voip.ms discussion - Page 3 - Canadian TV, Computing and Home Theatre Forums
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post #31 of 2800 (permalink) Old 2009-02-04, 02:34 PM
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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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post #32 of 2800 (permalink) Old 2009-02-04, 11:52 PM
 
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Quote:
Originally Posted by 99semaj
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.

Quote:
Originally Posted by 99semaj
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.

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Originally Posted by 99semaj
Aastra 9112i
WANT!!!!!! [wipes drool off keyboard]

-Mango
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post #33 of 2800 (permalink) Old 2009-02-09, 04:11 PM
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WANT!!!!!! [wipes drool off keyboard]

-Mango
Picking up my first 9112i tonight. If I like it, I'll probably get two more in pretty short order.
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post #34 of 2800 (permalink) Old 2009-02-10, 08:08 PM
 
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How much does this IP phone costs?

Panasonic TH50PX60U / Denon AVR-990 / 8642HD PVR/ Panasonic DVD-S77 / Pioneer DV-563A / Playstation3 / Belkin PF60
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post #35 of 2800 (permalink) Old 2009-02-11, 01:07 AM
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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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post #36 of 2800 (permalink) Old 2009-02-12, 01:36 PM
 
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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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post #37 of 2800 (permalink) Old 2009-02-13, 01:26 PM
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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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post #38 of 2800 (permalink) Old 2009-02-17, 01:43 AM
 
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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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post #39 of 2800 (permalink) Old 2009-02-17, 09:57 PM
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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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post #40 of 2800 (permalink) Old 2009-02-18, 12:57 AM
 
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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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post #41 of 2800 (permalink) Old 2009-02-18, 12:06 PM
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Hi Lawman:

You raise good points, and I did use an ATA for a full five-years before opting for this route.

For me, the main advantage is the number of channels. With an ATA, you are bound by the old telco paradigm that a line can be "busy" if any extension is in use.

With a pure VOIP solution, any free extension can recieve a call or make a call regardless of what the other sets' status is. A family of four could make four outgoing calls, recieve four incoming calls, etc, without every messing around with call waiting.

Also, there's some utility in having station-to-station calling, but that's pretty minor in a small home like mine.

WRT to the alarm system, I have an IP communicator on mine, so telephone doesn't play into the equation.
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post #42 of 2800 (permalink) Old 2009-02-18, 02:53 PM
 
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Originally Posted by 99semaj View Post
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.
This is perfect - thanks!

m.
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post #43 of 2800 (permalink) Old 2009-02-18, 04:35 PM
 
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True if the line is busy then extension is in use and no one can use it. That's why I set up another ATA within the house. One ATA for lower floor and another ATA for upper floor. Both uses the same ports as each ATA is a different MAC addr. So, if someone upstairs uses the phone, another person downstairs can also call out.

I like your setup - down the road, if money is not a problem, I would not mind trying it out.

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post #44 of 2800 (permalink) Old 2009-02-21, 01:50 PM
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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.
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post #45 of 2800 (permalink) Old 2009-02-22, 02:21 PM
 
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Quote:
Originally Posted by holymoly
If you are porting to voip.ms it takes longer for Canadian customers. Mine took eight weeks back in August.
My port hit eight weeks yesterday and I don't even have a FOC yet. I've also been waiting on a DID assignment that hit six weeks on Wednesday. My first DID was assigned in a few hours, and I suspect the length of time on this one is a casualty of the issues VoIP.ms was having with their ticket system. Anyway, they're telling me I should be able to use it by Tuesday of next week which will be one day short of seven weeks.

Quote:
Originally Posted by 99semaj
Surprise! The Montreal server had two less hops (12 vs 14) and the ping time was HALF versus Toronto
Heh. Curious the way internet routing sometimes works. One thing I like about VoIP.ms is the fact that we have choice in the server we connect to. The packet loss issue I was having wasn't VoIP.ms' fault as it wasn't their server causing the packet loss, and I didn't have packet loss with another ISP with different routing. Not their fault, not my fault, and nothing either of us can do about it except wait. In this case it took about a week and a half for the problem to be sorted, but I didn't care because I just used a different server.

m.
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