With today's (well-discussed) close of Ring.to I moved most of my RT numbers to Anveo -- both consumer and Anveo Direct.
Ring.to had a good SMS text messaging component. But now I'm looking for an easy way to send (and receive, though I have "receive" forwarding set up) text messages from those numbers for the very rare occasions I would need to (rare enough I don't want to make a big investment of time and effort to do so).
Textable was brought up. https://textable.co/ Plug in your API info for Anveo, Twilio, Flowroute, voip.ms, or Bandwidth.com (Telnyx and Nexmo "coming soon") and it seems to work well with only maybe 10 seconds of lag. But it currently only works with one provider/one number per Texable account.
I use Acrobits softphone app for iPhone, and https://smsglue.com/ provides some insight on tweaking Acrobits to handle texting for voip.ms, thus making it a texting app. I assume (perhaps wrongly) the same could be done for my Anveo numbers as well as the couple I have Twilo, Tropo and Voxbeam.
I want something easy from my phone that doesn't require logging via a browser into anveo's or others' dashboard to send a text message. And while it might be an interesting intellectual exercise to figure, I know I don't have the programming experience/patience/time combo to write my own program.
Does anyone else have suggestions for quick and easy apps (or tweaks of apps or other work-arounds) to make texting work easily for multiple existing VOIP numbers/accounts at the low bar, of say, texting on par with the quality of Google Voice, and without porting them to something else?
"Off the shelf" VOIP (anveo, twilio, etc) sms iPhone/Android app
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[PBX] Vitelity and 3CX - Busy Signal on inbound
Thought I would share this here - nowhere else do videos or instructions tell you when setting up a 3CX and Vitality (or possibly any sip trunks) that in the DID number field, you must add the +1 or the sip trunk registration won't recognize the call coming in. I hit my head against the wall for a week on this, and found a way to turn on extensive logging and it showed that the numbers didn't match (as the instructions on Vitelity's site and youtube and 3CX all say just put in the area code and phone number). As soon as I did this, I was able to receive inbound calls.
The more you know...
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[Unlock] Unlocking the BasicTalk ATA
Important NOTE: a better unlocking method has been posted later in this thread. Please see this post for a permanent unlock developed by uid://1479488. My soft unlock may help in some cases if the ATA has "called home".
I have some good news for those of you looking for an inexpensive ATA.
I've just got my hands yesterday on a couple of BasicTalk ATAs (I've had my eyes on them for a few months but I live in Canada and don't go to US that often) and I put together a small tutorial for unlocking them.
The ATA is a Grandstream HT701 with a customized firmware.
I posted it on my website at http://voipfan.net/unlock/ht701bt.php
I will leave the access open to everyone for a couple months then make it available to registered users (like my other unlocking tutorials).
Enjoy and if you run into any trouble please post here.
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Providers (through asterisk): voip.ms, freephoneline, smartcall.ro, ipcomms, callcentric. Hardware: Vonage VDV21, Moto VT2x42, Linksys SPA series, Grandstream HT series, Panasonic KX-TGP5x0
http://www.voipfan.net
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Nextiva faxbridge works well - $4.95 per month
A few weeks ago, I bought a $150 faxbridge (model FXB21) from Nextiva, for use with their vFax service. The device is almost unmentioned on their website. The manufacturer is Axacore, which doesn't mention it on their website.
I thought that the faxbridge was an analog telephone adapter optimized for faxing over VOIP, but it's not. The faxbridge connects to a standard fax machine. When you fax out, the faxbridge accepts the entire fax, then converts it to some digital format and uploads the converted fax to Nextiva's server, which then faxes it to the destination number that your machine dialed.
One clever aspect here is that no matter which number your fax machine dials, the faxbridge "answers" the call with fax tones and takes the fax. If the number your fax machine dials is not a fax number, you won't learn that until a few minutes later, when you get an error message from Nextiva, saying that your fax destination can't accept a fax. (Whereas on a genuine fax machine connected to the PSTN, you'd catch the error immediately upon dialing and getting something other than fax tones.)
Faxes sent to your Factiva fax number are routed in reverse: Factiva's servers receive the fax, then convert it to a digital format and download that to the faxbridge, which then "calls" your fax machine, which then prints the fax.
Functionally, this resembles faxing over POTS lines, which is essentially flawless (whereas faxing over VOIP lines works about 90 percent of the time, at best).
Faxtiva vFax service costs $4.95 per month for up to 500 pages.
I still have a number of small business clients who resist giving up their POTS fax lines because these lines are so reliable. The faxbridge might be the alternative, finally. (Also flawless is faxing over Comcast's telephone lines, but they're kind of pricey. Other cable ISPs may have similarly reliable phone lines for faxing).
By the way, Nextiva is a mildly dysfunctional company. Their website, customer service, and tech support are all a little goofy. Nextiva is owned by Voxee, a Singaporean company, and I suspect that the managers of Nextiva haven't bridged certain cultural gaps, resulting in the oddness that I've noticed (which is not a deal-breaker).
Has any of you used the faxbridge that I'm talking about?
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Twilio phone number lookup tool has been changed
The excellent free tool (to lookup the carrier and sometimes the CNAM associated with a phone number) has been tightened up in the last couple of days.
https://www.twilio.com/lookup
Now, detailed registration is required, and must be activated by receiving a text or a voice message to a cellphone. VoIP phones don't work for the confirmation process btw.
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[Equipment] Pogoplug Mobile 2-pack for $14.98 free shipping at Adorama
In case anyone is interested. https://www.adorama.com/cocpogov4a12.html
I got a used one from a friend and have a lot of fun with it - so much so I just placed an order for a 2-pack.
(edited) I should have mentioned that SSH to the stock OS is disabled. You have to open it up and use the serial connection to get started.
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Will VOIP providers offer RCS like they have SMS and MMS?
An interesting read on RCS and the future of messaging.
https://www.theverge.com/2018/4/19/17252486/google-android-messages-chat-rcs-anil-sabharwal-imessage-texting
Does anyone think VOIP providers will one day follow wireless carriers' lead again into messaging's next evolution, assuming it materializes for wireless carriers?
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Analog phone over ethernet adapter
Situation. Have AT&T DSL to a PACE modem. with 1 dial tone from it in a small office at front of property. Main house sits several hundred feet back and is connected via a point to point wifi bridge. That works great and get the full 6M from the DSL at the house.
Problem. trying to get the analog dialtone bridged up to the house. We tried Grandstream HT 502/503 After hours of configing I was only able to get calls out. The incoming caller would hear 1 ring then dialtone. Home phone did not respond.
Grandstream support sucks. Only done thru email.
So searching for plug and play FXO/FXS devices. NO fancy protocols are needed. just trying to bridge basic dial tone over ethernet.
Before it is asked. Running cable to the home is not an option. That's why we are using the wifi bridge.
TIA move to better forum if needed.
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Shared line on voip.ms
Is there a way to do shared line appearance using the voip.ms cloud service? I need to have incoming calls ring on four phones. When a call is answered I need to be able to put it on hold and pick up on a different phone. Each DID will have two appearances on each phone. The phones are Yealink T48S. Thanks.
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[Anveo] Anveo Voicemail Problem
Over the past few days, the "play button" for my Anveo emails is no longer visible for new voicemails appearing in my dashboard/inbox. It continues to appear for older voicemails that were present prior to the problem surfacing. Is anyone else having this issue? The problem doesn't appear to be browser-specific.
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[Asterisk] SIP Trapezoid Question
The famous SIP Trapezoid: When Alice wants to call Bob, Alice’s phone contacts her proxy server, and the proxy tries to find Bob (often connecting through his proxy). Once the phones have started the call, they communicate directly with each other (if possible), so that the data doesn’t have to tie up the resources of the proxy. https://www.safaribooksonline.com/library/view/asterisk-the-future/0596009623/tagoreillycom20070216oreillyimages62724.png.jpg
I've a few Asterisk users, mostly Cellphones out in the wild using their respective Data plans. Their RTP traffic seems to go through my Asterisk Server, forming a Star topology. I wish they would just talk to each other directly.
Is there a way my users' RTP media can follow the shortest straight path between them w/o traversing through me, given that they are all my users?
Thanks!
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Porting out of TextNow App
They told me I need to ask the carrier to email them at porting@textnow.com. The problem is that the receiving carrier has said I need to supply and account number, address and LOA as this is company policy.
After going back to textnow they have said according to the FCC law they must cooperate and email them as I expressed interest in porting my number over to them.
Has anyone here ported out of TextNow a number that they have in the App?
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[Equipment] Configure Digium D70 via web
Maybe it is my old brain, and from what I see it couldn't be simpler, but trying to manually put SIP settings into a factory reset Digium D70 and the web page shows everything twice for Line1 and then insists that the second instance of Line1 settings need to be filled in. Even if you click away to Line2 as the second part of the setup, it still is doesn't quite take. Reading the docs says it couldn't be simpler but why it has two setup instances for Line1 on the screen is puzzling. Any tips? (I've done Polycom/Cisco SPA/Yealink/Mitel without this much effort).
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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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[Anveo] Anveo hardware failure
Service Alert! Due to hardware failure we are experiencing issue with incoming and outgoing calls. Our engineering team is actively working on the resolution.
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[Anveo] Inconsistent Porting Requirements
Is there an Anveo representative that views these forums?
In late March I had ported over what I thought were all my numbers from RingTo. I had forgotten about one, and thankfully was able to have RingTo's lead developer (who I got to know during my service as a beta tester) retain the number for me despite RingTo shutting down and Bandwidth de-assigning numbers. He arranged for me to have SMS forwarded to me so I could authenticate ownership.
While Anveo's support is nice enough to allow me to port this number in at no charge, they keep stating however that, in addition to filing an LOA (which I'm happy to do) I have to "prove ownership with a screenshot of my account" which I obviously can't do now that RingTo has shut down. I have explained the predicament and they are unsympathetic on this last part.
What I don't understand is why SMS authentication - which was sufficient for this purpose in March (and still seems to be the case for the other competing services for which they offer free porting) - is suddenly insufficient now?
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[General] Can't reach number from VZ POTS line but OK on Wireless & Others
Just got a virtual voip number for handling clients in NYC from Localphone (Voxbeam/Magic Telecom). It works fine calling from Verizon Wireless, AT&T, Comcast, Sprint, TMo, and Google Voice. Trying to call it from my Verizon POTS home phone it is silent after dialing and then at about the 30-second mark, rapid busy. No error recordings or anything. I haven't been able to try from another Verizon landline.
Before I hit up Verizon, anyone know what what is happening, and what to ask them to do? The number is in NPA NXX 646-503.
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VOIP.ms Alternative (Canadian Preferred) ?
So I have been with voip.ms for many years, as long as I can remember, but I think it is time to move on.
I changed my email address probably 6 months ago, nothing unusual.
Anyway, I have just spent the last 3 days or more with their tech support trying to reset my password with no success. Every time I try change my password it rejects the code they sent me, saying the code does not match the email address. I suspect my old email is in the system somewhere and there is a bug in their system somewhere which they not willing to acknowledge, so much easier blaming the client.
The solution they proposed now is to manually set my password putting the blame on me for unsuccessfully using the broken system that they claim is working.
They now require me to send a photo ID from my email account proving who I am for security reasons. it was bad enough I had to do this when signing up, but seriously to get my password to work ?
This still won't fix the root problem and wreaks of incompetence on their end.
Unfortunately, the support team seems incompetent or not skilled enough to fix an obvious problem with their system but has turned to blaming me. I have loads of gmail acounts, 2 factor accounts across a slew of systems but never run into a system like this that thinks they need to be securer than fort knox.
I am hoping to keep it Canadian, but there aren't many Canadian worthy providers (I might be wrong), is there someone worthy to look at ?
I have used CallCentric and didn't mind their system and somewhat recall it been easy to use.
I don't think I want to use Avneo due to lack of any support, I heard there is paid support.
Looking for a simple easy but reliable system where I won't lose my phone number due to incompetence or the site shutting down.
Don't need fancy features other than blocking spammer calls.
Thank you
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[Anveo] As of 04/14/18 im not receiving caller ID info
The caller ID info is no longer being sent to my PBX.
The problem started on 04/14/2018.
I have since logged into my anveodirect account and downloaded some of the sip trace call log files.
The files that I downloaded with the date of 04/13/18 or older, have the the proper Caller ID info in the From: "CALLER NAME " sip:1XXXXXXXXXX@XXX.XXX.XXX.XXX ....
The files that I downloaded with the date of 04/14/18 or newer, does not have the CALLER NAME in the From.
It just says From: sip:1XXXXXXXXXX@XXX.XXX.XXX.XXX
I have had anveodirect for years, and this is my 1st time having this issue.. I don't think it is anything on my side that I can do to correct this.
What would cause this change?
Thoughts?
Thanks.
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Voip.ms
I have been having a problem with call groups with voip.ms. this is the function in which several devices in a group ring at the same time.
It just came to my attention that the configuration of my voip.ms portal has been changed. I believe the change was made by voip.ms without my permission. I have placed a request for email notification for portal access several months ago and nothing was done. Since voip.ms uses https and I have had other issues in which voip.ms in which they credited me $10 per failure incident I will be pursuing this issue.
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