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.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.
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.Are any of you guys using voip.ms during periods of high internet traffic?
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.1) Does the voip.ms "premium" rate guarantee that you're on the G.711 codec @ 64kbps?
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.3) Does voip.ms provide automated monthly (credit-card) billing?
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?"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?
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.99semaj said: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?)
No, those features do not require an Asterisk server.99semaj said:Do I need to use an asterix server to get other call features like call forwarding and conferencing? How about visual caller ID?
WANT!!!!!! [wipes drool off keyboard]99semaj said:Aastra 9112i