Setting up a home IVR

IMHO, the Asterisk running on a Linux machine is a best option to meet all your current and, or future needs and, or requirements. Instead of an ATA (Analog Telephone Adapter), try finding a suitable card with one FXS and one FXO ports.

you can connect your pstn and voip using combine-a-line.

and it does not need a usb port and coptuer to operate savings 100's dollars in wasted energy use.

here is a link to combine a line

pstn zero enrgy use adapter avoid usb adapters

Balwinder:

Thank you for your response.

Could you please elaborate a little on why you suggested a card with FXS/FXO ports instead of an ATA (Analog Telephone Adapter)? I am unclear if you suggested because it would just be better overall, or because it is the only option to make my plan work. (That is to say would an ATA not do the things I require?)

I was looking at that Grandstream ATA because it was cheap, and I do not not need a sophesticated PBX set-up or the like. I just want my computer to identify an incomming number and either ignore it (and therefore let the other phones in the house ring), or send it to voicemail (VM). VM would be provided by the computer.

Is the Grandstream device not capable of this? Due to your recommendation I will consider spending the extra money on a card, but I would like to understand the situiation before I do.

LVMarc:

Thank you for responding as well. I am still not sure what that Combine-A-Line thing does or if I need addiational hardware with it. (Such as a modem?) Every reference I could find it to was a copy and paste of the sames sales pitch, which is somewhat off-puting.

You're welocme!

Just, because you already want to deploy a full fledged machine for handling voicemail only, whereas IMHO, almost all the features that Gradstream ATA has provides are already available on a linux machine, except for the FXO and FXS ports.

I have not used Grandstream ATA, but I think, it definitely could be an embedded Linux gadget somehow; it might also be using Asterisk for handling VoIP and, or PSTN calls. BTW, better you read it's manuals and, or wait for other people's responses and, or recommendations who have used it.

I'm not sure that the said and, or any other ATA is capable of sending requisite CLI information to a computer, that's why I suggested you to buy a card having a minimum of 1 FXO and 1 FXS ports for your machine.

I suggest you to try miniSipServer. It is a SIP server for windows and very easy to use.

MSS can support powerful dialing plan, so I believe it is very easy to check call with caller ID for MSS.

But MSS cannot support VoiceMail in current versin.

B.R. Hongtian

I suggest you to try miniSipServer. It is a SIP server for windows and very easy to use.

MSS can support powerful dialing plan, so I believe it is very easy to check call with caller ID for MSS.

But MSS cannot support VoiceMail in current versin.

B.R. Hongtian

Balwinder:

Thanks for the follow-up reply. I am still unclear on what a good course of action would be as my reasons for looking at the Grandstream device were the exact same as your reasons for me not to use it. I was hoping I could ask some clarifiying questions of you and everyone else reading this.

As I have a machine I intend to run as a Linux based phone system thing I figured I just need something that will allow the computer to control a phone line. A voice modem without the modem part. :-) I know that a fancy PCI card with a FXO and a FXS port could do what I need, but it is more expensive than the Grandtsream device and I do not require an impressive production system in a critical environment. The Grandstream device may do less, but it should be enough and is cheaper. Quality can come latter when the situiation is not as experimental as it is now.

If the Grandstream device cannot allow my computer to accept and/or place phone calls over the PSTN, what are devices like it used for?

To further my VoIP education could you please tell me what the CLI in the "requisite CLI information" comment you made stands for and a little bit about the concept?

(Perhaps one of my problems is that I do not understand the limits of an Analogue Telephone Adaptor.)

I appreciate your help and certainly look forward to learning more about the wacky world of VoIP from you and everyone else willing to teach me.

B.R.:

Thanks for replying to my post as well. I already have a number of software options to look at and now I have one more, although a linux one is preferred. Still, I am going out to look up info on MiniSIPerver right now. My immeadiate problem is learning what hardware will be required for me to do what I intend.

Did a little more reading...

CLI is the same as ANI? If the Grandstream device can't read that, no caller ID for me. I understand a little more now if this is true.

That's what I do. (Asterisk, Sipura-3000, Grandstream bt-100 phones.)

That looks to be much the same as the Sipura 3k I have. It should work.

Note, learning the weird Asterisk regexp language will remind you of learning Sendmail's "cf" language. Programming in it will make your head hurt. Expect it not to make any sense for a week or two and then it will click. The other sip servers you mentioned might be better. I have no experience with them so can't say for sure.

The one thing that you probably won't be able to do is talk over he POTS line using this device. 2-wire POTS lines have much too much echo. (Without VOIP's added delay your ear will take out echo all on it's own, so you aren't bothered by it. Once you add a hundred milliseconds or two VOIP delay your ear won't filter it any more and the result is hearing yourself talk just delayed enough to make it impossible to continue.) This is why I ended up using my Sipura 3k as an IVR-based answering machine only.

-wolfgang

No. There are subtle differences between ANI and CLI. CLI is the consumer caller ID. That is the only choice you have for a POTS line.

ANI is the telephone system's internal billing or trunk line number associated with the outgoing line. For offices with trunk lines to the central office the ANI will usually be some trunk line instead of the calling party's desk phone's number. In general it is less useful for calling the person back on. The one advantage to ANI is that is isn't subject to the caller-id masking that allows phone spammers to hide their phone number behind a PRIVATE flag.

-wolfgang

Don't waste your time on Asterisk. It's too difficult for the average user. Check out

TekIVR

is an economical SIP (Based on RFC 3261) Interactive Voice System (IVR) for Windows. TekIVR is tested on Microsoft Windows XP, Vista, Windows 7/8 and Windows 2003/2008/2012 server.

Best regards,

Yasin KAPLAN