I'm a little surprised that there hasn't been a thread about how we all can harness VoIP to assist in the coronavirus pandemic.
At Nerd Vittles, we have offered some suggestions on working from home with Asterisk. http://nerdvittles.com/?p=31900
The 3CX folks have extended a 3-year free license to those needing to work from home. https://www.3cx.com/blog/news/covid-19-remote-working/
And 3CX also has offered a 100-user E-Learning solution at no cost to schools. https://www.3cx.com/blog/news/e-learning-solution/
Please post any other offerings in this thread so everyone has a quick resource for solutions in these trying times.
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[PBX] Coronavirus VoIP Resources
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[General] What you guys, guru of Voip would recommend for this need ?
Hi, I need to help a Non-Profit accomplish this (in Canada) :
- have a system with "extension numbers" for about 10 persons in the Office.
- have the possibility to be "remote" (at home or on the road) and be reachable by calling their Office "extension number"
- Possibly, have a way to use a PC/Mac/iOS/Android device as the "phone" instead of a regular phone.
- Support a local FAX (they work with Medic Clinics and they still use faxes)
- Have a way to log all the calls
- Lower the phone bill
- Nice to have : have a way to record calls to a local server, have a Web Interface to manage users, Extension, settings, etc.
I'm in IT since last 2 decades, so I'm able to install anything. It's just that I never did telephony / voIP before, and I want to help the Non-Profit, because they offer medical help at home, especially useful with COVID-19 pandemic...
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Linksys SPA2102 Assumed to be Locked to an Unknown Provider.
Hey, Sorry this is my first post so bear with me. I recently acquired a box full of Linksys ATA's and a majority of them appear to be refusing to factory reset with the prompt "Enter Password" when dialing #73738 However upon examination of the main user page. I get the information making it appear to be a standard unlocked model. (Customization = Open, etc.)
However I still can't reset it because there is a password on the reset.
I searched these forums before but found no logical solution because I am unable to identify if there is any company that put this password on the device to begin with.
So I have decided I have no choice but to ask around to see if anyone has a solution to this.
Any help would be appreciated. Would be a shame to have a box full of paperweights.
Thanks!
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Voipo acting up for anyone?
2 residential Voipo lines are acting up.
- One seems to be working for inbound/outbound after multiple reboots. However after hanging up, it repeatedly calls the last dialed number.
- Other has dialtone but doesn't connect outbound calls right away; e.g. called my cell from it, got nothing on the handset, hung up. 2 minutes later, call appears on cellphone. Multiple reboots to no avail.
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[Anveo] Anveo - using Obi 202 and SIP client on PC at the same time
I have Obi-202 connected to a regular phone, everything works fine.
What I would like to do is to run SIP client on my Windows desktop at the same time so that if someone calls me it would ring both the phone and SIP client at the same time and I would be able to pickup either.
Is this something that Anveo supports - 2 devices registered and active at the same time? If not - is there a way to still get it to work somehow using the Call Flow?
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[CallCentric] Callcentric Call Treatment Question
A while back I ported my home number from Ooma to Google Voice. I added Callcentric for 911. I have my Google Voice setup to ring the Callcentric number, and have a list of numbers in Callcentric that I have various call treatments applied to. One treatment I use a lot is "Number disconnected" for telemarketers, political calls, etc. I use this all with an Obi212.
What's actually happening with these calls? Is the call technically still ringing, but the caller is hearing a disconnected message instead of ringing? The reason I ask is because I have one that's set for "Number disconnected" but they somehow end up back in the Google Voice voicemail and leave a message. The phone never rings on my end - so the call treatment is working in that respect. It's an automated call, no live person. It's always the same caller this occurs with. None of the other numbers I have set to "Number disconnected" do this.
The only thing I can figure out that's going on is this automated caller isn't recognizing the "Number disconnected" and continues on with its thing until Google Voice voicemail grabs it because the call is actually still ringing. Or is there some other possible explanation? My whole reason for using the call treatment is so I don't have to deal with these calls at all. The phone doesn't ring, but then I have to go in and mess around with deleting the voicemails.
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[General] Providers offering connectivity to 211 state emergency help line
Good day.
Since the start of the present COVID-19 debacle, TV channels in Massachusetts are bombarded with PSA about calling the "211" Mass. emergency help line.
I had heard about municipal 311 information lines, but had never heard of 211 state lines.
Before I request information from my current VoIP providers about their ability to connect to the 211 line (or lack thereof), is any one aware of any such provider?
For extra credit :), has any one actually successfully connected to 211 who cares to share?
Thank you.
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Good pay-as-you-go France/EU mobile SIP termination providers?
I've been using a combination of Voip.MS and Localphone for French mobile termination, but Localphone has jacked their France Mobile termination rates up to $0.18/minute.
Voip.ms's seem to have been set ages ago and never really came down. Calls which terminate on the exact same carrier have wildly different prices just based on the phone number. If you get lucky with the phone number you are calling, the price is within reason, but if you don't, it is not.
France mobile termination is wholesale regulated by ARCEP to cost around $0.01/minute. There can be some additional charges if the originating caller ID is from a country which *themselves* charge high mobile termination rates (the idea is to allow operators to reciprocally charge higher rates). Most of these charges are not that significant. But they don't even apply to me - I have callerIDs in North America as well as in EEA. North America is not allowed much surcharge at all, and EEA caller ID definitely isn't allowed any surcharge.
Not only that, the French regulator mandates that French operators permit interconnection over SIP! Calls to French mobile phones just shouldn't be expensive.
Even if they need to plan for highest regulated wholesale rates and even assuming a provider isn't capable of rating calls using origin-based pricing, I don't see how the retail prices being charged are justified.
For French termination, Twilio is charging $0.05/minute for termination with an EEA caller ID for something that is going to cost them closer to $0.01 at most. Zadarma seems to offer prices closer to reality at $0.03/minute if I call with an EU origin caller ID. They are one of the *only* providers I've seen offering pay-G rates that aren't obscene and I don't understand why this is the case.
I have no knowledge of how good/stable Zadarma. I'd love feedback on other people's experiences with them. Also looking for suggestions for good alternative providers (I'm want looking for "load funds, use account balance, refill when needed") type service. I'd also appreciate any insights into why there is this huge disconnect between wholesale prices and the prices that I'm seeing in the market.
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Call Centric Incoming DID not ringing
this is new, prolly since everyone came home. i am now randomly not getting inbound calls, its a busy signal, looking at reports in call centric, it doesn't even indicate a call was received so its a call centric problem, its not even hitting my pbx.
anyone else seeing this?
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[Voip.ms] Voip.ms is busy - New Beta - Call Recording
You are receiving this email because you are signed up to the VoIP.ms open beta program. We would like to notify you of the start of the beta-testing phase of our Call Recording Feature.
The call recording feature is located under "DID Numbers" then under "Call Recordings". You can now record your calls whether they are incoming or outgoing. You may also consult the customer portal to manage all your recorded calls including listening to them, downloading them or even forwarding them over to an email address. For more information, you may consult our Wiki article here.
The purpose of the VoIP.ms beta program is to test the new VoIP servers, features and services before final release. Problems found during the beta test when using beta applications or servers may be reported to our staff and will be reviewed in order to fix them before final release. We do not recommend running critical production environments on our beta features and servers.
If there are critical issues reported, they can be reported via live chat or ticket and they will be reviewed/corrected immediately by the staff. Minor details will be corrected within one business day and can be submitted via ticket.
Your cooperation means a lot to us. We sincerely appreciate you helping VoIP.ms to be a better company by testing the beta features before they are released to the public.
Thanks for being a VoIP.ms beta tester.
Best regards,
VoIP.ms Team
https://www.voip.ms
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News Item: FTC Warns 19 VoIP Service Providers about robocalling
FTC Warns 19 VoIP Service Providers That ‘Assisting and Facilitating’ Illegal Telemarketing or Robocalling is Against the Law
The 19 service providers were not named:
https://www.ftc.gov/news-events/press-releases/2020/01/ftc-warns-19-voip-service-providers-assisting-facilitating
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[Other] Bandwidth API Discontinued
Bandwidth just sent out an email stating "We are writing to notify you that on March 31, 2020, we will discontinue our support of the V1 API platform." It states that Bandwidth is developing a new version 2 APIs. It states that if you’d like to be considered as a candidate for their V2 APIs, a new contract with a $250 minimum monthly service commitment is required. Alternatively one must migrate their services to another carrier. If nothing is done by March 31, all services will be terminated and, if applicable, refunded.
Is four days notice reasonable?
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[General] GVJack App Stopped Working a Few Weeks Ago
Help!! I'm a registered user of the GVJack App (version 3) and have been using it for a few years with no problems until about 4 weeks ago. The app uses the RTC Browser and the auto login with Google fails with the message "something went wrong". I can manually log on using the RTC browser but can't proceed - the app requires the auto login. I emailed PCphonesoft.com and have had no response. When I check for updates through the app it says I'm using the latest version. Any help would be appreciated.
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ObiHAI Obi100/Obi110 Firmware Mod Discussion
So many of us have the Obi1xx series devices that recently stopped connecting to google servers due to a certificate update. This thread is intended to discuss the possibility of modifying the firmware to update the certificate and let these devices work with Google Voice again.
These devices are based on a MIPS-X processor similar to the Sipura ATAs and there is not a lot of tools/docs out there about them except for a Yahoo Group mostly related to DVD player chipsets. The venerable DogFace05 who was an expert with these types of devices once posted that he was able to extract this firmware sucessfully. Not sure if he is still around. Anyone else familiar with this architecture?
It seems that the place to start looking is the end of the firmware update file which contains some kind of table. Then there seems to be a loader section which presumably decompresses one or more other sections and loads them to RAM before executing the firmware.
So the questions are:
Can we extract, modify, and repack the firmware and create proper checksums/signatures?
Where is the certificate stored and in what format?
Can we drop in a new certificate without messing up other things (e.g. if the length of the certificate has changed) or do we need to move the certificate and patch the code pointing to it?
Is updating the certificate enough or is the codebase missing support that is necessary (e.g. if key length has changed)?
Anyone who wants to participate please post your thoughts.
Thanks
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Cant log in to Obihai Forum
I like many others have recently suffered from the Obi200 hangup problem reported on the Obi forum several days ago. Long ago I created a unique login (username not e.mail) on that forum but since the Black Forest colorado fire took all my computers. I have a separate ObiTalk login for the Obi Dashboard for the Obi200 phone configs. But this will not let me login to the Forum! The Forum signin page demands a username and NOT an e.mail. I have one from long ago.
So, I want to log in and interact to get my 4 Obi200 working again.... but my username login will not work. AND I cannot find anywhere a password reset for the Obi Forum!! Searching the Forum provides nothing. Same for Google search. Does anyone here know how to get the Obi Forum login working, password reset etc?
thanks
rich
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[Equipment] Reccomendation for replacing Cisco 303 desk phone
Cisco 303 is EOL'ed now. https://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/small-business-spa300-series-ip-phones/eos-eol-notice-c51-741208.html
What is the forum recommendation for a low budget desktop VoIP phone with power supply? (NO PoE).
Two line phone is fine too, but it must have user buttons (hard or soft) for call forward and call hold that work with voip.ms service.
Thanks
BTW I am open to other hardware manufacturers based on your personal experience. Not just
cisco model replacement
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[Voip.ms] Achieving Per-Call CID Block using VoIP.ms
Yeah, so this is my (ultimately successful) experience in trying to achieve
per-call CID blocking using VoIP.ms.
Please gently correct anything incorrect.
If there's a better or easier way (with a freePBX in the loop) please advise.
First, VoIP.ms does NOT inherently support per-call CIDblocking
(via vertical service codes of *67, *85 or #31#).
The way that one must achieve this is by:
- configuring (at least) two sub-accounts on your VoIP.ms account
- using a front-end [software] PBX to route calls to a different
VoIP.ms sub-account depending on whether per-call CID blocking
is used on that account
- and as such your SIP client or softphone will be logging into
your own PBX, and not directly to VoIP.ms
- If you don't want/can't have the PBX you can do all the indicated
VoIP.ms setup and then on your softphone you'll have to setup
two different SIP-provider accounts (one for each sub-account)
and that way can achieve per-call CIDblocking - by choosing which
of your two SIP account_setups is active when you place an
outbound call
So for me, using FreePBX, I:
- setup two (outbound) trunks, each connecting via a different
VoIP.ms sub-account
- setup two outbound_routes, each using a different trunk
- set the trunks, under pjsip settings, with only one registration
set to send, the other to none
- set the trunks, under pjsip settings, with one trunk using udp
port 5060, the other uses udp port 5080
- for the showCID trunk, put the proper CID into the
OutboundCallerID field
- for the hideCID trunk, affirmed the HideCID option
- use dialplans on each of those outbound_routes to determine
which calls are handled by each outbound_route
- e.g. for the hideCID outbound_route, place *67 in the prefix
field of a dialplan number (so that it gets stripped from
the dialed_number sent to VoIP.ms), and set the
TrunkSequenceForMatchedRoutes to be the hideCID trunk
- for the showCID sub-account at VoIP.ms, set DeviceType to
be Asterisk and the CallerIDnumber to be "I use a custom..."
- for the hideCID sub-account at VoIP.ms, set DeviceType to
be Asterisk and the CallerIDnumber to be "Use one of my DIDs"
[yes, this appears to be contradictory; uncertain of the necessity]
Some of these settings (e.g. using different UDP port numbers for the two
trunks) were arrived at by trial and error, and solved intermittent
connectivity problems for me. (In general this whole setup was way more
finicky and problematic to setup than IMO it should have been.)
Anyway, that's what worked for me in the end.
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Need a FAX server solution
I have had good luck with Elastix and Hylafax on voip.ms g.711 codec. I just retired my in house server and moved my IVR to voip.ms.
Now I just need in-house fax solution.
RonR's Raspberry Pi solution will work (anybody using it in a small business?) but, is there another simple stand-alone solution just for the fax instead of installing Asterisk if there is no need for IVR? Winprint-Hylafax and fax to PDF worked great on Elastix and I would like to have a similar solution. I cannot use voip.ms fax service for privacy concerns. Yes, voip data can be tapped too but it is reasonable best effort if I have a in-house fax recipient server vs. incoming faxes sitting on outside server.
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[General] CallWithUs: This Is Most Unfortunate
Quoting CWU's website:
CallWithUs has always aimed to avoid working with telemarketers, as we disagree with their business on principle. Unfortunately, this has been difficult, as the majority of recent sign-ups have been from telemarketers, creating a lot of work for us in filtering them out. To help stem the tide of unwanted calls, we have temporarily disabled new sign-ups so that we can better focus on our existing, legitimate customers. Sign-ups will be reopened eventually, please check back in a year or two."
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[Voip.ms] Grrr.... Registration Failure
Today I'm battling as to WHY one line on my YEALINK W60B associated with a voip.ms sub account just won't register. I've even tried creating two new sub accounts and associating one with the Yealink line #3 that wasn't working, and one with line #4, and all I get is "Regsitration failed". The other two lines all register correctly with Voip.ms (and all are associated with the same voip.ms server) :(
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