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: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.
I agree! I wouldn't touch Netfone with the proverbial 39 1/2 foot poleschag001 said:netfone is not on my list after they dropped the ball on me before I even got started with them.
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?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'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.mot_guy said:Also for e911, is it a US customer only feature or is it available to Canadian customers?
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:- if I want to continue using my existing regular telephones, I need to get one of those Voice Gateway/routers.. (any recommendation)?
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.)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??
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.
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 )aooa said:- has anyone successfully used this for Home Alarm systems??
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.
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.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?
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:Voice Mail: Is voip.ms or my ATA providing this and where are the messages stored? Is there a mailbox storage (minute) limit etc?
This would depend on your ATA. My PAP2T does support this.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?
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.aooa said:Is it possible to have 1 DID # for multiple ATAs?