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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News Item: FTC Warns 19 VoIP Service Providers about robocalling
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What VOIP carrier has 603 (New Hampshire) dids available?
Just need 1 for commercial use. Must be portable.
Thanks
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needing help on ATA as trunk
Hello Guys
I am looking to make an ATA as my incoming and outgoing device to my freepbx and not sure how to really make it work I am connecting behind lets say charters eMTA I need to make that work coming from ATA to freepbx using sip what settings do I need besides the trunk setup?
I am using Grandstream HT702
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[General] Won 11 numbers in the 833 Auction, trying to pay less if possible
As the subject reads, I got 11 decent phone numbers in the 833 auction and I’m trying to be as much a cheapskate as possible. For the interim I’ve been paying Twilio their ~$2 a month per number to keep them active and forward calls to my cell phone at the couple cents per minute incurred. I’m wondering if anyone might know of any slightly cheaper options in terms of ITSPs to home these numbers on
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[Equipment] How to call a SIP URI with a Polycom VVX411 SIP Phone?
Good day.
Every time I try to call a SIP URI with my Polycom VVX411 SIP Phone I hear fast busy tones and there is no connection.
Below, my phone's specifications:
Phone Information
Phone Model VVX 411
Part Number 3111-48450-001 Rev:A
MAC Address [REDACTED]
IP Mode IPv4
IP Address 192.168.1.42
UC Software Version 6.1.0.6189
Updater Version 6.1.0.6163
I recently added the following configuration option to permit calls to SIP URIs:
<?xml version="1.0" encoding="UTF-8" standalone="yes"?><feature feature.urlDialing.enabled="1" />
However, I'm still unable to make calls to SIP URIs.
Any ideas?
Thank you.
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can i setup ooma with 2 ported numbers (one never ring)
I've paid for a private number from day one 30 years ago so i hope the privacy of the numbers would be maintained here if i switch to ooma. i literally get maybe one spam call a month. sometimes none.
i have a main phone number for family and friends but i have a 2nd number that i give other places that need to get ahold of me but i dont want to get the spam calls. (credit card companies etc).
currently the spam line is hooked up to a cordless phone with answering machine and never rings. i just look at the unit to see if there are any new messages daily. the 1st line is on another cordless phone answering machine and rings.
i understand for 2 lines on a box you need premier for 10 a month. ooma says my taxes are 5 per line. i currently pay 45 plus taxes (10) totaling 55 for the 2 lines so of course the monthly fees is what i'm looking to save. I would post in ooma forum but you need a number to post there.
1. so would the monthly fees for 2 numbers be 10 for premiere plus 10 in taxes? (this saves me 35 and is nice indeed)
2. i actually would prefer the option to keep using my 1st answering machine. can i turn off ooma VM for phone one while keeping ooma VM for phone 2?
3. can i dial out from both numbers with just one phone? (sometimes when you get a new CC you have to call in from that number), if so, how do you do that? it would be nice to eliminate the 2nd phone as it will not even hold a charge anymore.
4. obviously i can just buy 2 ooma boxes so then my fees would be just 5 in taxes per box, correct? this means only 10 bucks a month and while i realize it takes me longer to recoup the 2nd box cost, in the long run i make out by saving 10 more a month over premiere with just one box but am i missing out on premier features i may want?
all opinions appreciated. thanks!
my plan was get one box and just test it with the given number and then go from there.
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getting 401 with obi110 as fxo for freepbx
Hello Guys
I am getting 401 with obi100 thats connecting to my asterisk box and I am trying to use this as fxo setup and only thing I have setup inside of my freepbx as trunks to this obi100 is there something else I am doing wrong here?
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[General] two way txtmsg from PC
Is there an app, like instant messenger, that a person can sit at a Pc and txtmsg back and forth with cellular phone's? This would be for a receptionist to respond and contact customers on their cell in real time....
Thanks for any help!
brian
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Obihai OBi200/202/302 + OBi1022/1032/1062 + OBi50x firmware mods
So I want to add the ability to configure these devices for GV using oauth without obitalk, similar to the changes for the obi100 (and add an ssh server, for grins).
I think I have the MD5s in the firmware file worked out (its the same "Goodbye! Reboot Now" garbage as the 100), and I see where the oauth refresh token code is, so it should be pretty straightforward unless there is code signing that I missed.
The only hiccup is... I don't actually have an OBi20x :-(
Anyone have one of these devices that wants to be a guinea pig? You should definitely have a way to SPI the flash back *when* i brick the thing the first couple tries...
[or if someone has one sitting in the closet, you could just send it to me. ill name the fw after you :-)]
EDIT: speaking of flash, its supposed to have a w25x128 on board, but is it the SOIC package or some BGA madness?
QUICK SUMMARY:
Custom firmware made for all obi devices, thanks to the help of generous hardware donations and bold testers.
See obifirmware.com to download latest.
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[PBX] FreePBX: Removing "CID:YOUR_OUTBOUND_CID" from phone display
If, on your outbound calls your phone displays
ACID:<your_outbound_CID>and you would rather that it didn't, the solution is simple.
In FreePBX, under Settings->Advanced Settings, in the section named Dialplan and Operational, change the following setting from Yes (default) to No:
Display CallerID on Calling Phone : No
Source:
https://pbxinaflash.com/community/threads/yealink-phone-displays-our-trunk-instead-of-called.14730/post-94793
That post is for PIAF 3 (2014), so this should apply to a lot of FreePBX versions.
Note: I can see the value of displaying the actual outbound CID in an environment where different trunks and/or extensions have different CIDs. My PBX has the same outbound CID for all trunks and extensions, and I found that "CID:YOUR_OUTBOUND_CID" on the phone's display to be annoying.
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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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[Equipment] Hook OBi up to phone jacks
Hello,
I am a lurker who decided to finally post. I’ve recently gone to OBi as a way to get landline service at home (in addition to cell service). I would like to plug the OBi directly into a phone jack which, from what I’ve read, would then activate a dial tone on all jacks, provided there is no POTS connection coming into the house.
I went outside to check for the demarc box/NID (still not quite sure about the difference) and found the small box in the attached photos. There is a larger box that is bolted shut in my basement directly under where that box is located. My question is: since it appears that there is no connection in that outside box, does that mean that the POTS service has been cut and I am safe to plug in the OBi? Or should I open up the box in my basement (from which the Internet cables emerge to connect to my modem/router)? When I plug a phone into the jack there is no dial tone, only a sound sort of like a soft wind blowing. Sorry in advance for the noobness.
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[Voip.ms] Canadian Caller ID issues with VoIP.ms
A user on another forum reported that when he used VoIP.ms to make outgoing calls, sometimes the recipient's phone's Caller ID displays Peru because of his 519 area code (Peru country code is 51).
I have tested the value route and found the following:
Caller ID set to npanxxxxxx via the portal:
Some calls arrived on my TELUS cell phone with Malaysia Caller ID.
Caller ID set to 1npanxxxxxx via my PBX: (not possible via the portal)
Caller ID always arrived correctly, but sometimes my cell phone gave a half ring and then disconnected the call.
Caller ID set to npanxxxxxx via my PBX:
Always worked properly.
Since I have no way of determining what route my calls are taking, I don't know if my observations are valid or coincidental. Based purely on the above, it seems that there is a hidden difference between "I use a system capable of passing its own CallerID" and setting Caller ID via the portal, even if the Caller ID number is identical.
Has anyone else seen similar results and received a response from VoIP.ms that's something other than blaming the recipient's carrier?
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[Other] New SIP client for Android
Hi!
We're the developers of a new VoIP app, Sipnetic. If you use an Android SIP client, such as CSipSimple, Zoiper, or Linphone, you should check out Sipnetic. I'll be glad to provide any info about this app or SIP protocol in general.
link removed by moderator
Mod note: search Google Play Store for SIPNETIC
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Cheapest place other than VoIP.ms to park a toll-free DID?
I'm paying roughly $4 a month to park an unused toll-free number / DID. Any other providers where it may be cheaper?
I'm not a fan of the Walmart of VoIP, VoIP.ms.
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NumberBarn's fee for high-value DIDs went from 10% to 25%
As of December, NumberBarn's commissions were:
• 30 percent for sales of up to $19,999
• 10 percent for sales of $20,000 or above
Now the commission is a flat 25 percent for all sales, per:
www.numberbarn.com/brokerage
The only similar business that I know of is:
www.phonenumberguy.com/store
Are there other well-run websites that sell vanity DIDs?
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[Voip.ms] VoIP.ms introduces Elastic SIP Trunking
From the email:
VoIP.ms is now introducing Burstable Virtual PRI also known as Elastic SIP Trunking for greater flexibility!
For a while now, VoIP.ms has offered the possibility to have a Virtual PRI (VPRI) for unlimited inbound calls (non-metered) with as many concurrent calls (channels) as your business needs. Depending on your business needs, you may allow all your DIDs to share a pool of channels directly in your VPRI and the number of channels can be adjusted according to your needs. We also offer this service for several countries around the world.
The new burstable feature allows you to use more channels than your actual limit ensuring you to receive every call. We also implemented a hard limit that you may choose according to your needs.
To ease your understanding of this new feature, here is a quick example:
Say that you originally had a 50 channels VPRI with VoIP.ms, if at some point you received more than 50 calls at the same time, the caller would get a busy signal. To avoid this situation, with the addition of this new feature, you could decide to allow your VPRI to burst until 100 calls, and you would simply get charged a small premium for the additional channels used over your original limit.
Unlike most Burstable Virtual PRI product offering in the industry, we kept in mind our pay-as-you-go and hassle-free approach. Therefore, we charge on a pay-per-use model where if you use 10 extra channels only for 1 day in a given month, we won’t charge you the channels for the entire month, but rather only for this specific busy day.
As always, this feature is available to all our customers without any contract. For more information, you can read our Wiki article here.
https://wiki.voip.ms/article/Virtual_PRI
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Cheap Cellphones
Are there any current deals on Cheap Cellphones not requiring a plan to be used as a WiFi phone?
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[General] New user to voip
Hello,
I have a several questions regarding VoIP service. I'm looking to register 2 numbers. I've been looking at Callcentric, they look like a very good option for me. So with that said I just have questions below if someone doesn't mind answering.
How do these companies obtain these numbers? Are they legit numbers? Can I transfer them if need be?
I also see services such as Burnerapp where you rent a number, to hide your real number.
Can I spoof my number with a service such as Callcentric, so I don't need to rely on services such as Burnerapp?
Thanks.
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[General] Tuning network for realtime communications
Now with two adults at home doing high-def video meetings (plus other ordinary Internet traffic) all day long, I found I had to re-tune our home network a bit so that Zoom meetings don't stutter.
In almost every case the bottleneck is going to be on egress from the home. I'm happy to have a plan with 10Mbps uplink, but with two video meetings going, each one consuming 2Mbps up, and then cloud backups running hourly, you get hiccups.
My opinion is that it is very much worth sacrificing some speed in order to ensure latency and "buffer bloat" are at their lowest. A great tool for tuning that is right here on this site - choose the Speed Test on the left hand menu.
I have a Ubiquiti Edgerouter at home and used the QoS - Smart Queue feature, gradually lowering the numbers until my pings and buffer bloat numbers from the DSLR speed test were < 10ms.
Another good resource is Netflix's fast.com. Expose the settings on that site and have it show all the metrics as well as latency on both download and upload. Tuning for lowest ping/latency numbers is going to provide the best VoIP and Video-OIP results - don't worry about losing a few Mbps of throughput.
Speedtest.net and the bandwidth tests from the ISP are fairly useless and inconsistent. I would not waste any time with them.
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