voip.ms discussion - Page 2 - Canadian TV, Computing and Home Theatre Forums
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post #16 of 2800 (permalink) Old 2009-01-08, 04:00 PM
apn
 
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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?
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post #17 of 2800 (permalink) Old 2009-01-08, 05:40 PM
apn
 
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Other than the PAP2T and "Asterisk", does voip.ms provide a list of supported devices and standard configurations?
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post #18 of 2800 (permalink) Old 2009-01-09, 11:12 AM
 
Join Date: Dec 2003
Location: Regina, SK, CA
Posts: 620
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.
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post #19 of 2800 (permalink) Old 2009-01-16, 03:12 PM
 
Join Date: May 2006
Location: Gatineau, QC
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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.
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post #20 of 2800 (permalink) Old 2009-01-16, 04:36 PM
 
Join Date: Apr 2005
Location: Mississauga, Ontario
Posts: 179
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.
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post #21 of 2800 (permalink) Old 2009-01-16, 07:17 PM
 
Join Date: Jun 2006
Location: Ottawa, Rogers
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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?

Panasonic TH50PX60U / Denon AVR-990 / 8642HD PVR/ Panasonic DVD-S77 / Pioneer DV-563A / Playstation3 / Belkin PF60
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post #22 of 2800 (permalink) Old 2009-01-16, 11:07 PM
 
Join Date: May 2006
Location: Gatineau, QC
Posts: 1,333
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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post #23 of 2800 (permalink) Old 2009-01-20, 02:50 AM
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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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post #24 of 2800 (permalink) Old 2009-01-20, 10:51 PM
 
Join Date: Jan 2007
Location: Mississauga, ON
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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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post #25 of 2800 (permalink) Old 2009-01-23, 02:44 PM
 
Join Date: Nov 2008
Location: Alberta
Posts: 867
Quote:
Originally Posted by Mango View Post
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 I'm not yet sure if that's adjustable.

Quote:
Originally Posted by apn View Post
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.

Quote:
Originally Posted by apn View Post
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.

Quote:
Originally Posted by apn View Post
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.

Quote:
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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post #26 of 2800 (permalink) Old 2009-01-30, 04:09 AM
 
Join Date: Jun 2006
Location: YVR
Posts: 22
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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post #27 of 2800 (permalink) Old 2009-01-30, 09:56 PM
apn
 
Join Date: Jul 2007
Posts: 569
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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post #28 of 2800 (permalink) Old 2009-01-30, 10:25 PM
 
Join Date: Jun 2006
Location: Ottawa, Rogers
Posts: 555
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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post #29 of 2800 (permalink) Old 2009-01-31, 09:54 AM
apn
 
Join Date: Jul 2007
Posts: 569
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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post #30 of 2800 (permalink) Old 2009-02-04, 01:16 AM
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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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