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742K views 3K replies 244 participants last post by  Satellite97 
#1 ·
Thanks to some of the other threads on this forum, I decided to try and set up my own voip, and signed up with voip.ms. I managed to set up a PAP2T in about 20 minutes, and, amazingly, I was calling across the country immediately. So far, voice quality is excellent. We'll see how it goes over a period of time and I will give an update in six months or so. Assuming all goes well, it begs the question --'Why doesn't everyone do it this way"?
 
#2 ·
After one week, everything is going great. I should point out that the voip.ms configuration examples were very minimal, and I did use the examples from callwithus, and callcentric to set up the PAP2. The callcentric instructions for starting the configuration were particularly helpful.
 
#3 ·
I use them for a while, and it's all good. They are very reliable, especially compared to those $9.95/mo services we all know about.

They recently added very powerful routing features, so you can automatically manage your calls based on date, time, and / or caller id. Unlimited DIDs are (IMHO) overpriced, but PAYG DIDs priced very well.
 
#5 ·
I like them a lot. You'll get a lot more support from Vonage (it's more consumer-oriented) but this stuff is not that difficult. When I have had technical issues I've gotten reasonably fast answers.

My only problem with them is that they have a paucity of Canadian rate centres available. For example, you can't get a number anywhere in Saskatchewan.

If the rate centres they have are fine with you, use them and enjoy them.

I do find the premium rate termination is of significantly higher quality than the budget rate termination. The rate is still low.

The new callback service is really great. I have Canada-wide My5 on my Rogers account and got a Calgary DID to use with it. If I call my Calgary number, voip.ms will call me back and give me a dialtone. For a total of 2.5 cents a minute US I can call anywhere in North America and most of western Europe.

les.net is also worth considering. Slightly better support, slightly lower quality termination, but still good. les.net has Regina and Saskatoon numbers, which makes it more useful for me.
 
#7 · (Edited)
PhotoJim,

I was looking at vbuzzer but it didn't look all that professional so I came down to a decision between les.net and voip.ms as well. Perhaps you could answer a few questions about Les, their website is pretty sparse and crappy looking so it didn't provide a lot of info.

Do they allow you to select a preferred DID# of your choice? I think voice mail comes free. Do they allow for call waiting on the incoming calls?

Thanks.
 
#8 ·
Hi Bayroy,

Here is what I know about les.net:

- all prices are in $US
- voice mail and call waiting are free
- you can pick your own DID from a list of available DIDs. (This is pretty cool and as a result I've got one DID with a really fantastic phone number.)
- taxes are extra. They just charge me GST because I'm not in Manitoba.
- there aren't any hidden fees I'm aware of.
- I don't think balances expire within reason. I'm sure at some point they must, but it would be a year plus, if at all. (My speculation here.)

Jim
 
#9 ·
First Impressions of VoIP.ms ...are great!!

I am in the process of porting away from a spammer on this site to VoIP.ms. Here are my first impressions so far:

Setup was super fast. Within a minute of payment, I was talking on my phone. They had to set up my DID manually but did it in less than an hour even though they advertised up to four.

VoIP.ms is a geek's dream. There are a seemingly endless array of configuration options that I can play with to my heart's content. I think about the only step up from this would be actually having my own Asterisk box. It's all very well explained (easier to learn than Asterisk!) so that non-geeks should enjoy it too.

Two things in particular wowed me about VoIP.ms: the first is the diagnostics available on their website. You can tell whether or not your extensions are registered, what IP they come from, and when the next registration is expected. My last provider didn't have anything close to that. If you had a problem, you just had to guess at it :(

The second thing that wowed me is that VoIP.ms has SEVEN severs around the United States, Canada, and even the UK. This means that if I had a problem with a server, I could simply switch to another server in a minute or less. Or, I could set up extensions on multiple servers and have one failover to the other. I can also pick any server I want based on the one that has the best latency to my location.

There are three rate options for termination (outgoing calls):
- Special Rates, about $0.0045/minute to Canada. I found the call quality here too poor to be of use.
- Value Rates, $0.0049 to $0.0114/minute to Canada; $0.0105/minute to the US. Call quality was not as good as Premium Rates, but very usable. (And yes - that's less than half a cent per minute!) I had to set DTMF mode to Info for DTMF (buttons on a touch-tone phone) to work here.
- Premium Rates, $.0125/minute to Canada and the US. Call quality with these rates was excellent.

There are two rate options for DIDs (incoming calls):
- $0.0149/minute and $1.99/month. You can receive a theoretically unlimited number of calls.
- $8.95/month, no per-minute billing. You can receive a maximum of two calls at one time.

VoIP.ms has a 4.5/5 star rating on Voxalot (les.net has 4/5) and if things continue to go as well as they are I plan on rating VoIP.ms highly as well.
 
#10 ·
I agree with your review. voip.ms is a terrific provider. They don't have numbers in all provinces yet, unfortunately, but the quality of the service they have is really good.

I found the value rate quality was a little marginal but the premium rate quality is really good and the rates are still reasonable enough.
 
#11 ·
There is an enthusiastic review of voip.ms on another thread, so thought I may as well close this out with my last observation. After two months no problems worth mentioning. I had to switch servers at one point, but that took about 4 minutes to accomplish. All in all, if a person is willing to spend a little time investing in a learning curve, the downstream benefits are significant. Plus it is kind of fun to 'do it yourself'.
 
#12 · (Edited)
Today's update: I discovered I was unable to call lots of Canadian toll-free numbers. So, I switched to the Toronto server (I was on the LA server before) which solved the problem. However, call quality was not as good as the LA server. I discovered with a ping test that I was getting about 1% packet loss to Toronto which was coming from the last hop before the SIP server. So, I tested the Montreal server and found 0% loss. I switched to it and calls have been great ever since.

We had a really curious issue happen the other day. I used a landline to call my wife on the VoIP phone, a Linksys SPA-921. She answered and...it kept ringing. The call eventually went to voicemail. I have not been able to reproduce this. Has anyone else seen this?

One other thing: VoIP.ms tells me that they are working on getting a server up in Vancouver! Yay :D:D:D
 
#44 ·
Today's update: I discovered I was unable to call lots of Canadian toll-free numbers. So, I switched to the Toronto server (I was on the LA server before) which solved the problem. However, call quality was not as good as the LA server. I discovered with a ping test that I was getting about 1% packet loss to Toronto which was coming from the last hop before the SIP server. So, I tested the Montreal server and found 0% loss. I switched to it and calls have been great ever since.
I live near Toronto, and last night I decided to ping and traceroute both Montreal and Toronto servers. Surprise! The Montreal server had two less hops (12 vs 14) and the ping time was HALF versus Toronto. I switched everything over with the exception of the custom VM settings which tech support needs to do.
 
#13 ·
I'm thinking about switching to voip.ms to try it out. I'm currently using vbuzzer, and the quality has been great so far. I'm ony using SIP for long-distance calling as a pay-as-u-go. I'm wondering if anyone knows how good the voip.ms VALUE option compares with vbuzzer? Anyone know?

Thanks.
 
#25 ·
Had another interesting issue today. My wife (why does it always happen with her??) tried to leave a message on the VoIP phone and it cut her off after about three seconds. I have not been able to reproduce this.
I figured out this issue. My wife has a very soft voice, and the voicemail server's silence detection is overly sensitive, so it was not aware that she was still speaking :D I'm not yet sure if that's adjustable.

Are any of you guys using voip.ms during periods of high internet traffic?
As PhotoJim mentioned, QoS is typically your issue to solve, not the voice provider's. I have heard excellent things about WRT54GL routers with Tomato firmware. I personally do not use QoS on my network at the moment. Since I'm the only heavy user of the Internet, it is not a problem for me to shut down any large uploads when I want to use the phone.

1) Does the voip.ms "premium" rate guarantee that you're on the G.711 codec @ 64kbps?
No, call routing is something different from codecs. You need to specify the codec you would like to use in the configuration of your SIP device. If memory is correct, in the PAP2T, the setting is called "Preferred Codec". You may also need to turn on "Use Preferred Codec Only" to force G.711 at all times.

3) Does voip.ms provide automated monthly (credit-card) billing?
It's pay-as-you-go, so you deposit a sum of money into your account (minimum $25) and it uses it until it's gone. You can have the system send you an email when your account gets low.

5) This looks like a pure voip play, so I'm assuming that above and beyond caller ID, there are no telephony features like call-waiting or voice mail. Is that correct and if so, what do you do for voice-mail?
This question was already answered before I got to it. However, I thought I'd chime in and say that one nifty thing about VoIP.ms that my previous provider didn't support was the creation of SIP URIs that point to a voicemail account. "I'm sorry, Bob is not in the office. Would you like his voicemail?"
 
#15 ·
Your wife could be doing something at the same time that she is not aware of, which might be causing the hang-ups. Does she use a high wattage steamer nearby or something? :D

I was using a high wattage steamer to iron my shirts and the device draws a lot of power and my router was on the same breaker along with other powered electronics. Caused the breaker to trip... :D

In another room with all the multi-lights switched ON, the lights dimmed... :D
 
#16 ·
Are any of you guys using voip.ms during periods of high internet traffic?

The reason I ask is that I sometimes work from home and have received feedback that my voice quality is poor (Rogers VOIP) with frequent break-up/delays when network activitiy is high e.g. NetMeeting for presentations.

I'm looking for a new (long-term) VOIP provider and considering Vonage or voip.ms, but based on the above, (voice) QoS is important to me.

A few more questions;

1) Does the voip.ms "premium" rate guarantee that you're on the G.711 codec @ 64kbps?

2) Do you have to buy an Asterisk gateway or can any old e.g. Linksys PAP2 work? Do they provide a list of compatible hardware?

3) Does voip.ms provide automated monthly (credit-card) billing?

4) In addition to the $1.99 (per min plan) or $8.99 (flat-rate plan) are there any other billing surprises aboive and beyond usage e.g. System Access Fee, e911 etc.?

5) This looks like a pure voip play, so I'm assuming that above and beyond caller ID, there are no telephony features like call-waiting or voice mail. Is that correct and if so, what do you do for voice-mail?
 
#18 ·
There is voicemail and there is call waiting.

As for the QoS issue, because you use your own hardware that is your issue to solve, unfortunately. Some ideas:
- increase your upload bandwidth
- ensure you are using a router that supports QoS and ensure it is properly configured to give your VoIP phone, ATA or softphone higher priority than other traffic
- adjust the configuration of NetMeeting to use less upload bandwidth

I use an unlocked Linksys PAP2 myself. I have les.net on line 1 and voip.ms on line 2. Any hardware that supports SIP or IPX will work. The PAP2 is one of the devices they recommend, and they are available inexpensively.

There are no other fees other than the per-minute rates you may pay. There is a 911 fee but only if you configure 911 services. Since it's designed to be a portable VoIP service, configuring 911 doesn't always make sense. Name display is an option that is charged at an additional penny or so per call.
 
#19 ·
I have not seen much of anything else on their website. Usually the parameters are the same though from one device to the other. Especially if it is a Linksys/Supira device.

I recently became a member of voip.ms and the sound quality is excellent! Much better than CallCentric. I find it very similar to PSTN phone service as far as sound quality goes. Of course this means I'm using their premium service, not the low-cost one. For out-going calls you can pick and choose (value or premium quality at a cost). For incoming calls they are always routed through the premium network. I used a D-Link router with a Linksys SPA9000 VoIP device. I had no trouble getting it to work within 5 minutes.

On Wednesday of this week I purchased a new "N" router - the Linksys WRT610N. I replaced my D-Link router with this one. I'm disappointed because for some reason the WRT610N will not allow me to use voip.ms. I can hear the caller, but they cannot hear me. I spent over an hour online with the technical people at voip.ms last night and they were very patient and helpful. They basically advised me to set the following:

- Port 5060 UDP to my static IP-Address of the SPA9000.
- Ports 10000 to 20000 UDP to the same static IP.

We could not even get the SPA9000 to register after this was done. The reason is because there are 4 ports on my SPA9000, and I had Voip.ms on port 3. So the first port I should have opened was 5062. Once I did that I was able to register my SPA9000 with voip.ms. However, I cannot transmit. I can hear, but the calling party does not hear me. Very strange. The technical person went over a few more things with me. We even disabled the firewall on the WRT610N router, but still no go. Yet the D-Link router works. I made sure I had the latest router firmware. I re-booted several times to ensure my changes were being picked up. The only thing I can think of is that there must be some other port(s) that need to be opened. I don't know what else it could be.

If anyone else has experienced this problem and managed to fix it I'd sure love to hear from you. I could not find anything when doing my searches. If I cannot get this fixed I will be returning my new router. I'd rather keep it because it's a nice router with strong wireless reception. I think it just has a large amount of security and I don't know how to get around it.

Thanks.
 
#20 ·
I signed with them in late August for a number of reasons:

The service and support with Unitz was horrendous and I put up a year and a half with them through Teksavvy.

They seem an upcoming company with many new features coming, and I liked their price. My first choice was Callcentric but my number was not portable.

Porting my number with voip.ms did take 7 weeks and not sure that is still an issue.

But I still do have issues with the performance. I get a small outgoing number of calls that drop after a few seconds and callers feel they are talking in a tunnel. My stability Teksaavvy DSL could be casuing this. I'd like to get to the bottom of this.

On my end the sound quality is crystal clear, but before fully endorsing them, I need to sort these issues. Nothing like the problems with Unitz though.

I can suggest a voip.ms forum but please PM me as I could be accused of spamming. :eek:
 
#22 ·
No, at the time I signed up there was no "demo" version of this software. They basically have a minimum deposit of $25 US when you join. They charge taxes on the $25. Then they withdraw your service from that money. If you obtain a DID, they will charge you $1.00 service fee which is pretty low. I basically use the service for incoming calls, and the quality is awesome.

The only thing I wish is that they had a plan that offered unlimited Canada/USA calling. It's pay-as-you-go. For me it works because I use another line for outgoing calls, and when people call me they use my voip.ms line.
 
#23 ·
Sounds really good, and I think I'll give it a whirl!

The website's a little sparse, though. Is there a page that lists the features you get with your line? PhotoJim mentioned VM and Call Waiting - what else do you get?

I would like to try voip.ms with a softphone, primarily to check quality before plunking down money for a hardware SIP device. Should I expect any quality difference between a softphone and, say, a Linksys SIP device?
 
#24 ·
The nice thing about the SIP protocol is that it bypasses you computer because the Analogue Telephone Adapter (ATA) is connected directly to your router. In some cases, the ATA is the router itself.

Bypassing your computer provides a direct connection to the internet and prevents any complication associated with your computer bogging down.

Services like Skype rely on your computer to process the call, so if your CPU decides to take a break, your call gets dropped. So - the call quality with a ATA adapter should be better than any softphone.

I've been a subscriber to voip.ms for a month. So far I'm impressed.

I have a Linksys SPA3102 - which is connected to my conventional PTSN home line as well. The unit's dial plan is configured so all long distance calls (typically starting with "1" or "011) get routed through VOIP.ms, while all local calls get routed through my PSTN connection. All my inbound calls come through PSTN.
 
#26 ·
VOIP.MS Questions

Hi all,

I signed up for VOIP.MS yesterday and am testing out the features and call quality. So far so good pretty much.

Couple questions for any of you VOIP.MS experts...

1. Inbound caller ID.. I have it set on the customer portal for the DID to enable caller id with the small fee listed. I get a name when a call comes in but it either says Cell Phone, or British Columbia for example. It doesn't show the actual name of the user from telco database. Is this a known issue or would I need to set something else up? I am using Linksys/Sipura SPA-2102 as my ATA device.

2. To check voicemail, the say use *97 + mailbox number. I am not able to dial that (maybe ATA Setting). I can *98, and then am asked for mailbox number and password. Is there a way to have *98 actually enter the required mailbox number to avoid having to enter it every time, or program the ATA to allow *97 + mailbox as shortcut that they mention?

Working my way through it, but help from those who might have hit these issues appreciated. Thx.
 
#27 ·
Thanks for the replies, guys. I've been away for a couple of weeks and happy to see active discussion here.

Since posting I've learned that my voice-quality issues on Rogers occur regardless of whether their VOIP Router or my WRT54GL/DDWRT sits behind the cable modem. In the former case, I don't recall finding any QoS settings, but did setup the Linksys to prioritize the uplink for the VOIP protocols.

zoidberg: I'm curious if you're keeping the PSTN until completely satisfied with VOIP options?

Currently Rogers VOIP is my home phone service, but I'm tired of paying $50+/mo when our usage pattern fits within the Vonage basic $20/mo plan.

My wife and I both have mobiles, so for home service I just want reliable, good voice-quality service at the lowest cost. The only reason I've not already signed w/ Vonage is the minimum 2yr committment.

I've heard good things about Vonage, but I'd rather explore other options before getting locked-in.

From what I've read above, voip.ms continues to sound attractive, although I'd like to see an automated (prepaid) top-up option.
 
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