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Landline Out, VoIP In

James F. Carter <>, 2016-08-26

We get 10 to 20 junk phone calls per day, and most days have no legitimate calls. This is getting ridiculous; why should we pay big bucks to our landline provider (Frontier) just to get spam? The alternative is a non-carrier VoIP provider.

Abstract: We signed up with CallCentric and we got the OBi200 ATA (Analog Telephone Adapter) by Obihai. Setup is far from plug and play, but someone familiar with configuring services can get it right on the first try. VoIP service is working as it should. But we were unable to meet our goals for digital security and for communication during a disaster.

(There is a glossary of acronyms at the end of this document.)

Table of Contents

Goals and Issues

The goals of this service change are:

These goals can be turned around to these issues:

VoIP Companies

There are a lot of VoIP companies in the United States (and worldwide). Here is a list, far from inclusive, resulting from a search on Google for VoIP companies in USA. Numerous 10 best lists appeared; bestness is not necessarily rated according to our criteria. For cost quotes I'm targeting a plan with an incoming phone number and about 100 minutes per month of prepaid outgoing credit to PSTN within the United States. Setup charges and the cost of the ATA are excluded. Business-oriented services are excluded.


One of the earliest VoIP companies. $10/mo, I think you have to use their ATA, which may be included in the current promo. Hard to get info from the website.


$10/mo. ATA is free.


$6.21/mo ($149 for 2 year contract). Located in Irvine, CA.






$4/mo. Located in New York City. They don't sell the ATA; you buy your own, or use a pure software solution.

Our son has been with CallCentric for several years and finds their service to be excellent. The recommendation of a satisfied customer counts for a lot. I was also surprised to find that CallCentric has the lowest price of the providers checked, although I believe that several non-checked providers have slightly lower prices. What do they have available and how well does it meet our needs?

CallCentric Plans

We need a plan for outgoing and for incoming calls. Technically the incoming part is optional but it's a little bizarre to have a phone that nobody can call.

Outgoing plans: Calls cost $0.0198/minute after your plan minutes are used up. There is a $1.50 setup fee except not on the free plans. Taxes are in addition to these prices. The fee for 911 service is $1.50/mo but you can opt out (not recommended). All plans cover calling to North America meaning USA, Canada and Puerto Rico, but Mexico is an international call priced separately; see below for the (low) price.

Incoming plans: Setup is 1 month's cost except where noted. Phone numbers are available in a lot of places (I can't say for sure every place) in the USA and Canada.

We would get per call outgoing and incoming, which would cost:

To use the service you can use phone software on your non-mobile computer, or an app (theirs or a generic one) on mobile devices, or an ATA which you purchase (not from CallCentric). Android 5.1 "Lollipop" and above intrinsically supports SIP (I don't know how far back the support goes), so you don't need the app unless you need more than generic features. Asterisk PBX software (and 5 others) works with their service. All the software has to do is do the SIP protocol with their server; you can use any of these strategies at different times or simultaneously, from any IP addresses, or you can restrict to your fixed IP if you have one (typically for business).

Sample international rates (per minute, mid-2016):

Country To Landline To Mobile
Australia $0.0242 $0.0330
Germany $0.0132 $0.0385
Mexico $0.0088 $0.0220
UK $0.0132 $0.0275
North Korea Only $0.80/min
USA $0.0198 (after your plan minutes are used up)

There are about 35 features apparently included with all numbers. These are jimc's highlights. Some of them can be turned on and off on the fly with star codes.

Reviews of CallCentric on This page has a lot of customer reviews concatenated (like Amazon product reviews). Almost all reviewers love it, with nothing to complain about. There was one scammer troll (who the other customers unmasked and took to task -- a fake passport and drivers license). One reviewer knows what codecs he was using: G.729A and G.711u; he can't tell the difference by listening. (Jimc research: G.729a has good compression while G.711u has none; differences in audio quality would be small.) Reviewers say that CallCentric may not be the rock bottom on price, but the combination of reliable service and good (and wanted) features counts for a lot. And jimc says, at our usage level, these prices are essentially zero. Problems encountered: In Hurricane Sandy, at CC's building in New York, both primary power and backup power failed. Also there was a DDoS attack.

Some bloggers are leery of a VoIP provider that doesn't do phone support. I have interacted with CallCentric customer support through their web form and ticket tracker on several issues, and I have been very impressed with how prompt, helpful and professional they have been. New VoIP users considering CallCentric, who are able to describe an issue comprehensibly in written form, need have no worries about getting customer support.

Picking the ATA

CallCentric's ATA compatibility list:

OBi200 (by Obihai)

$50, SBSF Amazon. Includes T.38. Configure to use up to 4 VoIP providers at once. Available codecs: G.711 u or a, G.726 (32kbps), G.729, iLBC. Wi-Fi or Bluetooth (via USB, sold separately, $20). FXO = Can originate calls to PSTN carrier (via USB, sold separately, $40). Ports: power, RJ45, USB, RJ11. Configuration: Log in to their portal. (Jimc research: it is possible to turn on autonomous web configuration.)

Note: a wired network connection is always better than Wi-Fi or Bluetooth. Use the latter only if a wired connection is really impossible.


$70, SBSF Amazon. Includes T.38. Similar to OBi200 except 2 RJ11's (for 2 separate numbers, can chatter at once), and 2 RJ45's, 1 to wild side, 1 to LAN, QoS priority for voice chat. Dimensions: 4.1 x 4.5 x 1.2 inch. First on Amazon: 2012-02-25.


$47, SBSF Amazon. Similar to OBI200 except no USB, no T.38, only 2 VoIP providers, no iLBC. On Amazon: 2011-03-24. OBi website says these products are at end of life.


Same as OBi100 except can originate calls to PSTN carrier. (OBi200/202 uses a USB dongle for this.)

Reviews for OBi products (on Amazon, looks like all 4 products mixed)

88% 4+5*, 12% 1,2,3*. Most people love it; featured review tells of $2000/year savings for a home business. Most people are using it with Google Voice which is Obihai's recommended use case. 2 bad reviews I saw: 1 died in a year (others update reviews after 2-3 years saying it's still great); 1 had trouble getting warranty support.

Cisco SPA112 / 122

SPA112: $49 SBSF Amazon ($34 from affiliate). SPA122 about $7 more. First on Amazon: 2011-11-15. 2 RJ11 + 1 RJ45, serve 2 lines at once. Reviews: Apparently when first sold it had firmware bugs, fixed during 2012. Recent reviewers mostly think it's good, except T.38 may be flaky (but this was from 2012-12-23, obviously another firmware bug.)

Linksys PAP2

$38 sold by affiiate, fulfilled by Amazon. 2 RJ11 1 RJ45. On Amazon 2016-02-26. Not a lot of details. No reviews yet.

Grandstream HandyTone HT486 / HT702 / HT286

HT702: $70 for a pair, SBSF outside vendor. $34 from affiliate, fulfilled by Amazon. 2 RJ11 + 1 RJ45. T.38. Supports TLS/SRTP/HTTPS. HT704 has 4 RJ11s. Includes vertical mount stand. Codecs: G.711 G.723 G.729A/B iLBC. On Amazon 2012-02-18. Can log to remote syslog. Hard for ordinary user to configure and troubleshoot. One person (after extensive troubleshooting) complains about losing registration (firmware bug?) One reviewer junked his HT702 and replaced with a OBi200 which was much better for him.

Linksys SPA3102

$198.61, forget it.

Cisco ATA 186 / 188

$139, forget it.

Telco AC-211

Not on Amazon.

Innomedia SIP MTA-6328

Not on Amazon.

Zoom ATA 5801

$82, SBSF outside vendor

D-Link DVG-14025

Not on Amazon.

Other products

All the products sold by and shipped from Amazon were on the compatibility list. A fair number of products from Amazon affiliates (outside vendors) were not on the list, though. Generic products that can do SIP are likely to work, but no actual confirmation of success is posted.

Discussion of ATA: The Obi products are the most favorably reviewed; the Grandstream and Cisco (including Linksys) products have a lot of complaints. Obi's cost is competitive (some competitors are ridiculously expensive). Doubtful issues have been resolved to my satisfaction as follows:

Settings for a generic ATA or software:

Action Plan

We need to decide the network geometry that will make QoS most effective.

Action items:

Installing the OBi200

The Obihai OBi200 arrived. The box includes the OBi200 (RoHS compliant, about 69 x 69 x 25mm), a wall wart, a 1 meter Ethernet cable, and instructions. A phone number on the OBitalk network is preassigned, on the label on the bottom of the box. The MAC address and serial number are also shown. Apparently it is pre-configured for their network and you can use it to call and be called from within the network, without registering on their website.

Physical Installation

Initial Configuration

Sign Up With CallCentric

Configuring the OBi200 for CallCentric

See this useful blog post about digit maps from Aveo Systems. Digit maps are defined in RFC 3435. It's basically a modified POSIX regexp with alternatives separated by '|' (pipe). When the dialled number matches one of the choices there's a short wait for more numbers to come in, then the number so far is dialled. 'x' matches any digit; '.' (dot) means the preceeding unit is repeated zero or more times; digits or ranges in [] match any one of the enclosed digits.

Purchase Rate Plan

You have to have a working SIP connection before you can pay to use it. In other words, you need to successfully finish the above configuration steps. From the Test Call page, follow the link to Purchase Rate Plan.

More Setup and Testing

Call Treatments, Voicemail and Phone Book

The PBX software on the VoIP provider's server lets you route your calls in a wide variety of fantastic patterns, called call treatments. Log in to your CallCentric account and go to Call Treatments in the left column. Hit Add a New Call Treatment. These are the fields in the call treatment record:

Our call treatments: They should be applied in this order; on the summary page there are arrows to re-order the rules. At the bottom of the form put a title or description to help you remember what the rule does. All our call treatments apply at all times. Since my Spit (Spam via Internet Telephony) reputation is good, I can't really test the telemarketer blocking until after we port our number.

Internet Down (1)
If status is not registered, and the spam probability is low, transfer to my cellphone number. [Works.]
Internet Down (2)
If status is not registered, and the spam probability is medium or high, hit them with SIT tone 1 (number not in service).
If the sending number is in my phone book, send to my (only) extension without telemarketer hassle. [Works.]
Foes (Telemarketers)
If the spam probability is medium or high, play SIT tone 1 (number not in service).
Specific blacklists
We have a few charities that we've already contributed to, but they call us incessantly anyway. Even so, we don't want to report them as Spit, so we give them SIT tone 1.
The default is to send to my extension, or to voicemail if not answering. [Confirmed working.]

Discussion of the default rule: I could make the default to be the telemarketer blocker. But we get robo-calls from doctors' offices and pharmacies which we want to actually receive. So I'm only doing the telemarketer block thing when the number's spam reputation is high.

What to do to telemarketers: Initially we used the telemarketer block feature, and it was quite effective; we had a dramatic reduction in the number of unwanted calls. However, to do the telemarketer block thing, CallCentric has to accept the call and talk to the telemarketer, for which they have to pay their upstream provider, and charge the customer, whereas SIT tone 1 or another standard error message is free. We found that we were paying as much for telemarketer blocking as for the fixed charge for the line. It was a lot less than for our PSTN landline, but even so we were annoyed, and so we adopted a less nice strategy for the telemarketers, SIT tone 1 (number not in service).

In e-mail I had a very bad experience when my outsourced provider added a reputation-based filter and tossed (refused to accept) mail from any source with a bad reputation. My work server got blacklisted because a lot of users forward all mail, including spam, to Gmail. Of course the correct solution is to run all outgoing mail through SpamAssassin, but management didn't accept this solution. So I fired my e-mail provider. It's important when using a reputation-based filter to accept the traffic but to tag it or otherwise apply a non-fatal mitigation strategy.

Under Preferences - General - Send unanswered calls to message after: The default is 30 seconds. I left it that way, but the traditional value for PSTN is 60 seconds, to give the farmer's wife time to come in from the barn and answer the phone.

Voicemail is considered to be a separate product at no cost. Once you order it you get a tab for it under Preferences. The defaults look fine, except you need to set the PIN (4 digits). To do so, follow one of several links to Preferences - Voicemail and Edit the Global Settings. To listen to voicemail: from the CallCentric (home) phone(s) dial *123 or *86, then follow voice prompts. From outside, dial your CallCentric extension (home phone); wait 30 secs for the leave a message prompt; as soon as it starts playing dial * and your PIN ended by #. If you have separate mailboxes for multiple extensions, dial the extension before the PIN (no separator). Then follow the prompts -- 7 to delete the current or most recently played message.

To set up your phone book the easiest way is to export an existing phone book as a CSV file. Field order is speed dial code (4 digits, leave empty if not used), full name (in double quotes, last name last), phone number (with country code, digits only, no spaces), group name (quoted if it contains blanks; leave empty if not used). To make sure, add an item by hand, export it, and inspect the result. When you import, new entries will be added, duplicates will be ignored, but if the full name or the phone number matches an existing entry the imported record will replace the existing one. (So a person's phone number could be changed, or the person owning a phone number could be replaced.) I wrote a small script to extract the needed fields from a file of vCards (VCF) and write them out as comma separated values (CSV). Then the file needed hand editing to exclude irrelevant numbers. I then imported this file.

Porting the Number

See CallCentric's FAQ about Local Number Portability. Jimc's excerpts and history of what happened:

Internet Service Transition

We started out with Verizon FIOS Digital Voice, and Verizon's 15/5 Mbit/sec Internet service, later upgraded to 25/25 Mbit/sec. Verizon FIOS was sold to Frontier Communications, which continued the services unchanged. But there is a botfly in the ointment: both of them are phone companies, and the entire account is keyed on the primary billing phone number. Which now belongs to CallCentric. Therefore you need to guide them through the transition process. Other people who let nature take its course found that the natural state of disloyal customers is to not have Internet service any more.

On the porting day I checked when it actually happened. CallCentric and I notified each other when the ported number seemed to be functional. Then I got on the (CallCentric) phone with Frontier customer service. I'm omitting a lot of details in this tale…

Logging to the Syslog Server

SIP on Android

This is tested on CyanogenMod-12.1 based on Android-5.1.1 Lollipop. Hardware is a Samsung Galaxy S5 model SM-G900M (should be irrelevant). See CallCentric's support writeup about Android, which is actually for Android-4.1 Kitkat. To set it up:

Quality of Service

VoIP traffic needs to be sent out promptly and reliably, and in particular, it needs priority over hog flows like downloading a giant file. This kind of traffic control is generically referred to as QoS or Quality of Service. My traffic control script uses QoS code points (action numbers) from RFC 1349 to decide priority. But RFC 1349 is now deprecated and is superceded by RFC 2474. I need to revise and improve the script.

That turned out to be a lot harder than anticipated, but I finally figured out what was really going on, and now VoIP is getting its proper priority both outgoing and incoming. It turns out that RFC 2474 is totally useless, and I need to ignore the QoS code points and use other means to recognize the flows that need high or low priority. See my separate writeup on QoS (Quality of Service).

Issues That Were Not Productive

Security Setup

First we need to be clear about security in VoIP. There are two phases in VoIP: SIP signalling, in which you identify yourself to the server and tell who you want to call, followed by sending and receiving the voice payload, normally using RTP. Secure protocols exist by which you can keep both phases secure from prying ears in transit from your site to the VoIP provider.

However, in the United States telecommunication carriers including VoIP are subject to the Communications Assistance for Law Enforcement Act (CALEA) and the Foreign Intelligence Surveillance Act (FISA). CALEA requires that the carrier must design their system so law enforcement personnel could listen to and record the throughgoing traffic. Read the writeups and the law text to see when a warrant is required -- basically when one end of the conversation is a USA citizen or permanent resident -- and what the law enforcement personnel need to allege in order to get a warrant.

There is extensive discussion about whether these laws are appropriate and/or abused, all of which is off topic for this document. But also, USA government agents are not the only ones who might be interested in a high-value conversation: other governments are known to steal secrets, and industrial or business espionage is a well-known threat. If you have something to hide, you need to assume that provider personnel have gone over to the dark side (or to the side that you are not on) and are in collusion with the enemy.

In addition, once the traffic reaches your provider you do not have direct control over how it is protected (likely, not at all) on the often multiple hops to the recipient.

Thus it is pointless to use secure protocols to your carrier's server. If you are serious about security you need to run VoIP server software on your own server that is inside your own security perimeter, and to communicate only with partners who that server can reach directly. Or you should set up your and your parters' cellphones so they can accept SIP connections peer to peer (using security) rather than going through a central server. Then you only need to concern yourself with traitors within your own organization, and subpoenas directed to your own organization.

Nonetheless, as a wearer of the tinfoil hat, I want to finish the ultimately fruitless project of investigating what security the OBi200 is capable of doing, and how much CallCentric will support it. In the configuration pages of the OBi200 check out these items:

I raised the syslog level to 7 and turned on the above items one at a time in the order listed above. I could have diagnosed the problems better if I had used tcpdump or wireshark, but since failure is a foregone conclusion, because subsequent transport hops are likely unprotected, I'm not going to put a lot of work into these tests.

Survivalist Kitsch: Communicating Minus Power

The biggest problem with VoIP is, if your digital network goes down you get silenced. Threats to the digital net, ordered in jimc's opinion of likeliness, are:

Suppose you had solar power and a big battery, so you could run your house network through a long power failure or other loss of outside connectivity. What kind of a back door could you get? A wormhole through Hausdorff space is attractive, but the technology is not mature and there are no commercial products. The only credible alternative is satellite communication. I didn't put a lot of time into this research, but this article makes some good points: How and When to Buy a Satellite Phone in Forbes (2013-03-18) by Marc Weber Tobias. Jimc's highlights from the article:

Some research on prices:

Conclusion: This kind of backdoor may be attractive to some people, particularly if you have an ongoing business need or if you provide emergency services, but with those prices a casual user is advised to make do with 19th century communication (carrier pigeons) when electrical power fails.

Telephone Number Mapping (ENUM)

Wikipedia article about Telephone Number Mapping (ENUM). ITU-T E.164 is a scheme by which ranges of phone numbers are allocated to countries (using country codes) and within that to carriers, under national jurisdiction. Per RFC 2916 (2000) there is a domain $ (where $number is the phone number, little endian, with dots between the digits). The RR(s) with this key are a NAPTR record (per RFC 3403 (2002)) which indicate a URI that should be connected to, which could point to your VoIP provider or some alternate.

Unfortunately ENUM is not supported by many countries, specifically country code 1 (North America). Here are a few countries having delegated domains: 44 (UK), 43 (AT), 49 (DE), 46 (SE), 358 (FI), 86 (China, PRC), 886 (China, Taiwan). Countries checked but not having domains: 1 (North America), 7 (RU), 45 (DK). There is an alternative registry called and North American users can register there, but only a minority of users do so.

I do not have an operational need for anything other than the default SIP URI, so I'm not going to register with

Remaining Items

Yet to do:

Finished items:


Here are some suggestions I have for CallCentric:



Internet Protocol. This is the basic format of a packet on the Internet; within the payload area of the IP packet a variety of more specific protocols have been defined. IP is governed by RFC 791 (1981) and RFC 2460 (1998) (both as amended).


Voice over Internet Protocol. The voice chat payload is sent to (or partway to) the partner using the packet switched internet protocol.


The kind of telephone invented by Alexander Graham Bell. Wires run across the land (ocean cables have different technical issues) from the central office to your analog phone.


Public Switched Telephone Network. While it is relatively simple to connect phones whose landlines run to the same central office, citywide and nationwide service requires a network of trunk lines between central offices, using VoIP or older technology, and called the PSTN.


Analog Telephone Adapter. Connects the digital world, i.e. your VoIP provider, to traditional (analog) phones. All the phones in the house (within practical limits) can share one ATA.


Fiber Optic Information Service. Packet switched internet traffic is carried to and from the customer. Television signals can be distributed in parallel using different technology.


Optical Network Terminal. It receives the FIOS signal and sends the payloads out to the customer's home devices. Specifically, it can act as an ATA from the carrier's own VoIP service.

Whitelist, Blacklist

A list of partners, in this case phone numbers, which are either allowed or blocked from communicating.


Private Branch Exchange. The caller dials your company's central number, then can dial an extension number to reach numerous phones in the business. Modern software like Asterisk has features like voicemail and is useful for both large and small phone nets like a family or a home business. See also DID.


Direct Inward Dialing. A range of phone numbers is registered to be routed to the VoIP provider, PBX, or shared FAX interface. The term DID is also used to refer to a single one of those phone numbers that you rent from your provider.


Session Initiation Protocol, for setting up multimedia communication, not just voice chat but video calls, a shared whiteboard, and other features.


Realtime Transport Protocol, optimized for sending a media stream almost isochronously, i.e. the recipient gets the data at the same time it happens and is recorded by the sender, as near as possible.


Secure Realtime Transport Protocol. See also ZRTP.


Quality of Service. In IP this is handled by the Differentiated Services (DiffServ) field of the IP header; see RFC 2474.


An ITU (International Telecommunications Union) standard for transmitting FAX over IP.

RJ11, RJ45

Connectors for an analog phone line (RJ11) or Ethernet (RJ45). The RJ11 has space for 6 wires of which usually only 2 are connected; the RJ45 is similar but has space for 8 wires.


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