Note that you do not need a VOIP provider to place calls to other SIP destinations. With proper configuration and dynamic DNS you could call your friends using an address such as ffriend@yr.ip.add.res:5060. You do need a VOIP provider to call regular phone lines.

Major providers, Canadian flava

In North America, Vonage seems to be a major player; however, their advertised plans are not all that is available. Here’s what a user has reported:

I just got the $24.99 retentions deal ($40 plan for $25) and one month credit for my plan and my virtual line. I just called and told them I wanted to cancel. They transferred me to retentions. The lady seemed surprised since I've been with Vonage for 5 years. I told her that Rogers was offering their VOIP service with the $30 plan for $20 (see Rogers retentions thread) for a 1000 mins and that this was worthwhile since my usage tends to be between 700-900 mins per month. And I said that I was also getting a second line on cell with the same area code as my virtual line for 0 dollars a month (0.20 PAYGO) with Rogers which would meet my needs since my second line was limited usage.
She said that "New for 2010, I can make you a special offer..." And offered me the retentions deal and a month free. There's no contract or term. I can cancel at anytime. I said, "Fair enough. Set it up." Logged in to check. Credits are there.

Voip.m seems to have some unique and advanced features. It is also possible to use it without fully understanding all that they offer, but otherwise they seem to have no equal in the amount of power they provide. I will go one step further and say that I previously had a similar idea - “mass market” asterisk features – but they are implementing it so well, I lost the appetite to do it myself.

Skype has the most users. It is very simple to set up and use, however, its technology is proprietary, which makes it difficult to use with established open server applications such as asterisk.

MagicJack will send you not only service, but also a hardware device you connect to your computer through USB. For 2010, they have plans to send a GMS femtocell instead, allowing people to use their cell phones at home. The femtocell would be great, but I have yet to see it. The USB device I used needs a computer to be on at all times and is otherwise very limited in features and support; however, it is very attractively priced. However, it is worth knowing that MagicJack sued and lost boingboing after this website wrote that MJ spies on its users and has a failing customer service card from the Better Business Bureau. Personally, I thought there’s something spooky about this company when I watched a commercial with CEO Borislaw’s daughter who seems scared and coached when talking about dad’s “invention”.

ooma was (and may still be sold) at Costco. A very high initial cost (almost $200) gives you a “free trial”, but the service is not that cheap once the trial runs out. Some early 2010 news from about.com:

..unveiled a series of new features to its already rocking service. These features are quite new on the market and ooma is the first residential VoIP service provider offering them. The features include HD Voice, which doubles the frequency to provide a close-to-natural voice quality. Voicemail transcription comes here again, after Google Voice introduced it. ooma also provides extensions for Google Voice and allows mobile calling through the iPhone. But the feature ooma presents in the front is what they call Ooma Pure Voice, which is a kind of network sophistication that makes phone calls robust and delivers quality despite congestion and low bandwidth. These new features will be [most likely, were] presented at CES 2010..

Google Voice has an amazing offer, but it is not widely available yet and you need an invitation to use itonly available in USA. VoiceMail transcription (where you don’t receive just the voice recording, but also a transcription of messages in your mailbox), switching phones in the middle of an incoming call (just press *) and the very competitive cost of outgoing calls will make this service, when deployed, very tough to beat. The service is currently available only to users in US and Alberta (403). There are ways to obtain the service even when you are not among the lucky ones.

Freephoneline offers free phone calls in Canada using their softphone. If you would like to use SIP, the settings can be yours for $50. Though I received their phone call, I do not like the idea of “free for life” as that usually means “free until nobody signs up with us anymore”. This might be a good idea, but I declined. You do get a local phone number and Enhanced Voicemail with them and their offer is quite competitive otherwise.

Betamax (German) or Finarea (Swiss) offer various rates to most countries under several brands: 12voip, actionvoip, budgetsip, calleasy, dialnow, freecall, internetcalls, intervoip, jumblo, justvoip, lowratevoip, netappel, nonoh, poivy, rynga, sipdiscount, smartvoip, smsdiscount, smslisto, sparvoip, voipbuster, voipbusterpro, voipcheap, voipdiscount, voipian, voipraider, voipstunt, voipwise, voipzoom. Each brand has different rates with different countries and is SIP compatible. Furthermore, each brand has a number of destinations that are free for 30-minute calls. It is unclear however if they are free forever or the destinations change. Also, whenever you purchase credits, you also get a number of free services. I find the array of rates challenging and confusing, even when using backsla.sh. Furthermore, their users seem increasingly frustrated, complaining that calls do not get connected, that tech support is inexistent and that prices have been increasing for no reason.

Rogers, Telus and other ISPs as VOIP providers is not an idea I like. Large ISPs are known for poor customer service. Smaller VOIP providers are usually in a better position to provide superior service and work harder for their customers. The large ones get business mostly because of the inertia of their clients. Furthermore, the price plans they offer make sense only for people who make many local calls and almost no international calls and even then are quite expensive. Some plans have absurd restrictions – for instance, they would not allow you to travel with your adapter. Their adapters also tend to be locked on that particular network.

G3 Telecom was once my long distance provider. That has changed when I referred some friends who signed up with them but we were then denied the promised discounts. Their VOIP plan costs $10/month and it includes Caller ID, Call Waiting, Call Return, Call Forward and Call Back, but Voicemail is an extra $5. Their offer is not bad, but it’s more expensive than what I have and it comes with less features.

Teksavvy has a strong reputation for customer service, but they are far too expensive for my liking. With their unlimited local calling plans starting at a hopping $21.48 / month with $25 activations fee and all calling features extra, Teksavvy appears poised to compete with the major ISP VOIP offerings, aiming to simply provide better service. Still, their unreasonably high prices place them out of the reach of most customers. Visual Call Waiting adds an extra $9 / month, while Voice Mail is $6.

Acanac includes calling features in their standard offering, however, their customer service is lacking. I signed up for a free 6 month promotion they had a while back and when I wanted to cancel, they refused to disconnect me, attempting to fraudulently charge. I had to file a dispute with my credit card company to have the charges reversed. Their no-frills plan is $10/month while the “unlimited local” is $20/month. Many angry users report similar “internal misunderstandings” between Canadian billing and Indian tech support resulting in frustration and billing disputes with their

Primus, with their talkBroadband offer is also too expensive. Their Basic Plan at $15/month does not include any calling features. Visual Call Waiting alone is priced at $12 extra / month.

Other providers worth considering are callcentric, lingo, les.net and vbuzzer.

Codecs

Though this should not normally be a factor in your decision, here’s a list of VOIP providers and their codecs as reported by users not long ago. It is very possible that in the meantime, all providers have started to support GSM, G.711 and G.729. G.722 or G.722.1 aka HD Voice offer the best voice quality, but your hardware needs to support it as well. If you cannot use G.722, your best bet is G.711u – which we recommend.

AT&T Callvantage: G.726 (?) or G.711 with Fax and Modem support turned on

Broadvoice: G.711u (g.726-32 and g.729a are available on the Chicago proxy)

Sipphone is currently using G711 and GSM on their PSTN gateway; 800 numbers gateway is using G711 only

Stanaphone uses g.711u & GSM, G.723

SunRocket --- G.711

TalkBroadband uses G.711 for users with full broadband connections, and G.729 for users with 'Lite' connections.

TelaSIP uses G.711, G.726, G.729 and GSM.

Verizon VoiceWing has standardized on G.711. G.711a as default and G.711u when the connection allows

ViaTalk : G.711u (Ulaw Compression)

Videotron.com Cable VoIP G.711

Vodavi G.711 @ 110kbps for "full" B/W connections & G.723.1 @ 22kbps when B/W needs to be limited; G.723.1 can sound almost toll quality when you have some control over QOS.

VoicePulse: G.711/G.726-32/G.726-16

Vonage: G.711/G.726/G.729

Voz Online – G729a

The following are some example of voice quality MOS (Mean Opinion Score) scores collected before. G.711 has the best voice quality. However, you need to pay for the price of high bandwidth.
1. PSTN switch: MOS is close to 4.5
2. G.711 A/M: 4.4 or up
3. G.729: About 4.0
4. G.723: About 3.5
5. G.726: If I remember correctly, depending on its rate, the score is between 3.5 to 4.3.

Tech support

I first contacted the VOIP provider, prior to spending money. We had the following conversation, which hopefully will not only answer some of your own questions, but also give you an idea on how providers do or should deal with questions.

You can start the process via the customer portal under DID Numbers -> DID Portability.

If you decide to port your number, you will have to make sure that your balance cover the fee of the port ($25 per number). Otherwise, our staff will not start the process until the balance is in a positive standing after you have submitted the port order.

[Stephen] the average time for port process is 2-4 weeks [ib] if my subscription with skypein (winterpark) expires in 2-3 days (i.e., while the porting would take place) could i lose the number? [Stephen] probably, because we cannot port non in service numbers, in order that the port can be completed the number must be active [ib] what would the annual cost be for both numbers (unlimited incoming residential)? [Stephen] we do not have annual plans, only monthly plans [Stephen]

We have two offers for Canadian DID along with two billing plans. The Per minute DID's are $1.99 per month, $0.0149 (1.49¢) per minute with 6 seconds billing increment and unlimited channels (simultaneous calls). We also have per minute plan DID's that are $1.49 a month and $0.01 a minute. Flat rate DID's are available at $5.95 or $4.95 per month with 2 channels and free incoming (3500 minutes a month for residential usage)

[ib] pls explain simultaneous calls; is my PAP2T NA able to handle those, and would that be call waiting or just routed to vm? [ib] billing would be per call or per time used, irrespective to the number of calls in the unit of time? [Stephen]

the calls are billed per minute according to the duration, with Flat rate plan, the incoming calls are not charged

[ib] so if i get 2 simultaneous calls of 1 minute (each), I'm billed for only 1 minute, correct? [Stephen] no, on that case you will be billed for the two calls according to the duration of each one [ib] am I billed for calls going to vm as well? [Stephen] yes, because is considered as answered [ib] I'm trying to determine what's best - 1.99 + 0.0149/min or 4.95. It seems that flat rate makes sense only if I get more than 200 min incoming calls, correct? [Stephen] that's right [ib] now, to set up my PAP. i'd like to make sure the system works before porting my numbers. what is the best way to do that. do you have some dummy numbers to try with? [Stephen] no, unfortunately we do not provide test numbers&accounts. However, please note that if for any reasons you are not satisfied with the service provided, we'll refund what is remaining of your balance within 7 days on your request.

You can still tour the interface, register your device and do the diagnostic tests such as DTMF and ECHO without adding funds to your account.

[ib] I'm not too concerned with your performance, as you are very highly rated. I am concerned with my learning curve and the fact that Internet / router QoS may be inadequate. I want to ensure that there is no c [ib] "choppy sound" [ib] That's why I'd like to experiment first. What is DTMF & ECHO? I suppose ECHO is a service that will read back whatever I say, to show that outgoing calls work and DTMF the same for tones. How would I check that incoming calls terminate well? [Stephen] for the incoming calls tests you will need to order a DID to perform this tests, the DID can be cancelled when you want [ib] that would cost me the $25 port fee + usage, correct? [ib] i.e., if I order a number in Toronto for checking purposes, I'd be paying $25 + $1.49 for the first month + 0.0100 / min [Stephen] if you port the number yes, but for testing purposes you can order a DID number directly on our portal, they are delivered instantly [Stephen] no, to order a number, there's no need to pay the port fee [Stephen] that fee is for port your number from other providers to us, is a one time fee of $25 per number [ib] ok, got it [ib] so for the testing number it would only cost me usage, at 1.49/mth + 0.01/min [Stephen] that's right [ib] my PAP2T NA has 2 phone line jacks, each with 4 connectors. does that mean that it can handle 4 incoming numbers (L1 + L2 each)? [Stephen] no, as far as I know, PAP2T can handle only one call per Line [ib] Is there a step-by-step guide to set it up with your service, or should we do this online? [Stephen] on your costumer portal, you can find some configuration samples that you can follow to set your PAP2T, it's located on SUPPORT>>CONFIGURATION SAMPLES [ib] my LAN is behind a consumer-grade 10/100 Mbps router/firewall. i have connected to it another Gigabit router. could there be a performance issue if I connect the PAP to the Gigabit router as opposed to straight into the main router/fw? [Stephen] No, there should not be a difference [Stephen] Is there anything else I can help you with? [Stephen] Due to inactivity this chat session is being closed. Please come to live chat again or send an email if you require further assistance.

Second chat session (after I had completed initial setup).

[ib] would like to do the DTMF & ECHO tests [ib] (I just connected my PAP) [William] Hi ib [William] for the echo test please dial 4443 [William] and 4747 for the DTMF test [ib] thank you [William] You welcome , Is there anything else I can help you with? [ib] yes [ib] i was wondering if there is any charge for toll-free numbers (800, 866 etc) [William] if you use the value route no, if you use the premium route yes. [William] outgoing call. [ib] what is the difference? [ib] also, do you have an FAQ list or PDF guide of your service? [William] The difference between value and premium route is that in premium route we guarantee quality and caller id issues. The both routes are very good. Sometimes the value one get some problems that the premium route does not get

[William] We don't really have a FAQ or guide available , but we can guide you in everything related with the service. [ib] 1) what is the price difference and 2) what are the problems that are more likely to occur with the value option? [William] Toll free over premium is 1.06 cent per minute, the most recurring issues are DTMF tones and passing caller ID. [ib] caller id problems means that sometimes CID info may not be sent? dtmf means that incoming dtmf or outgoing dtmf have problems? [William] Yes the caller id over value route is not 100% reliable, dtmf outgoing. [ib] alright, thank you

---

Q:

1. How can I quickly configure voicemail to work (w/o IVR or queue yet) + customize number of rings + blocked CID straight to voicemail with different rec?
2. Fax Qs – Do you support T.38? Are all calls sent & received over TDM trunks?
3. Can I set premium or value routing based on the dialled number?

4. Can CID filters be set so that calls forwarded to my DID are treated differently than others?

5. Is there a charge for porting a number OUT of voip.ms? I've read that VOIP providers do not have to follow CRTC number portability rules, is that true?

Your support ticket has been updated:
Dear client:
1. You need to go DID numbers tab >> Voicemail. There you need to create a voice mail entry, then on Manage DIDs >> Edit DID, you can set the voicemail entry you have created. On the same link you can customize the TO of the DID number. To block a caller id, you need to refer to DID Menu >> Caller ID filtering and create a new rule.
2. Faxing is not officially supported. Many customers have reported using faxing successfully with their numbers and termination using premium and the G711U (ulaw) codec, however we can not guarantee that this will work for you. We do plan on offering virtual fax (web service) in the future, but we do not have en ETA for this feature yet.
3. You can set Premium or Value on Main Menu >> Account Settings, there you can choose the route for USA48 & Canada, Toll free numbers and International numbers, however you can only choose the route based on the dialled number when it is an international call, using the following codes 033+Country Code+number: International Value (override account setting) and 044+Country Code+number: International Premium (override account setting.
4. CallerID Filtering is a tool that let you create incoming routing rules according to incoming CallerID number. You can apply different rules to some or all of your DID(s) numbers. You can create as many filters as you want. Each filter you create can filter according to one the following criteria and change the routing if there\'s a match.
5. There is not any charge applied for porting out a number from VoIP.ms, only to port in a number.

If busy signal!

I have the PAP2T-NA and voip.ms. To get inbound calls to ring to your PAP2T instead of giving a busy signal you should check you main DID settings on your voip.ms account. By default it is set to IP-PBX but should be changed to ATA Device.

Good tip except it's in the Account Settings or Manage Sub Account area, not DID settings

Simultaneous rings (like Google Voice)

VoIP.ms does indeed have simultaneous ring - they call it "Ring Groups". It's in the "DID Numbers" menu.

If you want to ring some other phone such as a cell phone you must first set up a Call Forwarding entry to the cell phone. Note that
you will be billed for both legs of the call (unless you have a flat rate DID plan in which case you will only be billed for one leg.)

I will be adding to the above collection my other conversations with tech support as well.

Note that most VOIP providers, though they may be offering voice support, prefer web chat support, as that frees the ATA to have its configuration changed with no interruption in the conversation. Mango however, reports that this should not be an issue:

We called our internet provider one day to have them make a change on our account. We used the VoIP line, not thinking that in order to make the change, they would need to reboot our modem. Which they did. Which caused every device on the floor to lose its internet connection. But when the modem came back up, our call was still connected, and we finished the conversation. Colour us impressed.

Below, you can find the list of free or cheap DID numbers worldwide from backsla.sh, courtesy of Robert Siemer aka klammero. And now, once you have decided on a provider, you might want to fine-tune your ATA.