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Were you able to test your voip phone calls with voip.ms prior to signing up and putting money with them?
 

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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.
 

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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?
 

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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.
 

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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?"
 

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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.
 

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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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I signed up with voip.ms after a trying out vbuzzer. I like both of them so far. Good to have many options. Voip.ms has nice features, and I like the quality of the LD phone calls so far as a paygo setup.
 

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So based on what I've read here and at dslreports, I'm taking the plunge w/ voip.ms

I purchased a PAP2T this morning, so once that's delivered next week, I'll be setting up an account w/ voip.ms.

The only potential glitch I see ahead is that my wife wants to keep our number, so it may take a few weeks to get it ported, and since we're going from one VOIP provider to another, I assume that we'll have to setup everything on the very day the number is ported.

At least this ensures that we have service until transition day and it means I don't have to consider introducing the PAP2T to my Rogers VOIP box.

I purposely went with a Linksys ATA since they're inexpensive, ubiquitous, I don't need the FXO capability and finally, I figured it will smoothly integrate with my WRT54GL/ddwrt device.
 

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newbie question:

If I have a voip.ms account configured with one DID, can I have multiple IP phones (i.e. Aastra 9112i) that will behave like analog sets? (i.e. shared call capability which all ring at once, multiple sets can talk on the same line at once?)
 

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second newbie question...

Do I need to use an asterix server to get other call features like call forwarding and conferencing? How about visual caller ID?
 

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99semaj said:
If I have a voip.ms account configured with one DID, can I have multiple IP phones that will behave like analog sets? (i.e. shared call capability which all ring at once, multiple sets can talk on the same line at once?)
You can set up ring groups, however they won't act like shared line appearances. Once you answer a call with an IP phone, the way another phone would join it is for the first phone to conference or transfer the call. If you pick up another IP phone without transferring or conferencing it in, you get a dial tone. If you want shared line appearances, you would want to use an Asterisk server.

99semaj said:
Do I need to use an asterix server to get other call features like call forwarding and conferencing? How about visual caller ID?
No, those features do not require an Asterisk server.

99semaj said:
Aastra 9112i
WANT!!!!!! [wipes drool off keyboard]

-Mango :D
 

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How much does this IP phone costs?
 

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About $100, and I find it works quite well, after some fiddling to understand what the settings meant. I will test it on some conference calls tomorrow for terminating quality, and hopefully if all goes well I will get my DID ported over in a few weeks.
 

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If you are porting to voip.ms it takes longer for Canadian customers. Mine took eight weeks back in August.

Not sure how it is now
 

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9112i on voip.ms

This set is working very well...I picked up another three sets today and will convert my whole house to pure VOIP (no ATA like Vonage or Primus).
 

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Glad to hear the 9112i is working well for you. Did you end up setting up an Asterisk server? I've never actually set up shared call appearances and am curious as to how easy it was.

If it's not too much trouble, could you post some screenshots of the admin interface?

-m.
 

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No, I skipped the asterisk server at this point, although I do have a couple of Macs in the network, so it certainly wouldn't be hard to do.

I can post pics if you want, but I think the administrator's manual would be much more useful since it actually explains everything along with the screenshots.
 

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hi 99semaj,

Not sure why the IP phones were necessarily better than an ATA. If you bought 4 IP phones for $99 each, it would mean ~ $400 for making your whole house with voip.

I bought and setup a discounted ATA and connected a 3-handset cordless DECT 6.0 phone all over the house. Total cost under $90 including the cost of the ATA. All this with voip.ms and vbuzzer. If I want more handsets, I could buy additional DECT 6.0 handsets and expand my phone system to up to 6-handsets. But that is overkill.

I am keeping my Bell landline because I realized that I needed the landline as a backup, and for home security systems e.g. ADT, which requires a reliable phone line i.e. landline. A cellular based security system is more expensive if available (so this was ruled out), and some security systems do not work well with voip phone lines as reported by many users.

I believe IP hardware phones like yours are great if one has the money. The above works well for me and so far, I'm pretty happy with my low-cost setup. I like yours as well, but its costlier.
 
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