Bria is the defacto commercial softphone. It isnt free, but it is pretty much the standard from a commercial perspective.
There are lots of various SIP softphones (search for that term).
Zoiper https://www.zoiper.com/Works for Windows/Linux/Mac/Android/IOSYou can pay for commercial license that has Video Calling, Outlook Integration and Click to DialAlso supports TLS/SRTP/ZRTP for encrypted calling
Jitsi is fully open source and works really well.
They've got a section on zrtp and srtp and support secure SIP (if your provider does) if that's a motivator.
Signal does a pretty good job, Linphone for Android & iOS can actually be pretty decent too, it has near feature parity with Signal. WiFi to LTE handoff (and vice versa), Opus, SRTP and ZRTP support are all there and work fairly well.
Heads up, CSIPSimple hasn't seen any updates since 2014, and Zoiper isn't terribly great from my testing. Both work, but still not great SIP clients.
I use Logitech H800 when working remotely. It's wireless, has option for bluetooth or USB receiver. It's only downside is talk time is short (i keep it charged when not in use). Otherwise we use the Plantronics Blackwire 3220 which is wired USB that works really well too.
> GXW4104
Have you used a PCI type card before? Is it any better than getting a seperate gateway? Which would you advise if I was intending to install a FreePBX system anyway on my server PC? I would assume I need at least 2FXO ports. What is a FXS port on the card for? To connect a regular line out just in case..? All the Digium cards I see on the regular USA amazon website is like 500+! I can't find the model you quoted on the USA website.. (https://www.amazon.com/DIGIUM-1A4A04F-Digium-Voice-Plug/dp/B01LXAJ4FH/ref=sr_1_5?ie=UTF8&qid=1481556629&sr=8-5&keywords=digium+fxo+card)
For the FXO gateway, I did take a look at the Grandstream but as the previous poster mentioned, would it be hard to configure with the FreePBX I would install on my server? Does the Obi110, Obi202 or Cisco SPA3102 do the same thing?
Oh I guess this works? https://www.amazon.com/Analog-Express-Connector-Elastix-Freepbx/dp/B01KK0B1DE/ref=pd_sbs_229_4?_encoding=UTF8&psc=1&refRID=QFQ1GC50ZJ9MSNPBV3GQ
Not really able to help on your question, but why not go with something like this:
https://www.amazon.com/Cisco-SPA112-Port-Phone-Adapter/dp/B00684PN54
with a sip carrier or:
https://www.amazon.com/dp/B00BUV7C9A
if you really want google voice. Not sure that the second one is still supported.
For voice calls on VoIP.ms, you don't need a mobile app. Android has native SIP functionality. The base OS supports SIP calling. Here are the configuration instructions for VoIP.ms on Android. Your phone calls, both incoming and outgoing, use the same calling interface that all other Android cell phones use.
For SMS on VoIP.ms, there is an app. It requires some configuration to use. The app walks you through it when you start it up.
There is no way to get MMS on a VoIP.ms number. There is also no way to pay less than $25 since $25 is the minimum billing increment. That said, a single $25 payment should be able to last for years if you're a light user. Your credits never expire.
There is nothing else out there that will be cheaper than VoIP.ms in terms of both minimum payment and ongoing cost. Fongo doesn't require a $25 minimum payment, but the ongoing cost is $4.95 per month which is way more than VoIP.ms. Fongo's sister company freephoneline.ca has plans with zero monthly ongoing cost, but there is a $79 one-time fee.
https://play.google.com/store/apps/details?id=net.kourlas.voipms_sms works well for me
In your VoIP.ms account settings you have to enable the API and generate a key and set the IP address restriction to 0.0.0.0 for this app to work, but it does and it works well.
Sounds like a job for AutoHotkey. You can launch sound settings with:
Rundll32 Shell32.dll,Control_RunDLL Mmsys.cpl,,0
Then just automate that screen. Check the docs for commands like ControlClick to click buttons based on their text value.
Sources: https://www.thewindowsclub.com/rundll32-shortcut-commands-windows https://autohotkey.com/docs/commands/ControlClick.htm
Something like this
http://download.cnet.com/Virtual-Audio-Cable/3000-2168_4-10067766.html
Install a softphone client and map the audio input and output between Skype and the client software. Then set up the client to auto answer. Then you can dial out of the soft phone or into the soft phone and it will work with Skype.
Alternatively, use Google hangouts instead of Skype. You can dial any phone number from a hangout, so if you want to have some people on PC and some on handsets it's easy.
It's called Cloudflare Magic Transit where they advertise the address space, and do some DDOS protection at the edge.
Cloudflare doesn't do SIP natively, and it's unlikely they ever will, as that would mean that the provider needs to upload all of their valid credentials into cloudflare.
This is going to get a bit complicated if you don't want to do it on standalone hardware but still doable.
As others have said, FreePBX, Freeswitch/FusionPBX or practically any "PBX" will work for you. I heartily recommend FreePBX though, beyond the initial set up it's considerably simpler to use for what you need it to do and free.
You could also consider a piece of software called 3CX which is a native windows PBX. Very easy to set up and use but it's not free and gets expensive really quickly for an individual. Also the company itself is an utter scumbag if you care about that kind of thing.
Now if you really want to avoid using dedicated hardware for the PBX, you can setup the PBX as a "virtual machine" using software like VirtualBox that will let you run it as a virtual computer. Your computer will need enough resources to power it, but a PBX generally doesn't need a massive amount depending on the usage.
Best practise though would be to put it on it's own hardware. It's a lot more flexible than having to boot it with your computer. If you have an old computer kicking about the house, then that would be sufficient for the most part as it's a good way of repurposing old equipment.
why stereo?
dispatch means the audio is being sent out of a group of people? like a wrecker service? how's the audio being received on those devices?
i think if you're more specific, you may get better responses.
Personally, I would suggest you check out voxer for mobile smart phones http://voxer.com/
Issue is that this would require spawing a new process, since the phone isn't in a web browser.
That being said: Zoiper has a web phone, but the only reference I see is that it will open a URL "when one is received from the server", which is obviously not the goal here. LOL
I checked X-lite, Zoiper, and 3CX (all of which we use regularly), and did not get a hit.
However: LinPhone is open source, Functionality could be added, obviously. And a user in this thread had a "close hit" requiring a button press: http://serverfault.com/questions/718921/i-need-a-sip-softphone-that-would-on-incoming-call-open-an-url-containing-the-ca
Alternately, Vicidial's PBX has "start call url" feature that runs server-side and of course as many other have mentioned this functionality can be added to customize asterisk.
As a final "brainstorm" option:
1) pass your call through a Vicidial server 2) In the Vicidial DID settings, you can set the "filter" to pull a URL to determine the path of the call without answering. 3) Regardless of outcome, pass the call to your soft phone and the URL was pulled and the phone is ringing.
If you want to have the URL popped up on your workstation automatically, that's what Vicidial calls a "web form" which is one of several options that can "pop up" automatically at the start of the call. It does require, of course, that you are using the Vicidial Agent Interface. I expect that the Vicidial Agent Interface will soon have a built in soft phone (making it a phone that meets your description). Presently, however, if you purchase a Zoiper Web Phone it can do this Now (the expected future web phone will be built into Vicidial .. which is free) in Vicidial. Zoiper is the client, which runs in the Vicidial agent screen in an iframe and pulls the URL when Vicidial detects the inbound call is live.
Elastix was a GREAT distro, but part of the reason was they stayed with older, more reliable versions of FreePBX and Asterisk. Unfortunately, they just recently switched from FreePBX to 3CX so the old version isn't available from their site anymore. I heard there is a fork of Elastix 2.5, it may be the one at https://blog.sinologic.net/2016-12/openelx-el-fork-de-elastix-para-mantener-la-version-2-5.html but I do not read Spanish so I am not sure. This may or may not be an English version: http://www.issabel.org/
I haven't been following the ins and outs of Elastix but I have nothing bad to say about it other than it was getting a little long in the tooth. But it worked a lot better than the stuff that's out there now, and the only reason I might think twice about recommending it now is that it may contain security vulnerabilities that were never fixed in those older versions.
The challenge with troubleshooting hosted providers is that it's tough to discern between a problem with your ISP's last mile vs their upstream connections vs a problem with your hosted VOIP provider. I like MTR for showing response times and paths, be warned that if your poor audio quality is due to saturation then MTR will just put extra traffic over the wire though.
If you're looking for other services...
Disclaimer - I work for Star2Star
Friendly offer, no more no less - We're headquartered in Sarasota, FL but have partners all over the country. We don't do x dollars per phone per month as our normal pricing model, rather most customers lease or purchase our equipment outright and pay for PSTN service and support. We run a lightweight Linux appliance on site that acts as a gateway to our cloud services, and all config changes are done through a web portal. We use our box on site to monitor the connections we're running on as well as the LAN.
If you'd like to chat with one of our partners in your neck of the woods to get some info I'm glad to introduce you, no pressure though. Best of luck!
I'm not sure how DIDWW work - Is each DID a SIP account which registers as a SIP peer from Asterisk -> DIDWW's Call routing servers?
If each number does register to the server from the carrier over its own account you could use munin (http://munin-monitoring.org/) for monitoring - Theres plenty of plugins out there for Asterisk which will monitor SIP Peers every 5 minutes and report back how many are online. You'd just need to modify the script so that it only reports back on the SIP accounts which are DID's. Eg: http://exchange.munin-monitoring.org/plugins/asterisk_sippeers/details
This is what we load onto all of our customers Asterisk PBX's systems - Works perfect for us. You can even setup email alerts.
Or you could just run some basic command line stuff to check the log files for outages:
Check the SIP accounts and manually look for offline accounts: asterisk -rx "sip show peers" | less
Or IAX2: asterisk -rx "iax2 show peers" | less
Or look through the logs for peers going offline: egrep -r "Reachable|UNREACHABLE" /var/log/asterisk/messages*
Still be less tedious and alot less time consuming than continuously ringing the numbers.
For an enterprise that size you're going to want some redundancy. Look into sipXcom for telephony services and a Sangoma session border controller (appliance or virtual) for inbound/outbound call routing to multiple VoIP carriers. Combine all of that with some Polycom VVX phones and you've got a recipe for awesomeness.
eZuce Uniteme is the commercial version of sipXcom and offers more enterprise features that the open source version. PM me if you want to know more, I can connect you with someone who can get you on the right path.
(Disclaimer: I don't work for eZuce but I used to, still have contacts there.)
>My boss and I are running all apple products.
You will be making life much harder for your self if you insist on running a software PBX on apple hardware. I'm not saying you cant, but pretty much everything is designed with Linux as the main target.
>We want an inbound 800# with a Los Angeles. CA tag on it (we're not in CA)
Not sure what you mean, since a toll free number has no localized NPA on it (area code). Maybe you can clarify what you mean by "tag". I use Flowroute as a carrier for my personal phone system. TF will run you $1.25/mo per DID + minutes.
>We want all incoming calls routed to a voice message that lets you pick either myself or my boss. All VMs are to be turned into text and sent to our respective email.
IVR with a vm failover that sends email with recorded atachment - all pretty generic requirements.
>We also want built in capability of faxing and such.
Faxing is best effort over VoIP. You will need to hook an ATA up to a fax machine and have it properly configured to ONLY use g711 for audio codec.
If you are looking to DIY, all I can suggest is that you do a lot of reading, and build stuff in a practice/test environment before you go live.
Some people will recommend you use a GUI, but I think it is much better to learn the actual software you are using.
As far as software goes, I prefer Freeswitch, but Asterisk is also an option.
For debugging SIP issues I just place a call with this app and watch the latency and packet loss meter. You can call a conference bridge if you need a number to call and idle on for a while.
The combination of FreePBX and pfSense can do everything you need. FreePBX is an open source PBX for SIP which will allow you to manage and deploy several phones or softphones with all the features you require.
pfSense is an open source router device that will allow you to manage VPN's and have finer grained control over your internet connection. It can also be used to create a VPN between your home and your parents which will simplify the encryption side of things for the call.
sure you can start at https://www.asterisk.org/ or www.3cx.com. I think you missed the point .... if you can only make 20 calls in an hour that run 5 minutes a piece and you lose money BEFORE the cost for your time and any associated costs such as servers/hosting/etc.... does it really make any sense? What if the guy stiffs you and says you didn't do a good job and disputes the payment on upwork? I think you really should consider having the person hiring you provide a phone for you to make the dials on.....
In which case you do not really need a SIP trunk. Your just complicating things for yourself and wasting money. A SIP trunk is only useful for connecting to the PSTN so you can call other people via telephone numbers. If all your friends are going to be connected via the same medium, you may as well do one of these...
This is more about your endpoint than it is about VOIP. Here's some recommendations for clients....
I have heard good things about Zoom which basically does everything you need it to but it's a commercial service.
Jitsi is another one. Free and open source. It would have been my first recommendation but it requires quite a lot more setup to get it to work as smooth, particularly server side.
Try Signal Private Messenger, it uses Opus which can seamlessly handle 30% packet loss and varying network conditions. Its way simpler than any of these VOIP providers, and it supports video calling too.
By the way, CSIPSimple is not under active development, with the last stable release being in 2014.
Uvital RJ11 Duplex Wall Jack Adapter Dual Phone Line Splitter Wall Jack Plug 1 to 2 Modular Converter Adapter for Office Home ADSL DSL Fax Model Cordless Phone System, White(2 Packs) https://www.amazon.com/dp/B07K4Q14NM/ref=cm_sw_r_cp_api_glt_fabc_18MXW7QJ3KWR3N38H25H
OBi ATAs are the best/most feature-rich/most configurable. Unfortunately, the company was purchased by Polycom recently, and since then their products have been more expensive and harder to find. Here's a refurb 200: https://www.amazon.com/dp/B083P4WS2S/
If you don't care about playing with Google Voice, the 3xx series doesn't support it, but works perfectly well with SIP service providers such as VoIP.ms: https://www.voipsupply.com/manufacturer/obihai/obihai-adapters
I see that Grandstream ATAs are available for about half the price. They don't do near as much, but as long as all you need is a simple phone, and you don't expect to ever do advanced call routing with it, Grandstream makes a decent second place.
> Does anybody have Voip.ms that uses a phone with a standard phone jack?
Many of us use IP phones for their superior audio quality and better call handling features. My "daily driver" is an OBi1032; the phone in my kids' room is a Grandstream something-or-other.
To me Bria feels the most feature complete and stable. But they switched to a bullshit $1/Month pricing so I can't really recommend it.
The Grandstream is promising, although I have had some connectivity issues I didn't have with Bria. They seem to be pushing it as more of a Video app, but it is a full featured VOIP app. It has improved quite a bit since released so I would recommend giving it a look.
You could use that after an ATA and get what it is that you need.
Line simulator like this exists: https://www.amazon.com/Viking-1126-VK-DLE-200B-Two-Way-Line-Emulator/dp/B004PXK314
However, this is a much cheaper DIY version: http://www.instructables.com/id/Simple-Intercom-From-a-Pair-of-Old-Corded-Phones/
The basic idea is that if both phones are off hook, and connected to the same fake phone line, and there's a bit of voltage to simulate a regular phone line, they will be able to talk to each other.
What if you ported the number to a SIP Trunk, then put their phones behind something like this: . http://smile.amazon.com/Grandstream-HT-502-Dual-Port-Gateway/dp/B002FA1MUK/ref=smi_www_rco2_go_smi_g2243581662?_encoding=UTF8&*Version*=1&*entries*=0&ie=UTF8
You might look into something like this:
http://www.amazon.com/CradlePoint-MBR1400-Mission-Critical-Broadband/dp/B00649T9WG/
...and get something a little more reliable than DSL as your primary. If cable, VZ FiOS, or some other fiber is available in your area, make that your primary, then get a LTE wireless data plan for your backup.
One thing to keep in mind, however, with a redundant/failover WAN link setup, you might have peering issues with your ITSP. Typically, for over the internet type service (not a private circuit to the ITSP/ISP), the ITSP will want to peer with, perhaps even IPSEC tunnel to, a single static public IP address.
Now, as an example, say you have VZ FiOS for your primary and VZW for your backup. You'll have two unique IP addresses on those two circuits. Make sure your ITSP can support this.
EDIT: The cradlepoint link is just a point in the right direction. Make sure that whatever you purchase supports all the features you require (IPSEC VPN, QoS, whatever).
EDIT 2: As far as having a second downstream router with it's own static, consistent, public IP range, your ISP(s) would have to configure both your primary and backup circuits to route to that range. What you're describing, will require two circuits with unique public WAN addresses, that both route to the same public LAN IP range. Make sure you make that requirement clear, if that is what you intend to do. If you can get that set up, you wouldn't have to worry about peering issues with your ITSP and SAP, as the traffic source/destination public IP would remain the same regardless of which circuit is active.
honestly .. i went and got a cheap android "trac phone" on sale at my local grocery store for $9. It just uses wifi - no cellular data. I put Grandstream Wave on it and gave it to my 2 year old so she can "make phone calls".
It's crazy to me how much you can get for $9 lol.
Any "wifi sip phone" is gonna cost you a lot more, and probably not function any better (if not worse) IMO.
You can use what'sapp, or kik, or whatever other messenger for your phone as long as you have a data connection. you can also use this app for android for voip.ms SMS. It is a little buggy, but works. Also, look into the new Grandstream wave app for Android; I find it does MUCH better with calling situations and better battery performance than Zoiper. While you won't have a seamless experience with these apps and voip.ms, you'll still get a good simulation of a 'cell phone' if you can deal with it.
Skype over WiFi to keep it easy... But I hate the Android Skype app...
I dropped my POTS landline 2 years back and moved to Callcentric. We rarely use it anymore and keeping the number and our minimal calling costs around $9/month (that's actually for 2 lines). Callcentric is perfect for home use; the price is right and they have great features (I love the blacklist feature -- our spam calls are way down!).
Once you have VOIP service, you can install a SIP client on your phone. I use CSipSimple ( https://play.google.com/store/apps/details?id=com.csipsimple&hl=en ). My house phone (hiding behind a Grandstream ATA) is extension 100 and my CSipSimple client is extension 101. When I call home, i just call 100 and it works.
SIP over WiFi or GSM ain't all that great, but it works in a pinch.
I also run Jitsi on my laptop and that's a big step up if you have a good connection.
Ah, that's because Google makes it a separate download from the Play Store. Install it and you'll get a Hangouts Dialer!
https://play.google.com/store/apps/details?id=com.google.android.apps.hangoutsdialer
I've been using my Google number for years now. Most of GVoice features intergrate into Hangouts just fine. There's a Hangouts Dialer so you can send/recieve calls using your GVoice number. If you would like to use a regular phone I have a OBi that also lets me send/receive calls using GVoice. I think I could answer any of your concerns about it.
Understood how it can seam intimidating. But with a little setup well worth it. Set it and forget it
PBX -
$ 261.00 US
Additional SIP to analog adaptors T801 $32.00 US
I've had good luck with the OBi2182 WiFi VoIP phone. It is currently (11/10/22) $70 at https://smile.amazon.com/gp/product/B076JKV5CL .
It has a good speaker phone, can use PoE or the included wall wart for power. The WiFi is both 2.4 and 5GHz. It has built in web setup or you can use OBiTalk.com. It has 12 soft buttons, most of which you can setup as speed dial.
I would advise to keep the phones and get VOIP phone lines that feed into your Toshiba system. They make converters that you plug into the internet that give out 1 or 2 or 4 or 16 lines, however many you want to have. You could go with Broadvoice for instance for probably a fraction of AT&T. I signed up a doctors office last week with 4 lines and it was like $100/month with that device. Its called a SIP ATA. Heres an example device that gives out 8 phone lines, https://www.amazon.com/dp/B07B6TL7N6?ref=nb\_sb\_ss\_w\_as-reorder-t1\_ypp\_rep\_k6\_1\_13&amp&crid=O5OHURBH869J&amp&sprefix=grandstream+h
Haven't used this specific adapter (https://www.amazon.com/dp/B06XW1BQHC/), but I've used a VoIP phone from the same brand and worked pretty well. This adapter for $39, should be hard to beat.
The PC may have a firewall, or odd network config. A crazy challenge of find the problem via reddit using subnets and vlans. A simple fix is to add a CAT6 to Wifi adaptor to the phone.
https://www.amazon.com/IOGEAR-Ethernet-2-WiFi-Universal-Wireless-GWU637/dp/B018YPWORE/ref=sr_1_3?keywords=ethernet+to+wifi+converter&qid=1656007138&s=electronics&sprefix=ethernet+to+wifi%2Celectronics%2C93&sr=1-3
Take a look at this thread: https://www.dslreports.com/forum/r30661088-PBX-FreePBX-for-the-Raspberry-Pi
The first post explains how to download a script that will set up FrteePBX on a Raspberry Pi.
You will need to have either a VoIP adapter and a regular phone, or an IP phone at each house. Grandstream has some inexpensive models such as this cordless unit: https://www.amazon.com/Grandstream-GS-DP710-Cordless-Expansion-Handset/dp/B008OFSP6O/
For your purposes you just need to get extension to extension calling working, once you have that worked out then you can decide if you want to add any other type of service. But at least that would get you started.
Try one of these IOGear Ethernet to WIFI adapters. You need to plug it in ethernet to your machine, then bring up a web page to set the wifi SSID and Password. So you might get her a USB to Ethernet adapter also and set that network adapter in Windows it to the fixed IP subnet needed to program this Ethernet to WIFI adapter. Once you have set the SSID and password, plug it into the phone and hopefully there is a USB port on the phone for power. Power up the VOIP phone with the power cable and you should be all set.
https://www.amazon.com/IOGEAR-Ethernet-2-WiFi-Universal-Wireless-GWU637/dp/B018YPWORE/
You're welcome. Note sure where you're located but if you're in the US, here's the Amazon link.
TP-Link AC750 Wireless Portable Nano Travel Router(TL-WR902AC) - Support Multiple Modes, WiFi Router/Hotspot/Bridge/Range Extender/Access Point/Client Modes, Dual Band WiFi, 1 USB 2.0 Port https://www.amazon.com/dp/B01N5RCZQH/ref=cm_sw_r_apan_i_RNF5Q5T1T5KR2EE8MWZQ
THIS is what you are looking for.
I have had something similar that worked very well for simple calling.
Didn't work for my old alarmsystem (modem) but calling was the same as with cell
Audio latency is defined as the time it takes for audio data to travel from its source (computer, smartphone, mp3 player) to your headphones or speakers. These are the milliseconds (ms) it takes to process digital data and convert it to an audio signal that can be streamed through a wired or wireless connection to your headphones.
In a regular wired connection, the typical audio latency is 5-10 ms.
In a wireless connection, Bluetooth latency can go anywhere from an ideal 34 ms (aptX LL) up to 100-300 ms for true wireless earbuds and headphones.
The required Real time quality metric is Below 30ms to 1minimize customer talkover and echos. Your customers may not care, but Mine (Critical to who we serve) are used to a phone call and object to quality impact. As this required metric reduces the possibility of Buffered and dropped packets. Bluetooth unlike other voip codec protocols has a "Flexible" packet size to accommodate other types of services (Especially in its advertisement segments of the packets) whereas lets say G.279 can be hard set to 20ms
To inspect your assumptions math must be applied to the physical setup in use. You can see the results propagate with more and more stacking affects.
https://www.amazon.com/RUNMI-Converter-Adapter-Connector-Ethernet/dp/B09MHX1VLT/
Just buy an adapter like this. I have a similar setup and it works. The only disadvantage is that you lose that Ethernet connection to that room. Use right adapters, the Ethernet cable is fine to use.
You would be much better of to buy a VOIP ready phone then you would just connect it straight to your router.
This one is an excellent choice https://www.amazon.com/Yealink-Cordless-2-4-Inch-Display-Ethernet/dp/B076WVZY2P/ref=sr_1_3?keywords=VOIP+cordless&qid=1647632422&sr=8-3
Expensive but can be found on eBay very cheap
Paul
So you’re going to have the laptop connect to wifi (which doesn’t actually exist right now because the user is waiting for internet installation) and then you’re going to plug the phone into the laptop via a USB Ethernet dongle and let the laptop be the router?
That’s such a Rube Goldberg solution.
Get a wifi/ethernet adapter and plug the phone into it instead of relying on Windows to be your router.
Something like this: BrosTrend AC1200 Ethernet-2-WiFi Universal Wireless Adapter for Printer, Smart TV, Blu-Ray Player, Game Console, PS4, Xbox https://www.amazon.com/dp/B0118SPFCK/ref=cm_sw_r_cp_api_glt_i_893NNEJJC85AYBJ3GJEC?_encoding=UTF8&psc=1
Check the codes in your area if they even allow fire and elevator lines to run on SIP. Many places do not allow this.
But if they do, yes that is one of the options.
​
Or
​
you could simply get a basic load balancer specifically for the elevator(s) and fire lines in the building and use multiple WAN/Wireless Carriers for redundancy/backup. Something basic like TP link should do the trick (if you're not looking for super advanced stuff). Link
Forget that garbage. Sign up for google voice and get an ObiHai (now poly) adapter. Those work great and the calls don't sound like garbage. Best part it is all free other than the device.
https://www.amazon.com/OBi200-1-Port-Adapter-Support-Service/dp/B00BUV7C9A
Something like this: Grandstream Hybrid ATA with FXS and FXO Ports (HT813) https://www.amazon.com/dp/B07GY4WWP3/ref=cm_sw_r_apan_glt_i_ECEXYH27W14A4VT6QF7Y
So I plug my pots line into fxo port, and ethernet to my switch? How would I be able to use a voip phone over my network in let's say another room etc?
Thanks
You say "Cloudflare doesn't do SIP natively", but https://www.cloudflare.com/en-au/magic-transit/ specifically mentions "Voice over IP (VoIP) and custom gaming protocols." If they can DDoS protect random companies' gaming protocols, shouldn't SIP be easy?
This worked!
Background information on what caused this breakage in the first place:
https://letsencrypt.org/2020/12/21/extending-android-compatibility.html
https://blog.germancoding.com/2021/04/16/lets-encrypt-and-expired-root-certificates/
Oh yes SIPREC. Thanks I've corrected the post.
To be honest, I don't know how to do that at all. I looked up some public materials for the tool we're using and found this deck, slide 13 is basically what I'm working on - IBM Voice Gateway would be the SRS that's in #2 or something that's conferenced into the call in #1.
Instead of the Watson services on that slide, we're demoing some other tools. All the analytics tools are going to be services with some REST APIs so I can't configure the networking settings.
That one is designed for a Star pattern network - i.e. direct runs from a switch to the phone, it won't work well in most residential settings if your phone lines are run in a daisy chain pattern.
You can opt for a tester that also as a tone test + tone probe or, just buy a tone generator + tone probe:
https://www.amazon.ca/Network-Tracker-Telephone-Ethernet-Handheld/dp/B07MTFCYXY/
>cable tester
I will buy a cable tester, thanks. Is something like this $15 one good enough, or will it not reveal what you want me to check?
I am in the same boat and have been exploring platforms.
I'm leaning on Freshworks CRM for my business and it checks most of these boxes that you would like to have.
they have a free trial, i'd recommend checking it out.
Atom is pretty decent, I use it daily. Shows you what you've changed since you last saved, has a few XML plugins (schema linters and similar) and handy diff, find and replace and other tools to make editing easy.
> But to help voip.ms I guess this won't suffice. Like you're saying, something needs to be done at the edge
Exactly ... It all breaks down to the "scale" of both defense and attack.
There are many solutions on the market. From commercial for a lot of $ through proprietary solutions of specific operators.
Unfortunately, no matter how much we fight, when the attack exceeds a certain scale, unfortunately we will die.
For example, such French OVH claims that they are able to filter out the 4Tb / s attack. Cloudflare claims they are capable of 100Tb / s.
It is said that the attack on voip.ms could have been plus or minus 350Gb / s ... enough to kill small players, and their infrastructure was not ready, even on a smaller scale, and there are unfortunately the effects of this.
My servers quite often report attacks like 80-150Mb / s which is nothing tragic but if I get 350Gb / s I can turn off the light in the office and go home. :)
https://www.ovh.com/world/anti-ddos/
Not everyone can afford equipment per unit at the starting price of $ 300k and entrusting the matter to a dedicated ASIC or FPGA in such toys. I am left with a modest x86 :)
> Some kind of Cisco box probably. Inside it must be dedicated hardware, probably a network optimized flavor of Linux, running on (a bunch of) ARM cpus maybe?
Cisco has its own dedicated OS like most manufacturers of this class of equipment. While the typical unix roots ... ARM happens, but usually there are dedicated ASIC chip or FPGA and dedicated network chips that carry a lot of weight.
Processors in architectures such as x86 or ARM are also often found, especially in switches, but such cpus services for operating the OS and heavy tasks related to all traffic are performed by a dedicated network chip. You can still find a whole mass of 100G switch which have Intel Atom or Xeon low end for OS support but they don't do anything advanced
Have you considered purchasing a dedicated server package from OVH or similar? I can guarantee you that the virtualization factor of these VPS's is what is giving you the jitter. Been there, tried that. As mentioned by others, they are not designed for real-time applications that require dedicated resources like a bare metal server can offer.
Stats On 100% Open Sourced FreePBX Modules Over the last year:
5779 Commits...Up + 84 (1%) from previous 12 months
58 Contributors...Up + 7 (13%) from previous 12 months
50 Commits to OSS Endpoint in the last 150 days. (https://github.com/vsc55/endpointman)
I appreciate you providing options but we are already heavily embedded with 3 different providers currently. It's just confusing because everyone here is saying "no its not possible". But when you look at real world examples like https://close.com/calling/ they even state publicly on their page that they charge Twilio rates directly.. thereby letting Twilio collect the tax not them.
hey - david, co-founder from Telnyx here. We run a geographically diverse network that prevents failures like this. Unlike most providers, we control our own IP address space https://ipinfo.io/AS63440. This allows us to use anycast to distribute traffic and limit failures. We can always pull BGP for a particular site if it goes hard down.
We have points of presence in Vancouver, San Jose, Toronto, Chicago, Ashburn, and London (and growing). All infra is 2N and the underlying infrastructure is dockerized and split across all the cloud providers. We also have our own backhaul network, so we can pull traffic off the internet at the earliest opportunity to give you the highest quality calls.
Always happy to answer questions on how we make our infrastructure stay up all the time.
Happy to give any redditors $15.00 in free credit to test our service - TXKARMA is the promo code.
I use Filmstro for royalty free music. You can get 90 second clips in many genres.
They have a paid tier where you can modify the tracks intensity, etc. But most tracks are free to download and are even YouTube approved, many YouTube personalities use them.
FusionPBX is light enough as to easily fit onto a $25 OrangePi PC if you want to do an on site PBX. Thing is, for a few dozen users or less, its better just to host the PBX off site and save the hardware cost and centralized updates and maintenance.
This is what it can report:
line registration, incoming call, outgoing call, onhook, offhook, user login/logout, call state change.
Probably wouldn't want a state change for each phone call.
​
My simple test was to have it trigger a script on a webserver where that sends a command to pushover: https://pushover.net and finally to my smart phone. It looks like I could even grab the callerID, since that's submitted via XML.
​
OpenVPN on the handset is not the recommended way to go. It's incredibly fiddly and if you have a config change it starts becoming a nightmare if you have more than a few devices.
It's better to have a correctly set up router/firewall with OpenVPN. My recommendation is to use pfSense.
If you have got one, get an old computer and put a second NIC in it and go and install something called pFsense. It's open source router software. It comes with many, many, many network analysis tools and sip optimisation features. Use that as your router for a bit and log the traffic for analysis, should help you out (Might even cure your problem)!
IPSec is a vpn protocol.
Any vpn that can route the required packets can be used for voip traffic.
You can make vpns using hardware or software only.
popular software only ones are ZeroTier, https://www.zerotier.com , or Tailscale, https://tailscale.com/ . Mainly due to how easy they are and free.
You basically install a client on the devices you want on the vpn, and that’s it. They are now in the same network.
Depending in your Voip server and client it uses, you’ll need to ensure they talk over the vpn network.
If you have the resource, you could configure an open source SBC (Freeswitch is capable of this) and use the CallRecording functionality in the software.
The other easy option is to use tcpdump or wireshark https://www.wireshark.org/ to capture the SIP and RTP packets then export the media as a sound file, more info here: http://wiki.wireshark.org/RTP_statistics
If you want to diagnose what the issue is, you should probably start by taking a couple packet captures. First from your phone to your router/firewall, and second, from the router to the outside.
This link will point you in a good direction: http://wiki.wireshark.org/CaptureSetup/Ethernet
Have your SIP provider collect their packet captures as well so you have something to compare to.
A popular reason for choppy audio is due to lost or out of sequence packets. Wireshark has a whole menu dedicated to telephony troubleshooting.
https://www.wireshark.org/docs/wsug_html_chunked/ChUseTelephonyMenuSection.html
The free one is limited to 40 minutes (unless they lifted it for the COVID stuff), you need to pay for the next tier to get 24 hour meetings.
If you want to tie into your own PBX for PSTN calling, they charge you $48/port/mo
If you want to do your PSTN calling through Zoom, they charge you per minute at a higher rate than Webex charges. They have a rate table on their web site but it's a few cents higher than it should be.
So yeah, Zoom is cheap if you just need basic small meetings with no PSTN access, but it can get expensive really quick.
From the looks of their page it seems like BYOC was the way it works, it's mentioned in more than one place prominently.
https://zoom.us/phonesystem
But, I'm happy to be corrected.
Perhaps Zoom? $20 a month for one host so if I'm looking at it properly, any one of the 3 could be that 1 host for the meeting, so you wouldn't need 3 pro licenses or if they're always going to be the same host, it still works out.
Can try it for free as a test
SIP softphones in the background are hit and miss. I've used Bria and while it does seem to work well it WILL smoke your battery life through the course of a day.
FreePBX + voip.ms trunk on Linode / Digital Ocean / vultr.com / AWS for this size of operation would probably be ok. I tend to see problems with non-voice specific hosting providers at higher concurrent call counts.
Another option would be any of the numerous shared providers such as grasshopper, telzio, vonage, 8x8, etc. You trade away the ability to customize your PBX and functionality and get back ease of use and less time managing your PBX; it's up to you what your needs are and this may be the best fit for you.
Good Luck!
sipXcom is probably one of the better open source platforms with redundancy/HA. If you want commercial support and more enterprise features you'll want to get in touch with someone at eZuce. I used the eZuce platform to replace a 1,000 seat Meridian system at a county government in Texas with great success.
Oh, it works fantastically with Polycom phones.
If you have questions, feel free to message me.
sipXcom is the extension of sipXecs. There was some sort of split, I dunno the whole story, and the result is sipXcom and seems to have more activity now. sipXcom is the community edition of EZ UCE, if you need formalized support. Both are well supported with an active community.
TrueConf is a self-hosted video conferencing software system that performs well even with slow Internet connection or low bandwidth. The free version supports up to 12 users and can be installed on a regular PC, so you won't need to invest into hardware endpoints: https://trueconf.com/products/tcsf/trueconf-server-free.html
You should check out TrueConf Server, it’s a self-hosted UC system that supports audio and video communications, integration with telephony and business processes. Plus, there are client apps for every major platform, both mobile and PC-based.
Hey I setup a custom VOIP box for a small business which I'm currently upgrading now. It's using Elastix for the VOIP software on top of CentOS for the OS. We're using Flowroute for SIP. Phone bill was over $300/month and is now down to under $30/month with significantly more functionality than they had before. The server is running on an old laptop and we're using Polcyom IP550 and 650 phones.
Since you're looking for something easy to setup I'd go with a hosted service. I never came across any that I really enjoyed. Used RingCentral with a client but was dissapointed.
TextNow's $4.99/year plan will preserve the number. This is a relatively "pricey" $9.99/month plan but it seems worth it for the various features and no ads and also the number preservation you mentioned.
I don't know what TMO towers is. They do say "new faster coverage for Sprint and GSM devices" on this page: https://www.textnow.com/wireless/coverage
[Full Disclaimer: I work for this company. Shameless but relevant self-plug]
You may want to look into Toktumi. Everything is hosted on our servers, purely mobile. (Software phone clients for Windows, Mac, iOS, and Android. Mobile app, called "Line2", can be downloaded from respective app store for free.)
You can sign up for a free 7 trial. (Truly free - you don't add your billing info until you want to pay for the service.)
This is likely the only comment I will ever make because this is my "lurk reddit at work" account, but if you have any questions feel free to ask.
[Ninja edit: Formatting]
+1 on MicroSIP. I have used MicroSIP with my FreePBX setup at home, no issues. With our Grandstream UCM PBX at work (asterisk based), MicroSIP didn't perform well over the WAN or maybe just not compatible with Grandstream. But Blink worked great across the WAN with Grandstream.
Buy some cheap Android handsets and install a sip soft phone on them. Bria by CounterPath is what we are using across all of our platforms (Windows/Mac/Android/iOS). You should look at their saas version as it gives you a means to manage the end points.
Counterpath Bria has a softphone for most platforms, definitely for Windows, Android, and iOS. I haven't used it in a touch interface on Windows, but have used it on Android.
If you are looking at multi-platform clients, Counterpath should be on the short list.
Check out Openfire with the Asterisk plugin plus the Spark client.
http://www.igniterealtime.org/projects/openfire/
http://www.igniterealtime.org/projects/spark/index.jsp
It takes some work to set up, but you can do some cool stuff with it.
I've never had a problem with Zoiper on iOS. maybe my iOS is broken.
edit: You may find this helpful: https://www.zoiper.com/en/support/answer/for/ios/22/Receiving_calls_in_background
Admittedly, my phone can't run the latest iOS, they may have fucked things up more.
For most phones, I like AAstra (Mitel now) 6730 series. They're built like tanks, easy to configure, and have good speakerphones. And the 6739 is a pretty sweet phone- as long as you don't mind paying $300+ for it, you get a big color touchscreen and Bluetooth built in. I personally haven't tried the 6800 series because I don't like the interlaced softkeys...
For conference phones we run the Polycom SoundPoint IP 7000. Not cheap but it sounds bloody amazing.
I also like Snom 300 series for a hobbyist perspective. They're very tweakable and they can load custom ringtones on the fly.
The newer Grandstream's aren't bad, but I'm not a fan of the Android ones because there are tons of settings you can only change from the LCD panel.
Yealinks I've heard good things but never actually used one.
Finally, for a platform try FreePBX
//edit: Why the downvotes? Is it because I suggested another vendor (which has since been removed)?
Its merely security through obscurity, which might be your thing. Most of the SIP brute forcing we see is directed towards our IPv4 address, with the occasional attempt towards our rDNS domain. Setting up Fail2Ban and configuring LetsEncrypt are much more practical steps to take than trying to use DNS as a way to hide, reminds me of Sandstorm.io's silly use of the same technique. DNS will leak your "secret" subdomain like a sieve by the way.
I just tested Jitsi. The option to record a call does not become available until AFTER the call has already begun. On top of that, it prompts for a file name to save the call to, so by the time the recording starts, substantial portions of call are missing. If I were to use my recordings to verify business agreements, important parts may not be recorded, like the name of the person I'm calling ("Hi, can I speak to John?" "This is John speaking"). Worse still, if I had to use a recording in court to prove an oral agreement, the obviously missing portion of the call makes the whole recording useless, because being incomplete, an adversary could say I just cut out the important parts. Jitsi is Java too, ewww! Haha. Thanks for this suggestion, it might be useful for something else, but the call recording feature seems to be intentionally designed to not be useful, just so it can be in the feature list:
https://jitsi.org/Main/Features
It looks like the call recording feature got serious attention based on this advertisement for a Google Summer Of Code 2009 candidate to add call recording to Jitsi:
https://jitsi.org/GSOC2009/CallRecording
How they managed to screw this up so badly, I have no idea. I guess it starts with bad leadership and bad decisions...like choosing Java. I really wanted this one to work!
G.729 won't fix this issue. At best, G.729 can do a so-so job of covering up 5% packet loss (assuming you have a good implementation of G.729 and you don't loose too many packets in the wrong places for the concealment algorithm). A simpler and free solution like Signal Private Messenger will work much better, fully concealing up to 30% packet loss while providing full band audio, as compared to G.729's tin can audio.
Of course it can, that's the whole point of google voice.
check https://mail.google.com/mail/u/0/#settings/chat and make sure that chat is on and that you have the plugins installed under voice & video. There is also a link there to verify your settings.
Also check the settings for Google Chat in https://www.google.com/voice/b/#phones to make sure that you don't have any restrictions on when that phone rings.
here is how you "decruft" an amazon link
> https://www.amazon.com/Obihai-OBi2000-Gigabit-Phones-OBi2182/dp/B076JKV5CL/
Just remove everything after the amazon stock number. And that's before we start using their own URL shortening syntax.