We've talked here before about Voip customers using MaxEmail for sending and receiving faxes.
MaxEmail was a good service, reasonably priced.
All of a sudden the company was bought by j2 Global®, who already own eFax® and various other brands. :(
Some people choose not to do business with j2 Global® due to various policies.
https://www.dslreports.com/forum/r26580959-MyFax-eFax-500-if-you-port-out-AND-we-ll-take-it-back
Existing customers should be receiving an e-mail about the change, which includes new rate plans and new TOS.
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MaxEmail Fax Service taken over by j2 Global®
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[Unlock] Unlocking an Linksys SPA2102?
Well, I've finally decided to switch from POTS to VoIP, hopefully this month! All I need is an ATA box...
I do have fully working Linksys SPA2102 I'd like to use, but the problem is its locked. :hmm: In this other forum thread, with the help of a DSLR user, we were able to determine the ISP it was formerly provisioned and locked down to, but didn't get very far in trying to actually unlock it. I'm hoping someone here has experience with that and may be able to help me.
I should mention I also have a second Linksys SPA2102 and RTP300, both of which are fully unlocked and ready to go. Only problem is there appears to be something wrong with both. The 2102 has a constant static/hissing that never goes away, and the 300 intermittently makes clicking and crackling noises after being powered on for 10-15 minutes. I'm assuming they both have bad capacitors or filtering circuity so I've given up on those for now (unless there's a firmware setting that could resolve their noise issue?).
I can access the firmware and phone voice-prompt menus on the locked SPA2102, I just cannot get into the admin settings to provision it (it's password locked). Looking forward to any help here! :)
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Once again I am reminded of why I dislike "Incredible" PBX
I have been wanting to try to get a Google Voice oAuth-capable PBX running on a Raspberry Pi here, since it's currently not doing anything else useful. So today I thought I'd try installing Incredible PBX (not the XiVO version, I've looked at XiVO and I'm not that much of a pioneer, and besides I don't have a Raspberry Pi 3 here), just because it's been so long since I've tried it I thought maybe the things I most objected to would have been removed.
So I went to this page: https://sourceforge.net/projects/pbxinaflash/files/IncrediblePBX13-Raspbian8%20for%20Raspberry%20Pi%202%2B3/
I followed the instructions on that page, and it booted up and I was able to get it up and running after a couple of false starts (entirely my fault so I won't dwell on that) but I was very disappointed to find that despite the fact that Ward has said on at least a couple of occasions that there's now a menu in Incredible PBX where you can pick which "incredible" features you'd like installed, they were all installed by default. There is an entire extensions_custom.conf full of stuff I REALLY don't want. Nobody would ever use any of those things here, and I have my own extensions_custom.conf I use.
So before I go any further with this, I just want to know, if I blow away the existing extensions_custom.conf and substitute my own, will it leave it alone when I allow the upgrade script to run at reboot, or will it overwrite it with the default "incredible" extensions_custom.conf each time?
The other thing that ticks me off it it implemented some "security" features that I personally think makes the system more insecure. For example it set up some "Knock ports for access" which I don't need or want. Also, it says "To restart IPtables, ONLY USE THE FOLLOWING COMMAND: iptables-restart" - does that mean I can't restart it from Webmin like I always do? Is this thing going to make my life hell when I try to use it with an offsite extension? And also, by default iptables allows SIP connections from a bunch of IP addresses, but there are no comments to tell you who they are. I assume some are providers I will never use, so why would I want to allow them access?
I went into Webmin and it said it needed an upgrade, I clicked the upgrade button and it download the new version but then something went wrong during the install - it says:
Running DPKG to upgrade Webmin .. Please wait a minute until it is complete before continuing.
(Reading database ... 75892 files and directories currently installed.)
Preparing to unpack .../6780_1800_2_upgrade.cgi ...
Unpacking webmin (1.810) over (1.780) ...
And there it's hung. Never had that happen before. Rebooted and tried again, same result. This may be a trivial thing but it's an example of the type of little annoyances I've found every time I've ever tried to use any version of PBX in a Flash or (now) "Incredible" PBX. Webmin upgrades have never given me a problem on any other system I've used, and I have installed it on just about every Linux server I've ever owned or used.
Believe me when I say that if RasPBX included oAuth support for Google Voice I wouldn't even be trying this. What I'm really kind of P.O.'ed about is that Ward has said on at least a couple of occasions that you can now select the features you want from a menu, and that was really the only reason I was willing to give this a shot. Well, unless there's a newer version of "Incredible" PBX for the Raspberry Pi hidden out there somewhere that has that menu, it hasn't found its way into the Raspberry Pi version yet. I am already getting the sinking feeling that I'm wasting my time, and I haven't even really done much beyond install it and look around a bit.
I could just forget about oAuth (after all, Google Voice still doesn't require its use) but if I'm going to do that I might as well install a newer version of RasPBX than the one I'm running on an original Raspberry Pi. Other than the Web interface being a bit sluggish, it's never given me a bit of trouble.
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Observations using g.722 with CallCentric and Anveo.
Since I have a Snom M700 that was doing nothing for a year and there was recent firmware upgrades, I decided to set it up with my second CallCentric account which sat there for years with a small balance and start playing.
I noticed that in order to even get HD (g.722) within CallCentric, one must use the 777 number, not the DID if any. The calls were between a Panasonic TX-TGP600 with a UDT121 handset and a Snom M700 with an M65 handset. I added a free NY DID to the second account and set up five extensions on the second account since the Snom M700 unlike the M300, can only have one handset per sip account as it were. Among the five M64 handsets, by calling extension to extension eg 100 to 101, it also is in HD.
In any CallCentric account it is easy to add a sip uri to the phonebook so calling wbdemo@conf.zipdx.com from any CallCentric account also works in HD so one would see that a sip url will render HD as well outside of CallCentric. Now with an Anveo account which has a paid plan and is supposed to support HD, I set up a sip url in my Anveo account in the IVR (no, it is NOT your did@sip.anveo.com) and set it up in my CallCentric phonebook, while it works as far as making a SIP call, it does NOT happen in HD. In case anyone wonders, using the sip uri inward to Anveo uses platform minutes. I have not yet figured out how to set up a phone book in Anveo that takes sip uri's rather than just numbers. Huge learning curve as I am new to Anveo.
Any thoughts?
PS: I notice that the CallCentric directory lists the 777 numbers which means that anyone capable should be able to enjoy HD that way.
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[Future9] No incoming calls for a week.
No incoming calls for a week, no response to support ticked. Nitzan, please check PM from me with account number. Or, should I post account number here?
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[Apps] Voip.ms Digital Receptionist/Auto-Attendant with POTS
Hello,
I have a Voip.ms line, and a POTS line from way back.
I'm wondering if there's a easy way to forward my POTS calls to the Voip.ms digital receptionist/Auto-Attendant, in order to filter out telemarketing calls and then forward back to the POTS line.
I have a Linksys SPA3102 that can handle both POTS and Voip calls, but I haven't really explored its functionality much, because I chiefly use it as an ATA, but I could easily mount it at the DEMARC and have all my calls go through it with relative ease.
Thoughts?
edited to reflect proper name of the feature
EQ
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VoIPtalk admits to possible data breach
VoIPtalk says that user VoIP and SIP passwords may have been compromised after the company's staff uncovered suspicious activity and external online attempts to "exploit vulnerabilities in our infrastructure" in the quest for customer data.
The company, which provides VoIP services to both businesses and individuals in the UK and Europe, is investigating the incident.
However, VoIPtalk has decided to work on the assumption that user accounts have been exposed, although there are no reports at the time of writing that information belonging to users has been used fraudulently.
http://www.zdnet.com/article/voiptalk-admits-to-possible-data-breach/
Similarly:
http://www.theregister.co.uk/2016/09/13/voiptalk_suspected_breach/
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From cord-boards to VoIP, the stars still get top-notch service
http://www.nytimes.com/2016/09/04/nyregion/bells-are-still-ringing-if-less-often.html
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[Anveo] Is video calling a VoIP topic?
said by uid://1710150 :Calls to 1555XXXXXXXXXX are forwarded directly to your device (without proxying audio/media) and thus supports any codec your device is configured with including Video call.
I don't recall video calling being discussed on this forum... now I'm wondering which ITSPs support it; what are the pros and cons; and how does implementation and cost compare to voip. Anyone?
OE
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[General] what can I do with a BasicTalk VOIP box? (ie google voice?)
Hello, just to explain i'm totally new to google voice (know of it, have yet to use it/dont have a gmail or anything yet), somewhat new to VOIP (have used it but don't "understand" it), and mistakenly thought BasicTalk was an OBiTalk which i'm told can directly configure to plug into Google Voice.
I bought a BasicTalk box used for $5 from a thrift store and I have two days to return it if I cant do anything useful with it. :)
I saw some site suggesting it's possible to reset the box to zero/"as shipped", which then supposedly lets you put a general firmware on to recognize as a Grandstream HT701 - which can be reconfigured to work with general SIP where i'm not fully sure what that is actually. I was hoping it could directly connect to google voice (but am not even sure if OBitalk still does? I saw confusing articles) but if not, it sounded like there was some other Simon Telephonics service which enables a SIP gateway to googlevoice with some one time fee.
What I dont understand is how this works, if there are alternatives, if this is something that can "go away" later (if the company goes away do I lose the Google Voice number entirely, or just the ability to use the Grandstream VOIP box), if the features it supposedly adds to google voice (it says incoming caller ID but i thought it already had that?, and incoming fax to email) can be added in other free ways, etc etc. in short even if I got an OBiTalk maybe i'd want to add these features anyways - but I don't know if that uses SIP itself either.
I have alot to learn and don't know where to start.
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features Google Voice dont have but can add?(or other VOIPs have)
Hi again, i'm trying to learn more about some of my landline/VOIP/forwarding options whether for personal use or potential future small business use for the impoverished.
Google Voice seems like a win because apparently most of the calls are free(? i dont have it yet and see conflicting reports that some places charge). I am curious if there are features it should have but doesn't - for instance my roommate got an OOMA but unless you pay for the premium service you miss critical features like the ability to block harassing callers(!) and if the hardware box ever dies or wears out even if by a lightning strike or power surge you have to buy a whole new service box/package, you can't just 'port' to a different box.
I'm aware of other providers like Magic Jack which is cheap but only has a point if maybe it accesses area codes google voice doesnt for instance. Or if you have to do alot of sales or conference calls (like over thousands of minutes a month, even if not unsolicited) it might make sense to take some load off google voice and have a second cheap system to run up other minutes on. I don't like that MJ apparently doesn't like to let you port out numbers and sometimes engages in randomly abusive practices which only matters if theyre sitting on a phone number of yours and you cant get it back.
I stumbled across a place called Simonics which lets you use general SIP phones through their gateway to work on google voice for a one time setup fee - they say they add some features on top of google voice. (including incoming caller ID though I thought that was already supported) Which makes me curious if there are other services (esp one time ones) or programs or hardware or whatever out there which lets me add features GV may not have but should or could. (like maybe some of the fancier PBX abilities where it could use multiple outgoing VOIP providers like GV and magic jack and Ooma at the same time and track minutes of usage on each to not overuse anyones service)
I'm also curious what else is out there/things I don't know of whether pay (preferably one time unless a really super good feature) or free, hardware or software, useful strategies to apply (like best ways to use E911 or faxing services or what) and more.
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recommended cheapest ATA (SIP and nonSIP) hardware?
Since i'm still in the process of learning...
I'm trying to learn what is the least expensive available Analog Telephone Adapter hardware out there for VOIP use. I'm also trying to understand what nonSIP standards exist out there. (apparently Google Voice and Skype are two of them/not compatible with anything, without some kind of gateway anyways... at least for GV)
I'm aware I can use softphones on computers and wifi softphones on android handsets/tablets as well but i'm specifically wanting to use normal office desk phones and cordless units instead because wifi gives me bad headaches and I don't always want to have a computer on.
I'm aware of the Grandstream HT701 for SIP use (and the hacked Basic Talk if I can get it working but mines probably a brick)
I'm aware of the OBiTalk models (but dont know the differences between different models, and heard something about a 'backdoor' that concerned me) which can work directly with Google Voice and apparently other SIP clients? (what about skype, does anything plug LAN to phone for that?)
I am curious what else is out there especially at the cheaper end of the spectrum since i'd like to set up like a dozen to twenty max physical seats or so on a total shoestring. (already have the phones just need the ATA's) Super low budget because it's going to be so rarely used (its more of a preparation for a near future use that may not even occur) and I dont have much $ to put into it on a college student budget so a cost of $5 instead of $50 matters when it multiplies by up to x20. :P
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[General] VOIP.ms same product but at lowest price!? What Provider ?
Hello !
I have a trouble when I try to fight with many other company how sell voip phone service.
Lot of that have a REAL unlimited incoming and outgoing but me with voip.ms the incoming is 3500 minutes and outgoing is at 0,01$/min.
Do you know a provider like voip.ms with real unlimited incoming and outgoing or lower price provider with same features ?
thanks in advance!
nicK
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unlocking a BasicTalk box that's called home?
There's a few years old long thread that i'm scanning through but i'm wondering if there's info not in that thread or what the latest info is...
I have a box I got from the thrift store for $5 that I can still return by friday evening if it's not worth even $5. :^) It doesn't respond to the 'factory password' so it obviously was used by someone else first.
Is what i'm looking to do a "factory reset" or something else? I would like to permanently unlock this box to use with something else entirely/if possible return it to the "as bought" state before it phoned home and changed anything, so I can permanently unlock it.
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[Asterisk] OAuth 2.0 Support for Asterisk 13
The attached package should provide OAuth 2.0 supoort to any Asterisk 13 system:
1. Extract oauth2.tar.gz to the /root directory.
2. Edit oauth2.creds. The first line must contain your Oauth 2.0 Client ID and the second line must contain your OAuth 2.0 Client Secret.
3. Make oauth2 executable: chmod +x oauth2
4. Execute oauth2: ./oauth2
5. Use your OAuth 2.0 refresh token(s) as the Password(s) in FreePBX's Motif module (or as the secret(s) in xmpp.conf of plain Asterisk).
6. After clicking Apply Config in FreePBX, wait 15 seconds after the Reloading dialog box disappears before doing anything else.
.
If you don't already have an OAuth 2.0 Client ID, Client Secret, and refresh token(s):
1. Go to Google Developer Console: https://console.developers.google.com/project
2. Log in with your Google Voice username/password
3. Click CREATE PROJECT
4. Enter a Project name
5. Click Create
6. In the left pane, click Credentials
7. Click OAuth consent screen
8. Enter a Product name shown to users
9. Click Save
10. Click Create credentials
11. Click OAuth client ID
12. Select Web application
13. Enter a Name
14. Enter https://developers.google.com/oauthplayground at Authorized redirect URIs
15. Click Create
16. Record Client ID and Client secret
17. Go to https://developers.google.com/oauthplayground
18. Click the gear icon
19. Check Use your own OAuth credentials
20. Enter OAuth Client ID and OAuth Client secret
21. Click Close
22. Enter https://www.googleapis.com/auth/googletalk at Input your own scopes
23. Click Authorize API
24. Click Allow
25. Click Exchange authorization code for tokens
26. Reopen Step 2
27. Record Refresh token
To create a refresh token for additional Google Voice accounts, log out, log in to the desired account, and go to step 17.
.
Credit to Ryan Tilton, dziny, carlb8, phonesimon, and others for the original res_xmpp.c modifications.
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[Voip.ms] VOIP.MS accounts having random, brief outages on Grandst GXP2140
I use VOIP.MS with a Grandstream GXP2140 4-line desk phone. On
its screen, each line normally shows green when not in use. The
firmware is 1.0.7.15.
For the past two weeks or so, at times when I'm not using the phone,
each of the lines goes out for roughly 15 to 30 seconds, one at a time,
randomly. During an outage, the line in question goes red. The vast
majority of the time, all lines are green.
During such an outage, on the VOIP.MS control panel, the sub-account
status shows as unregistered, and when I attempt to get a dial tone on
a red line, I can't. These facts suggest that it's a real outage. (However,
I have no basis yet for concluding that VOIP.MS is to blame.)
My lines are split between SANJOSE2.VOIP.MS and DALLAS.VOIP.MS,
so the particular server doesn't seem to matter. I now run ping on
both of those servers 24/7, and the ping is flawless, even during an
outage.
To troubleshoot, I've installed a second GXP2140 next to the original,
registered to a different set of VOIP.MS sub-accounts on those same
two servers, and the same thing happens randomly on the second
GXP2140.
I haven't changed any network hardware lately. I'm not using QOS
on my router (ASUS RT-N12, hardware version D1).
There are other things that I could try, like swapping in a different
router or modem, or testing my setup from a different ISP. But at this
point, it's easier to find out first from you guys whether you're seeing
anything similar.
Have you seen anything similar on your VOIP.MS setup lately?
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[Future9] New feature - blacklisting.
I know this has been long overdue, but it is finally available: phone number blacklisting (blocking).
Currently supported:
Blocking based on full number (no wildcards)
Either send to voicemail, or reject the call
Planned to be supported soon:
Wildcards
Blocking only on certain dates, days, or times
Other actions than voicemail/reject (suggestions welcome)
To use:
Click on the BLACKLIST link inside the web console.
Any comments or suggestions welcome. Please keep in mind this is still in beta and a work-in-progress, so if you find any issues please let me know. Thanks!
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[General] E911 Over VoIP
Just wanted to say I had to use it for the first time (everything's ok) and it worked flawlessly. It was at home so it was over my Obi100 routed through VoIP.ms. The call rang twice and then the county PSAP answered. Didn't get to verify if they had the correct address because I wasn't the one that spoke to them I just dialed, but there was no issue.
This probably wasn't worthy of it's own thread but I couldn't find any fairly recent E911 threads.
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[Equipment] VoIP Phone + Bluetooth Cell Connection Support
I'm looking for options / recommendations for IP phones that in addition to multiple SIP accounts, support pairing with a Bluetooth phone to make and receive calls from the phone over the cellular connection. So far, I've only found the OBi business phones, (1022, 1032 & 1062) some of these require the purchase of a separate dongle that plugs in to the USB port and possibly the Cisco SPA525G2. I'm still trying to confirm the feature on the SPA525G2 and if it needs additional hardware. Any suggestions?
Thanks!
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differences between OBi devices?
So I see a bunch of different model numbers, and have in my notes someone talked of an OBitalk 202 backdoor (or 2-something model) to be wary of... and someone mentioned there's regular sales (but i'm wondering when the next one would be - if it's something routine/expectable, vs random).
What are the critical differences between the different devices and their abilities, or are they all largely the same?
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