. If you're content with 50mbps, you can even use Oracle's
| Always Free tier.
| z3t4 wrote:
| VoIP used to be standard on phones, even mobile smartphones. So I
| setup Asterisk so that family members could call each others as
| long as they where on WIfi. Unfortunately VoIP is no longer a
| built in standard so you need to download an app to use it.
| Before you could just dail 1 and my phone would ring, dail 2 and
| you would reach my wife.
| lormayna wrote:
| One of my first job, in 201, was to create a spam faxing machine
| based on Asterisk. Once the initial setup was completed (T38 is a
| bit tricky to tune), it was very effective, it sent more than one
| milion of faxes every year.
| forgotmypw17 wrote:
| Thank you for sharing your experience. How do you feel about
| it?
| lormayna wrote:
| I am not the only sharing his experience with Asterisk in
| this thread. Why this criticism?
| forgotmypw17 wrote:
| I commented because you reminded me of a similar experience
| I had.
| cyberax wrote:
| I have a VoIP system at home as well. I first used an RPi with
| Asterisk, but later switched to a Yeastar box with FXS ports (to
| connect a couple of Old School wired phones).
|
| One thing that is making me REALLY MAD is that there are NO IPV6
| TRUNK PROVIDERS in the US. Not a single one. At least none where
| I can just enter my credit card and get a phone line.
|
| Somehow, the protocol designed to restore the end-to-end
| connectivity is not used for the poster child of end-to-end
| connectivity.
| hashstring wrote:
| > Use a SIP Application Layer Gateway. This is a horrible feature
| offered by some routers. Basically, it deep-packet-inspects your
| SIP traffic, rewrites the headers, and creates port forwards on-
| the-fly to make sure the inbound audio stream makes its way to
| your device. SIP ALGs are a total hack and notoriously buggy.
|
| Yes, these hacky ALG features also allowed internet users to
| access internal IPs on arbitrary services (!); named "remote
| arbitrary firewall pinhole control". The attack was published in
| 2020 and named NAT slipstreaming [1].
|
| [1] https://samy.pl/slipstream/
| jasonjayr wrote:
| This was on by default on a consumer router that was used at a
| small office. Their VoIP phones would get phantom calls until I
| discovered this and disabled it.
| hashstring wrote:
| Woah, good find. How did you actually find out that this was
| happening? Network captures?
|
| When was this approximately? I'm wondering how widespread
| this (still) is.
| jasonjayr wrote:
| It would have been 2-3 years ago. I was getting reports of
| phones ringing, but no audio, no calls record from the VoIP
| server CDR (which all phones were configured to proxy
| through) and when looking @ the system logs from the
| phones, they were reporting nonsensical IP addresses, which
| pointed to something at the edge of the network.
| jeroenhd wrote:
| I'm pretty sure you've solved a mystery I was confused by
| years ago, back when I was doing tech support.
|
| It wouldn't have mattered much because the router/modem
| combos in use didn't have a switch for SIP ALG anyway, bit
| it's good router finally know what could've caused the
| phantom phone ringing.
| agwa wrote:
| Nice article. Is there a benefit to using a queue for incoming
| calls instead of just dialing multiple extensions with the &
| operator?
| deno wrote:
| You get some statistics for queues but also for example you
| might want to only take one call at a time even if you can
| answer from multiple. Also Asterisk queues make it very easy to
| do things like 'there's N callers waiting' etc.
| kamma4434 wrote:
| A queue shines when it can distribute incoming calls based on
| it knowing agent availability. (Shameless plug: the company I
| work for does cloud reporting [1] for Asterisk and FreeSwitch
| queues - that is then whitelabelled and sold by a lot of big
| name telcos worldwide. Not sexy but effective!)
|
| [1] https://www.queuemetrics.com
| aftbit wrote:
| I have been tinkering with a personal VoIP system in my spare
| time over the last couple of months. At this point, I have
| rescued the 3 lines of house wiring in my 1970s house and
| connected them to Asterisk on a VM via a Cisco MC3810 and a Adit
| 600 channel bank. This has involved messing with T1s which was a
| childhood dream of mine. I have 12 phones connected to the
| system, using my house wiring, a few point-to-point wires, and
| some SIP VoIP phones over ethernet. In turn these connect to
| Phreaknet, C*NET, and of course the normal PSTN via a pair of
| different ITSPs. I now feel almost qualified to build an early
| 2000s business voice phone system, for what very little that is
| worth. More importantly, I have a rotary phone on my desk. ;)
| sylware wrote:
| There is SIP and XMPP, but they may be overconvoluted for what I
| am trying to achieve. Namely, I am not aiming at internet
| universality (no IPv4 sharing abomination built into the
| protocols).
|
| I am looking at a modular set of protocols built mainly for IPv6.
|
| The base: the "telephone number" would be ipv6:port. "Ringing"
| and video/audio streams setup would be done here. End to end
| encrytion right from the start (only manual key exchange, zero
| automatic, even before ringing).
|
| On top, a "comfort protocol"(one level of indirection) for those
| changing ipv6, but not "accutely roaming", namely changing ipv6
| while in a video/audio call: a "DynDNS" but simpler, more a
| "current IPv6:port of 'name' kind of thing", "address book with a
| drop of dynamic", "name@server" and you get the current
| ipv6:port. Unfortunately, it means "accounts" and real time
| updates. Of course, "server" could be a local/dns/ipv6. I am
| thinking zero password, only a public key.
|
| For video/audio streams, I may not bother and go TCP. The main
| constraint would be the timing information shared among
| video/audio streams. No "internet weather" dynamic
| reconfiguration.
|
| I even consider going "horribly horrible" for internet: idiotic
| binary based instead of text based protocols.
| KaiserPro wrote:
| my wife had a requirement: be able to have an intercom for the
| house and shed. We used to have a baby monitor, but that was one
| way, for one room. Now we need many more rooms to talk to each
| other.
|
| I looked at some intercoms on amazon/ebay, they are all RF and a
| bit shit. I saw some wifi ones, but nothing cheap enough to take
| a punt on. I did think about trying to make something with an
| ESP32, but that would be too hard for me in the time.
|
| So I bought 6 cisco 7962 sip phones for PS35 in total, and
| installed freepbx.
|
| It took a bit of effort to bring the tftp server online, and make
| sure all the dhcp info was being passed on correctly. Once that
| was complete, freepbx makes most things pretty simple.
|
| Now, I have ethernet is most rooms, and a switch that can do PoE,
| so this solution is for a niche of a niche
| numpad0 wrote:
| I actually run 7962g + FreePBX too, using sccp_manager and
| phone-integrated SSL-VPN. It took more hair pulling and shady
| patching than the SIP route, but now I can make intercom calls
| over the Internet! That way I don't need a direct L1/L2 link
| between "offices". Solves NAT problems too, because it's all
| in-band signaling and media transport over single TCP
| connection.
| nicolaslem wrote:
| I remember playing with an old device as a child at my
| grandparents' place, it was similar to a walkie-talkie but
| transmitting using the mains power instead of RF. It was
| supposed to plugged to a socket and hanged on the wall. It was
| made exactly for the use case you describe, as an intercom
| between rooms. It even had a button to ring a bell in another
| room.
| hiatus wrote:
| These still exist, you can buy them on Amazon. [1] I figured
| these had to exist as I recently saw a device to make a lan
| using mains circuits.
|
| [1]: https://a.co/d/31rKR02
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