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post #61 of 2800 (permalink) Old 2009-02-27, 01:43 AM
 
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Quote:
Originally Posted by schag001
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.

Quote:
Originally Posted by schag001
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

Quote:
Originally Posted by ronepowell
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?

Quote:
Originally Posted by mot_guy
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.

Quote:
Originally Posted by aooa
- 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.

Quote:
Originally Posted by aooa
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.

Quote:
Originally Posted by aooa
- 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.

Quote:
Originally Posted by aooa
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.

Quote:
Originally Posted by apn
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.

Quote:
Originally Posted by apn
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.

Quote:
Originally Posted by aooa
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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post #62 of 2800 (permalink) Old 2009-02-27, 06:25 AM
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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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post #63 of 2800 (permalink) Old 2009-02-27, 11:15 AM
 
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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.

Last edited by Mango; 2009-02-27 at 12:25 PM.
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post #64 of 2800 (permalink) Old 2009-02-27, 02:41 PM
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Quote:
Originally Posted by apn View Post
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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post #65 of 2800 (permalink) Old 2009-02-27, 02:45 PM
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Quote:
Originally Posted by aooa View Post
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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post #66 of 2800 (permalink) Old 2009-02-27, 02:49 PM
 
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Quote:
Originally Posted by 99semaj
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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post #67 of 2800 (permalink) Old 2009-02-27, 04:36 PM
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Quote:
Originally Posted by aooa View Post
- 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.

Quote:
Originally Posted by Mango View Post
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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post #68 of 2800 (permalink) Old 2009-02-27, 06:44 PM
 
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Quote:
Originally Posted by 99semaj
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"?

Quote:
Originally Posted by 99semaj
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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post #69 of 2800 (permalink) Old 2009-02-27, 08:49 PM
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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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post #70 of 2800 (permalink) Old 2009-02-27, 09:05 PM
 
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Quote:
Originally Posted by aooa
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 Congrats!

Quote:
Originally Posted by aooa
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.

Quote:
Originally Posted by aooa
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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post #71 of 2800 (permalink) Old 2009-02-27, 09:17 PM
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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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post #72 of 2800 (permalink) Old 2009-02-27, 10:08 PM
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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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post #73 of 2800 (permalink) Old 2009-02-27, 11:44 PM
 
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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
http://www.netphonedirectory.com/pap2_dialplan.htm

Quote:
Originally Posted by aooa
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/

m.

Last edited by Mango; 2009-02-28 at 12:48 AM. Reason: I suck at bbcode.
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post #74 of 2800 (permalink) Old 2009-02-28, 01:39 PM
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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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post #75 of 2800 (permalink) Old 2009-02-28, 02:06 PM
 
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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.

Last edited by Mango; 2009-02-28 at 02:10 PM. Reason: REASON? REASON?! WE DON'T GIVE NO STINKIN' REASONS!!
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