[MillsapsPE]

I am researching the idea of setting up an * box. But surprise, I have several questions.

I currently have 1 POTS line and 1 Vonage line at my office with 4 extensions in use. I want to add a 3rd line in near future. My current phone system is a 2 line system (Panasonic KX-TG2000B). I use the auto-attendant feature to direct the caller to 1 of 4 extensions (i.e. dial 1# to reach John Doe). Since I'm going have redo my phone system anyway, I thought about replacing it w/ a * box and IP or analog phones.

Obviously, I need a (dedicated) computer to install Asterisk and 4 IP phones.
-Do I need a Digium Wildcard TDM400P? I assume I need at least 2 FXO ports (since I have 2 existing lines)(TDM02B)? or, would a Sipura box (2100 or 3000) be better? I think each 3000 provides a FXO and FXS port.
-The future third incoming line will likely be from Voicepulse Connect, Broadvoice BYOD, or similar. Therefore, I don't need an additional FXO port, correct?
-Which protocol should I use for the third line, SIP or IAX? Advantages/disadvantages of each?
-Do I need at least 1 (or 2) FXS ports as backups?
-What do you have to watch out for if purchasing IP phones? I'm guessing they have to at least support the correct protocol. Do different brands mix together well?
-What exactly are "line appearances"? How many do I need for my setup?

I know almost nothing of Linux, though I was able to install RH on my computer (dual boot). That was about the extent of my involvement in Linux. I am planning on using the Asterisk@Home iso to get setup, or should I use a different setup?

This is for a business application, not residential, so it needs to be pretty solid (in terms of call quality and reliability).

I'm still reading information here and at voip-info.org trying to educate myself. Thanks for any help in getting my understanding a jumpstart.

[Blue]

you might want to post it to * user group directly.. I am also interested in hearing replies to your question

[MillsapsPE]

Originally Posted by Blue

you might want to post it to * user group directly.. I am also interested in hearing replies to your question

I thought this was the * user group? Can you provide a little more information as to which web site or usenet group you are referring to? Thanks.

[Blue]

http://www.asterisk.org/index.php?menu=support and look for user mailing lists

[daryll]

Well let's see.

First, you've got to have an Asterisk box. It should be dedicated to the function if you want good voice quality.

Seond, you need a way to connect two POTS lines. That means you need FXO interfaces. The Sipura 1000 and 2000 line provide FXS interfaces, not FXO, so they won't help you. You could use 2 3000's or a TDM card. Analog is a bit ugly no matter how you do it, but I suspect the TDM card is the right answer. It can be configured with 4 ports that are either FXO or FXS, so it leaves you room to grow.

Third, you may want to think about an ISDN solution for two lines. It'll cost you a bit more, but staying digital the whole way is a big win. You're probably not getting enough lines to make a partial PRI a reasonably priced choice.

If you are getting VOIP service you don't need any additional hardware. BroadVoice (or whomeever) will reach you via the Internet. Realize that you do need about 80k (upload and download) available for each voice channel you want to have active. So a 256k upload speed supports 3 voice channels if nothing else is using the bandwidth. Also consider getting a router that support QoS. That can prioritize voice traffic for particular machines.

Backup choices? Today I wouldn't run my entire business on a VOIP connection. Although I like it and use it all the time, it's nice to have a fallback POTS line. It's also great for 911. So keeping at least one of your POTS line (even after setting up VOIP) is probably a good idea. Do you need backup FXS ports? I don't really think so. If you have a backup POTS line you can always plug a phone in to it. Since FXS boxes (Sipuras for example) are just as likely to break as anything else I don't think you're buying much security that way. You can always reconfigure your SIP phones to talk straight to your VOIP provider if your asterisk box goes down for some reason. Of course, you could make your asterisk box redundant as well.

SIP is pretty standardized. There are features that are different among different units that you should read up on, but they that should be pretty clear. You'll get a lot of opinions on different choices.

I responded to your other post about call appearances on the 841. Basically a call appearance is a phone call. The base price for the Sipura supports 2 call appearances which means you can do 3 way calling, call hold, etc. That's probably the way you want to use it in your office.

As was pointed out there is an asterisk-users mailing list. It's really high volume, but if you read through the archives there are useful gems. Also check out http://www.voip-info.org which has a lot of Asterisk stuff in it. Finally, http://www.asteriskdocs.org has a PDF book on Asterisk. Asterisk really takes a technical person to run, more so that some of the other PBX in a box type setups.

[mberlant]

This is the right place. Let's get started.

Originally Posted by MillsapsPE

-Do I need a Digium Wildcard TDM400P? I assume I need at least 2 FXO ports (since I have 2 existing lines)(TDM02B)? or, would a Sipura box (2100 or 3000) be better? I think each 3000 provides a FXO and FXS port.

It's up to you and it's mostly a financial question. Each of the two lines you have now requires an FXO port from your Asterisk. This can be one of the lines on the Wildcard 400, a Wildcard 100's single port or the FXO port of an SPA3000. Your Vonage service, since they will not release the credentials to you, will have to enter your Asterisk box via FXO port, but that can be any FXO port. Your PSTN Line, though, should probably come in to the Asterisk via SPA-3000, so that in the case of a power failure (or internet failure) you will have failover service to the telephone plugged into that SPA's Line 1 jack.

Originally Posted by MillsapsPE

-The future third incoming line will likely be from Voicepulse Connect, Broadvoice BYOD, or similar. Therefore, I don't need an additional FXO port, correct?

As long as you use a service that is willing to give you your SIP credentials, like the ones you mentioned, they will come directly in to the Asterisk via ethernet.

Originally Posted by MillsapsPE

-Which protocol should I use for the third line, SIP or IAX? Advantages/disadvantages of each?

Many providers do not support one or the other protocol, so that may be moot if you are choosing on price alone. That being said, IAX is better suited to multiple simultaneous calls than SIP. Also, IAX requires less overhead to support additional simultaneous calls than does SIP. For SIP, you should plan on having 80kbps in each direction for each conversation. IAX requires the same 80kbps for the first call on a service provider, but only 65kbps for each simultaneous additional call using the same provider (every little bit helps). Also, in order for Asterisk to work with SIP it needs either to be in your DMZ or to have its required ports forwarded through your router, since Asterisk has neither STUN nor Outbound Proxy capability.

Originally Posted by MillsapsPE

-Do I need at least 1 (or 2) FXS ports as backups?

You need 1 FXS port for each ordinary phone you wish to connect as an extension.

Originally Posted by MillsapsPE

-What do you have to watch out for if purchasing IP phones? I'm guessing they have to at least support the correct protocol. Do different brands mix together well?

If you are going to put in Asterisk as your PABX you only need to worry about each device being Asterisk compatible.

Originally Posted by MillsapsPE

-What exactly are "line appearances"? How many do I need for my setup?

A line appearance is an instance of a single extension number showing up on a single line telephone or on a button of a multiline telephone. For example, two buttons of a multiline phone programmed with the same extension are two appearances of that extension. A boss' extension number showing up on a secretary's phone is another appearance. And so on. I don't know the limitations of Asterisk in this regard - you will need to get that advice from someone else.

Something to keep in mind, although it may be the identical case to your Panasonic PABX, is that the Asterisk is unable to send a hookswitch flash to the PSTN Line. This means that you cannot have Call Waiting or 3-Way Calling from the phone company (and you will need to disable it within Vonage).

[MillsapsPE]

Originally Posted by mberlant

This is the right place. Let's get started.

A line appearance is an instance of a single extension number showing up on a single line telephone or on a button of a multiline telephone. For example, two buttons of a multiline phone programmed with the same extension are two appearances of that extension. A boss' extension number showing up on a secretary's phone is another appearance. And so on. I don't know the limitations of Asterisk in this regard - you will need to get that advice from someone else.

Something to keep in mind, although it may be the identical case to your Panasonic PABX, is that the Asterisk is unable to send a hookswitch flash to the PSTN Line. This means that you cannot have Call Waiting or 3-Way Calling from the phone company (and you will need to disable it within Vonage).

Thanks everyone for your responses. I'm quite familiar with Asterisk wiki and have printed the manual from asteriskdocs.org. I'm still reading through the various wiki pages and will start reading the manual tomorrow. I did get *@Home installed this weekend on a AMDK2 350Mhz, 288MB RAM machine. I can access the AMP page from another computer. I haven't played with any of the config files yet.

I guess my real question is (re: IP phones), if I have 4 employees w/ 4 phones with 6 outgoing lines (2 FXO + 4 through VoicePulse Connect), I can still use single line (single line appearance) phones because * takes care of what outgoing line to use (with the correct dial plan). Basically, the user will be able to pick up the phone and make an outgoing call and be able to make conference calls. The extra line appearances are more for convenience?

Another question (related?). If Person A is on a call with Person B and someone else (Person C) calls in and wants to reach Person A's extension, how will Person A know another call is coming in, or will they, or would it go straight to voicemail?

[PhoneBoy]

Depends on the SIP device/phone being used. On my SPA-841, if I have another line available, it will simply ring on my second line. On an analog handset connected to a SPA-2000, I'd get the familiar Call Waiting tone.

[MillsapsPE]

Originally Posted by Blue

http://www.asterisk.org/index.php?menu=support and look for user mailing lists

I signed up for Asterisk-User list and have read/skimmed through over 600 messages. Unfortunately, it doesn't appear most of the guys/gals are to easy on the new kids. According to one or two posters, you better not dare ask a question that has ever been ansered before. I believe in doing your own research, but some of them seem to take it to the extreme.

[daryll]

Originally Posted by MillsapsPE

I signed up for Asterisk-User list and have read/skimmed through over 600 messages. Unfortunately, it doesn't appear most of the guys/gals are to easy on the new kids. According to one or two posters, you better not dare ask a question that has ever been ansered before. I believe in doing your own research, but some of them seem to take it to the extreme.

Yeah, it's sort of a rough crowd. A list with that much traffic is just really hard. I skim it just to keep up with what's going on, but rarely post or reply.

There's a few of us who really know Asterisk here and with the lower volume it's easier to write good responses. So go ahead and post questions here.

- |Daryll