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schag001 said:
Wow..had a chat with VoIP.ms support and found out that call display is charged extra PER USE...
I have never seen somthing that strange.
Keep in mind that VoIP.ms also has to pay for CNAM lookups. They just chose to provide the service a la carte so that those who did not care for Caller ID wouldn't have to pay for it. And, as HDTV101 said, at 1.2 cents per lookup, it shouldn't break the bank. The enhancements that VoIP.ms is planning (cached queries, contact lists, skipping the lookup if the name is passed as SS7/IAM data) should reduce this cost even further.

schag001 said:
netfone is not on my list after they dropped the ball on me before I even got started with them.
I agree! I wouldn't touch Netfone with the proverbial 39 1/2 foot pole :)

ronepowell said:
Is anyone else having problems with server dropout? I seem to be changing servers from montreal to toronto, and back again, on fairly regular intervals -- most recently this Monday. What might be causing this problem?
I have not experienced this on either server. However, a few weeks ago, I experienced some sort of routing issue to the Toronto server in which about 1% of packets were lost. Oddly, this only happened on one of my ISPs, not both, which made me assume routing issue rather than server problem. What happens if you do a traceroute while the problem is occurring?

mot_guy said:
Also for e911, is it a US customer only feature or is it available to Canadian customers?
I'm in Vancouver and the system allowed me to order e911, though it hasn't yet been activated. I'll post back when I find out for sure.

aooa said:
- if I want to continue using my existing regular telephones, I need to get one of those Voice Gateway/routers.. (any recommendation)?
We typically call the device you're thinking of an ATA or Analog Telephone Adapter. The one I see mentioned most often is the Cisco PAP2T. I have one and I like it a lot.

aooa said:
Is it really that easy? If so, I guess the only advantage of going with Primus or Vonage (and paying more $/mo) is the support for the device and not having to purchase devices??
Not quite. You haven't yet taken into account bandwidth sharing. Because VoIP travels over the internet, any time you do anything bandwidth intensive with your internet connection you reduce the amount of bandwidth available for VoIP. If you're a light internet user, you might not have a problem. Otherwise, you would want to use a router that supported Quality of Service such as a WRT54GL with Tomato firmware. (Google for more info.)

Additionally...I haven't used Primus TalkBroadband since about 2005, but at the time their call quality, technical support, porting process, and finally cancellation process was ATROCIOUS. Maybe they've improved since then, or maybe I was just unlucky. In any case, Primus will never get one more cent of my money, ever.

aooa said:
- has anyone successfully used this for Home Alarm systems??
Using VoIP for alarm systems is technically possible but not always reliable. The reason for this is that alarm systems, modems, and fax machines are much more sensitive to latency (delay between you and the VoIP server), jitter (variation in latency), and compression (lowering quality in an effort to conserve bandwidth) than a voice conversation is. If you decide to cancel your analog line, I would recommend upgrading your alarm system to an IP-based system like 99semaj has or a system with cell backup. (WRT = with regards to :))

I spoke to one of the techs at my alarm company who said that if I did decide to use VoIP for the alarm system, I would need to sign a contract saying that I understood that VoIP was less reliable than an analog phone line. One other interesting thing the tech said - completely without my prompting - was that he's never been able to make an alarm system work with Primus TalkBroadband, ever. He said the call quality just wasn't there.

aooa said:
If I'm understanding [termination and DID rates] correctly, why is there 2 options for outgoing quality and only 1 option for incoming? What is the quality of the incoming call?
You're correct. My guess is that the termination provider controls the routing, so VoIP.ms has no say over how an incoming call should be routed. The quality of the route would also depend on the termination provider. If someone called you from an analog phone, the quality would be similar to that of the premium route.

apn said:
Voice Mail: Is voip.ms or my ATA providing this and where are the messages stored? Is there a mailbox storage (minute) limit etc?
That's up to you. Very few ATAs have a voicemail server, although the odd one does. Personally, I use VoIP.ms' voicemail and store my messages on their server. That way, if my internet connection is down, all my calls will go to voicemail. I haven't hit a storage limit.

apn said:
VMWI: I recall reading something about voip.ms and/or my ATA providing stuttered dialtone. My (Nortel/Panasonic) phones also have the (CLASS) VMWI feature to flash the msg waiting light. Is this an option on voip.ms?
This would depend on your ATA. My PAP2T does support this.

aooa said:
Is it possible to have 1 DID # for multiple ATAs?
It is possible for one DID to be routed to multiple ATAs. The way you would do this with VoIP.ms is set up a sub account for each ATA, place each sub account into a ring group, and then route the DID to the ring group. Keep in mind you can connect multiple analog phones to a single ATA, you do not require one ATA for each phone.

*phew!* done!
 

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Fantastic post, Mango!
Thanks for taking the time to compose/post that. It's helped a lot of people.

To use the poker term, I'm all in... planning on funding and requesting my DID port later today.
 

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Awesome :) Keep us updated!

Edit: I forgot to answer the question about visual call waiting. VoIP.ms does support it. As with most features, your ATA must also support it. The PAP2T does, but I had a hard time figuring it out. Here's how to do it:

1) Connect to the web interface of your PAP2T using the Admin Login.
2) First go to the Line 1/2 tab. Be sure that "Call Waiting Serv" is set to "yes".
3) Next, go to the Regional tab. You need to set up four activation codes. If they're already set up, then that's fine, just make a note of them. If there is no code listed, make one up (that is not already in use on that page) and type it in. The four features that require activation codes are: "CW Act Code", "CW Deact Code", "CWCID Act Code", and finally "CWCID Deact Code". Note that these codes must all be different. I used *56, *57, *58, and *59. It doesn't matter what you use as long as you remember it, and as long as the code is not already in use for some other feature. Save your changes and wait for the device to restart.
4) Pick up your telephone and dial your Call Waiting Activation Code. Wait for the dial tone and then dial your Call Waiting Caller ID Activation Code. Visual Call Waiting is now ready for use.

Take a look at the other features on the Regional tab and see if you would like to use any of them. For example, you may also set up an activation code to deactivate Call Waiting for a specific call.

Hope that helps,
m.
 

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I've been testing voip.ms termination for a couple of weeks (using PAP2T-NA) and I'm very happy with the results.

I was already thinking about DID porting, but Rogers' announcement to drop its VOIP offering (4/22/09) is going to accelerate my decision. However, before taking the plunge, I have some (noobie) questions about voip.ms DID services;

1. Voice Mail: Is voip.ms or my ATA providing this and where are the messages stored? Is there a mailbox storage (minute) limit etc?

2. VMWI: I recall reading something about voip.ms and/or my ATA providing stuttered dialtone. My (Nortel/Panasonic) phones also have the (CLASS) VMWI feature to flash the msg waiting light. Is this an option on voip.ms?

3. SCWID aka Visual Call Waiting: My phones can also display the CallerID of incoming call-waiting calls. Is this an option on voip.ms?
Voicemail is done by voip.ms It ain't the prettiest setup I've seen though

VMWI is supported.

SCWID is also supported.
 

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Lawnman,

newbie Q: Does each of your ATA have their own DID #... ?? Is it possible to have 1 DID # for multiple ATAs?


99Semaj, same question... does each VOIP phone have their own DID # or are all configured for the same DID #? Also, what do you mean by "WRT to alarm system"?


Thanks
I don't use an ATA; I have acutal IP terminals, so my reply may not be useful.

In my case, though, they are all configured to the same DID using the ring group functionality. They could have individual DIDs if I chose.

WRT=with respect to.
 

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99semaj said:
Voicemail is done by voip.ms It ain't the prettiest setup I've seen though
Why do you say that? As far as I've been able to tell, it works just like any other hosted voicemail system I've ever used, with the useful addition of voicemail-to-email.
 

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- has anyone successfully used this for Home Alarm systems??
I can speak to this one with authority since I manufacture alarm systems in various markets globally.

Ironically, the older pulse formats work best with VOIP connection, and they work flawlessly. (These are similar to the old dial-style phones, and transmit more slowly than more modern alarms and cannot send as many messages)

FSK formats work equally as well on stable VOIP lines. FSK is best characterized as sounding like a fax or modem....that "white noise" sound. SIA is a common name for this method.

The most problematic has been DTMF (touchtone, etc) formats. Honeywell/ADEMCO Contact ID uses this. VOIP systems just seem to want to handle touchtones differently, and the speed and length of each tone causes problems.

Why do you say that? As far as I've been able to tell, it works just like any other hosted voicemail system I've ever used, with the useful addition of voicemail-to-email.
The access code to dial in is almost impossible to remember!
 

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99semaj said:
FSK formats work equally as well on stable VOIP lines. FSK is best characterized as sounding like a fax or modem....that "white noise" sound. SIA is a common name for this method.
So you don't find that this format is affected by jitter and, say, 1% packet loss? Or is that what you meant by "stable"? ;)

99semaj said:
The access code to dial in is almost impossible to remember!
Why not add it to your dial plan? <*98:*97[mailbox number]>S0 should do the trick :)
 

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I took the plunge!

I got myself a SPA2102 and a voip.ms account! Got the basic stuff setup in 10-15min.. thank you all for your very informative replies! they have helped me very much!

One thing I noticed is that when I'm making an outgoing call (value or premium) after dialing the #, there's about 10-13sec pause before I hear any ringing or when I get response when trynig to do echo or 811 test.. anyone else experiencing this?? I'm working with online support now..

Also with the SPA2102.. it says to ONLY connect phones/fax to the phone port of the device and NOT to plug a phone jack into it (or it might damage the wiring??).. I was thinking of hooking up the SPA2102 in the basement so all my jacks will be tied to it.. I live in a 2yr old townhouse.. is there a way for me to do this?
 

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aooa said:
I got myself a SPA2102 and a voip.ms account! Got the basic stuff setup in 10-15min.. thank you all for your very informative replies! they have helped me very much!
I am very glad to hear that :D Congrats!

aooa said:
One thing I noticed is that when I'm making an outgoing call (value or premium) after dialing the #, there's about 10-13sec pause before I hear any ringing or when I get response when trynig to do echo or 811 test.. anyone else experiencing this?? I'm working with online support now..
That's probably due to the dial plan - I believe your device ships without a North American dial plan for some reason. I'm sure their support will be able to help you out with that. If not, I'll send you the one I use. I'm at the office right now but I can get it later tonight for you.

An alternate solution is to press the # sign after dialing to have the call go through immediately without waiting for further dialing.

While we're on the topic of incorrect ATA settings, be sure to change your RTP packet size from the default of 0.03 to 0.02.

aooa said:
Also with the SPA2102.. it says to ONLY connect phones/fax to the phone port of the device and NOT to plug a phone jack into it (or it might damage the wiring??).. I was thinking of hooking up the SPA2102 in the basement so all my jacks will be tied to it.. I live in a 2yr old townhouse.. is there a way for me to do this?
The reason why they say this is that you do not want to connect the ATA to your household wiring if you are still connected to the telco. If at any time your phone company decides to send voltage down the line, (ring voltage is quite high I hear) it would likely brick your ATA.

If you want to hook up the ATA in the basement, be absolutely sure that you've disconnected the telco's wiring and you should be fine. One other thought. Most homes are wired with at least two pairs of wiring. You could run the telco on one pair and VoIP on the second pair. If you need more direction, post back and I'll help you out if I can.

Edited to say: if your alarm system is connected to your analog phone line, and you disconnect your phone line, be sure to call your alarm company after you've got your ATA hooked up so that they can test everything out with you to be sure everything is working as it should. Also, most alarm systems will want to "sieze" the phone line so that an intruder cannot pick up a phone and interrupt the alarm system's transmission. If you are running your alarm system over VoIP, your wiring should be: ATA->Alarm system->telephones.

m.
 

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as Barney would say AWESOME! you're right.. it's with the dial plan.. support gave me this long dial plan

(<:1416>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|[3469]11|0|00|[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|4XXX|xxxxxxxxxxxx.)

that reduced the pause by more than half... now off to google dial plans to understand what all these means! :)

Thanks for the info about the phone line! I was worried I needed to get a ATA for each phone! talk about a noob! :)
 

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Let me know if you find a good tutorial. I need to get my head around how dial plans work....then I can go back to seven digit dialling!
 

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This is the tutorial I used when I was learning - I thought it was easier than the one in the manual :D
http://www.netphonedirectory.com/pap2_dialplan.htm

aooa said:
talk about a noob! :)
Hehe, not at all. You'll be here on the forums answering questions in no time, I'll bet!

I've written a bit about VoIP on my blog. I'm not sure if it's bad form to post a personal website here, so Hugh, feel free to let me know if so and I'll remove this. In the meantime... http://www.toao.net/category/voip/ :p

m.
 

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I'm thinking about switching to voip.ms and was wondering from thouse that have been using it for a while how is voip.ms for reliability? many/any droped calls echos?
 

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No dropped calls, no echos here. I'm not sure exactly what they do for echo cancellation but whatever it is is very effective - much more so than my last provider. With my last provider, I had to tweak the volume of my phone at every call to eliminate echo. With VoIP.ms I don't need to think about it at all.

If you use an ATA, you will want to adjust your FXS Input/Output gain appropriately. Mine is set to -1/-11 and that seems to work well. I don't notice any echo at all and volume seems very appropriate.

One other thing you can do to prevent echo from occurring on your end is attempt to reduce latency (delay between you and the VoIP server). Try changing your RTP Packet Size from the default 0.03 to 0.02 or even 0.01 if you have bandwidth to spare. Also, you may want to try reducing your jitter (variation in latency) level. If your internet connection experiences a lot of jitter, this will actually add to your problems, but if it's stable it should reduce latency substantially.

One last thing is that I find that later versions of firmware do a better job of preventing echo, at least on the Cisco devices I've tried.

m.
 

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Mango, my FXS Input/Output gain is set to -3/-3 . My callers say they sometimes get echo, though I don't. Am using Linksys 2102.

Also testing Echo 4443 I get a muffled abrupt echo return. Does this mean anything?
 

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Are you hockeynomad from DLSR? I told you to fix your input/output gain two weeks ago ;)

In case you missed engineerdan's post, here it is:
http://www.dslreports.com/forum/remark,21973898

I believe you mentioned an echo cancellation card. Those do work well, but from what I've read, they're really only best used at the point where the VoIP server meets the analog phone system...in other words, on VoIP.ms' end. So I think you should be fine without one, as long as you're not causing echo by, for example, having your input/output gain too high.

4443 should be abrupt but I'm not sure what would cause it to be muffled. If your voice sounds appropriate to the people you talk to, then I wouldn't worry about it.
 

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Hey OlDrunk,

I just signed up with them yesterday... so far very pleased with the ease of setup and voice quality - I was testing both value and premium voice quality and couldn't notice a difference so went for the value..

Support is very helpful as well... this site & posters on this thread is awesome - always willing to help!
 

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Dialing Plan rules

... (voip.ms) support gave me this long dial plan;
Great topic, aooa, since I noticed the same thing with my PAP2T (I was using the factory default dialing plan). Based on the reading from the link provided, I've parsed your dial plan and found the the following;

Code:
(<:1416>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|[3469]11|0|00|[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|4XXX|xxxxxxxxxxxx.)

<:1416>[2-9]xxxxxx:	prefix 7-digit local calls w/ 1613, straight out.

1[2-9]xx[2-9]xxxxxxS0:	NA long distance dialing, straight out.

[2-9]xx[2-9]xxxxxxS0:	10-digit local dialing, straight out.

*xx:			telco/ATA feature codes.

[3469]11:		311, 411, 611 and 911

0:			Operator?

00:			2nd Operator?

[2-9]xxxxxx:		7-digit local dialing.

1[2-9]xx[2-9]xxxxxxS0:	NA long distance dialing, straight out.

4xxx:			Apartment door code?

xxxxxxxxxxxx.:		Anything goes?
It seems to me that you have a few repetitions and conflicts in there, that if removed, might further reduce your wait time.

1[2-9]xx[2-9]xxxxxxS0 (NA long-distance dialing) appears in the list twice.

[2-9]xxxxxx seems redundant, since <:1416>[2-9]xxxxxx further up the list is going to add the 1-416 prefix to all 7-digit dial patterns. Like my area, I believe TO uses 10-digit local dialing, so the [2-9]xxxxxx would only get triggered if you're into the habit of dialing 7 digits for local calls. I imagine the (non-required) '1' prefix is ignored by the switch.

If you're already in the habit of dialing 10-digit for locals, then [2-9]xx[2-9]xxxxxxS0 is already in your list and further eliminates the need for <:1416>[2-9]xxxxxx

[3469]11 seems like a good idea to handle 311, 411, 611 and 911. However, you might want to ensure that they're valid. Note that voip.ms disables 411 and 911 by default, but you can enable them in your account profile (both are paid options).

I'm guessing that 0 and 00 are operator services. 4xxx looks like 4-digit in-house (hotel/apartment) dialing, but the final xxxxxxxxxxxx. "anything goes" string is a little puzzling.

For telco features, NA 10-digit local, NA long-distance and international dialing, I'm thinking something along the lines of the following should reduce the delays we're experiencing (remember the delay is caused by the ATA processing the string as it's entered, pattern matching and waiting for extra digits to match patterns);

Code:
(*xx|[2-9]xx[2-9]xxxxxxS0|1[2-9]xx[2-9]xxxxxxS0|011[2-9]x.)

*xx:			telco/ATA feature codes

[2-9]xx[2-9]xxxxxxS0:	10-digit local dialing, straight out.

1[2-9]xx[2-9]xxxxxxS0:	NA long distance dialing, straight out.

011[2-9]x.:		international dialing
Finally, I found this little gem on the dial plan web-site, too;

Code:
1900xxxxxxx!:		block 1-900 service #'s
...which could be improved as 19xxxxxxxxx! to remove variations on the "900" theme.

I'm going to test the shorter dial plan and will post the results.
 

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apn,

yeah.. .after breaking down the dial plan I realized they had duplications.. I guess support was just trying to troubleshoot by putting local 10 digit dialing closer to the front (which worked).. couldn't find a yellow page around the house - so not sure what 311 or 611 or 00 does...

I'm playing around with dial plan myself... I like your sample code - I like the idea of keeping it simple... for me, I would probably add the 911..

(*xx|911S0|0|[2-9]xx[2-9]xxxxxxS0|1[2-9]xx[2-9]xxxxxxS0|011[2-9]x.|1900xxxxxxx!)

For the 911 - I notice the samples on the web doesn't add S0, is there any point in adding S0 or as soon as 911 match it will dial it right away and skip trying to match with the rest of the dial plan?

I'm in GTA so for me, local area codes are 416, 905, 647.. other area codes would be long distance and should require a 1 in front.. based on this, is it better to individually add 416xxxxxxx, 905xxxxxxx, 647xxxxxxx or keep it generic with [2-9]xx[2-9]xxxxxx.. guess benefit with generic is to accomodate future new local area codes..

Just curious.. if I wanted to mimic a regular POTS line that has no restriction, would it be just (x.)?
 
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