I was trying to get more info on Anveo sub-accounts that is somehow missing or hard to find and coordinate on their website and this forum. More or less discussions were run around their attempts to charge 911 fee separately per each sub-account even registered at the same location https://www.dslreports.com/forum/r29262969-Anveo-Setting-up-sub-acnts-in-relation-to-subscription-packages-codec. Beyond that info is scarce, namely:
- can 911 fee on sub-account be waived upon request by filing a ticket with Anveo, if the sub is registered at the same location and on the same name - did anyone try that? Charging a separate 911 fee seems to contradict their Terms and Conditions, where the fee is to be charged "per address", not per sub or name?
- when several sub-account SIP phones are registered online, and someone dials your main account DID, will it ring simultaneously on each SIP phone and softphone? If one of these is answered, can another registered SIP phone be picked up & join the conversation any time later during the call?
- while 911 registration is required to make outgoing calls, its not needed to receive incoming calls on any active sub-account SIP phone or softphone - correct? Is any balance required on a sub-account to receive incoming calls, despite its linked for payments to the main account?
- some suggested to register sub-account address in Mexico or another country without 911 requirement. It was mentioned, in such scenario a caller would need to dial 1 to call local phones in her area - is that correct? If yes, does it mean, your per minute charges depend on 911 address instead of your and the contact DID area codes? Also, if dialing 1 would be needed, will such local calls be charged at international call rates?
- what are the differences between Anveo sub-accounts and extensions? Can each sub-account SIP phone be assigned its own extension, and how? Lets rely on Anveo basic Free sub for all the answers.
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[Anveo] More on Anveo Sub-accounts
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[Other] Ooma dropping support for it's older white (first) model
Ooma will be dropping support for it's first model, all older users should upgrade to a new Telo HD by 10/29/2016 (IIRC). Ooma will still respect older users 911 fee and tax exempt status (that is no charge for 911 and taxes) however a new model Telo will be required which costs $79.99. If you have an old Ooma account with no monthly charge, this is your time to continue free service. My use has been since early 2009 on about a dozen phones.
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Federated VOIP
I would like to start a discussion about federated voip.
The wikipedia article, for what its worth is at https://en.wikipedia.org/wiki/Federated_VoIP
I disagree with some of the requirements in the article primarily because they are barriers to entry
I think federated voip adoption has been stymied by these requirements.
enum is a single point of failure and is not required.
TLS is a barrier to entry and removes the individual's anonymity.
Here is starting point for a usage model I propose.
Your input is requested.
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alice@sip.atlanta.com calls bob@sip.biloxi.com
Alice's sip client sends a call request to sip.biloxi.com
sip.biloxi.com looks up Bob's contact address to send the call request to Bob's phone.
URI's look like sip:alice@sip.atlanta.com which should be intuitive to anymone who has used the internet.
The key point is only the callee's server is involved, not the caller's.
Server considerations
The server MAY be a proxy or B2BUA.
The server MUST be a registrar.
No PKI or enum is required.
The server MAY stay in the signalling path but SHOULD not be in the media path.
Removing the server from the media path makes the server much more scalable and reduces media latency for the users.
For maintenance and billing purposes logs and CDR's that contain user information MAY be generated.
The server is not required to support ICE or any nat traversal unless it is a media endpoint
TCP
MUST be used for signalling to avoid fragmentation issues.
TCP keeps the NAT pinhole open whereas UDP often times out.
Spam
System wide blacklist
User specific
whitelist
blacklist
time of day filter
User considerations
NAT traversal
ICE support is required at the client side for NAT traversal.
TCP MUST be used for signalling.
Privacy
Assume the server retains logging information about calls.
ZRTP
Using ZRTP assures the privacy of the actual conversation.
Unless of course you don't use it or some hacker(s) breaks ZRTP.
Security
?
Why no pki
?
Other
It is hoped that the decentralized nature of the system keeps things simple and open.
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[Equipment] Looking for IP phone suggestions
I believe my pap2t finally crapped out as it randomly disconnected one night and wouldn't reconnect. Had to do a factory reset, connected now but it refuses to dial out, even though I can receive inbound without any issues. I tried doing everything I can possibly can, so I imagine this is equipment failure at this point.
I never used IP phones, but figure I will give them a try. I'm not really looking for any features as it's for residential use to simply make calls, so I am looking mostly for something affordable.
I'm currently looking at the Grandstream DP720 with base for $130 CAD or the Yealin W52P with base for $150 CAD.
I'm looking for long term reliability more than anything else. Anyone have any good experience with either one of the two, or something else that's in the similar price range? Thanks!
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[Voip.ms] Switching from Ooma to VOIP.ms - ATA question
We've had Ooma for about 2 years now (not the premium service) and as we rarely use the phone I've decided to switch to something that doesn't have a persistent monthly fee. I did some research, and really like voip.ms. Correct me if I'm wrong, but I can go on pay as you go or load it with a set amount of account credit, vs a monthly fee right?
I'm unsure what ATA to use, tehre's a few to choose from. Are there any differences between them on the whole, or one that's better than another? Our usage isn't much, maybe an hour or two (tops) each month), I'd like it to support a wideband audio codec so calls sound good. The option for 2 lines would be good as well.
Thanks! :)
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experience with www.freephoneline.ca?
does anyone have any experience with www.freephoneline.ca? Looks like you buy the VoIP Unlock Key once and never pay anything again? Not sure if thats true..
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[VOIPo.COM] Voipo eliminating prepaid and discounts?
Just got an email about renewing for another 2 years on Voipo (which I had already done 3 weeks ago) and in it the included news that they are eliminating the 2 year prepaid and its associated discount (which came out to around $6/month). Instead they are instituting month to month premium plans with the hint that the monthly price will double. I am locked in for another 2 years but doubled price may have me shopping again.
Anyone else on Voipo see this and/or have more info?
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[Other] future nine set-up
hello everyone! i have a samsung galaxy S4 and i reinstalled the CSIP with future nine... i use this application to call home in Romania... can someone tell me what are the settings to set and make the nesessary adjustements for CSIP i do not know the server domain...for future nine...
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[Future9] F9: Website and Miami server down?
The generic outgoing.future-nine.com resolves to the CA server.
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[Anveo] the balance of your account is too low...
Hey Guys,
I am currently run a couple phone numbers for family and friends with voip.ms. Never had any problems.
For my personal phone, I use Anveo.
I have recently tried to port over some of my voip.ms phones over into a reseller account on Anveo. The porting seems to have gone ok, incoming calls etc work.
However, everytime my user tries to call out they get "the balance of your account is too low anveo". Of course, I made sure my main account has plenty of money in it $50+ and I've even credited the customers virtual account with 30. No matter what I do, they get this same error sending out.
Of course opening tickets with them is an absolute nightmare and customer service is bad as everyone here knows, I guess I was just banking that I wouldn't have to deal with them much (I never had to deal with voip.ms support!)
To make matters even more confusing, 2 days ago the customer was able to call out no problem. It's almost like this "balance of your account is too low" message is intermittent.
Hopefully someone else can point me in the right direction. I really don't want to go through porting all my numbers back to voip.ms if I dont have to.
Thanks!
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Send2SIP
We're happy to announce the launch of Send2SIP: https://www.send2sip.com
Send2SIP provides a free service allowing you to forward or simulring calls from a traditional landline, cable voice line, or cell phone to a SIP URI over VoIP.
Most telephone lines will work if they support call forwarding or simultaneous ringing, and we've provided setup guides for some of the larger carriers on our website. If you don't see your carrier listed, but they do support call forwarding or simulring, give it a try and let us know your experience.
Additional details are listed on the Send2SIP website, but if you have questions please feel free to contact us via the contact us form on our website.
NOTE TO MODS: this posting was discussed with dbmaven and fourboxers before posting.
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[General] Caller ID Spoofing on VOIP
Thread got deleted -- this is a different topic moderators, but I would say they're not doing more as they see the PSTN as dying, and it's not being enforced by the FCC, plus, we're dead. And, it's money lost, so yes, you're right.
CPN Screening on DMS came out in 1998 -- maybe earlier.
DMS would send the following bits when CPN Screening was enabled for PRIs.
SI of 01 for user provided number, verified, passed
SI of 11 for network provided number
SI of 00 for user provided number, not screened for spoofing
SI of 10 for user provided, verified, and failed (may be a spoof)
10 could be set to swap to 11 without telling you, and throw out the spoof, or reject the call.
So, I can't spoof. I could spoof on VOIP though. We did a friendly spoof to test security.
So I'm asking VOIP providers why they aren't doing this. They can't be bothered. We did something like this with private and unknown callers too.
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MaxEmail Fax Service taken over by j2 Global®
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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Is Nuvio still around?
Or is their website just online with no one at the helm? Unrelated to my ongoing VoIP decisions for residential, but their name came up in conversation the other day. Perhaps someone here knows?
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Trying to reach Sam@Circlenet
I am trying to reach Sam about a refund. Emailed him it will soon be 3 weeks ago. Thought he might still be reading this board. Feel free to PM me or email me if you recognize the nickname.
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[Anveo] Anveo Direct: Any News On E911 Coverage?
Is there any estimated release date available for Anveo Direct's E911 coverage? The feature was announced as coming "soon" about 3 years ago (February 2014). It has been mentioned and confirmed several times in 2016 with no definite release date other than "soon".
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[CallCentric] Non-API hack for spam score of incoming CC calls w/ Asterisk
I have an Asterisk PBX and route incoming calls that are likely to be spam in different ways. Historically I have done this with TrueCNAM's API.
Today I came up with a simple hack to eliminate the API. There's nothing wrong with using the API of course, but since Callcentric already looks up the spam score, me doing it a second time is redundant. Some service providers include the TrueCNAM score in the SIP headers, but Callcentric doesn't. This hack however effectively does the same thing.
For this example, my Callcentric number is 17770000000 and my DID is 134711111111.
Simply create a Call Treatment setting Called Number to 13471111111 and Spam call probability to High. Configure the treatment to redirect the call to This number: 17770000000.
Now, simply inspect ${SIP_HEADER(TO)} in your dial plan. If it contains 17770000000, the spam call probability is high. If it contains 13471111111, the probability is medium or low. For example:
same => n,NoOp(Spam probability is ${IF(${REGEX("17770000000" ${SIP_HEADER(TO)})}?HIGH!:low.)})
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[Future9] How to block a number?
Greetings Nitzan,
Is there a way to have blocked callers list on F9. I have this Commercial Magazine Service (of Holland) that calls me all the time and I can't get them to take me off their list.
Is this possible?
Thanks....
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[Asterisk] OAuth 2.0 Support for Asterisk 13 or Asterisk 14
The attached package should provide OAuth 2.0 supoort to any Asterisk 13 or Asterisk 14 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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Best sip account for Answering Calls via iOS
I currently want it where people who call my main cell number that it instantly fowarded to a voip number which then rings my mobile phone via an app. (Call forwarding for me is free) My current issue is LAG, I have my phone forward to a Ringplus number which is hosted on my own connection that plays hold message/music till I awnser while its ringing a Freedompop number Voip app on my iPhone. But answering is 50% working along with awful LAG! Its not my connection as NetTalk or other plain voip calls work perfectly fine.
I just want it were whoever calls my mobile hears music till I answer. Any service thats not over $5 that can have hold music till answer? Would need incoming calls only as I'd be answering via a sip app
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