Implementation Questions, advice

I work in a small law firm and we are going to be moving to larger offices soon. Part of this move will require a new phone system. I would like to push an Asterisk solution so we can take advantage of a greater range of features than would otherwise be available in a system in our price range. Initially, we would be using IP phones internally and all inbound/outbound calls would be through POTS lines via FXO ports. Ultimately, we’d like to start moving to a completely VoIP implementation.

I’ve perused the forums, been through a good chunk of the VoIP wiki, and I have one question for which I can’t find a solid answer. My other questions are really requests for advice, based on your experiences.

Is there a way to identify inbound calls by the number on which they come in so that the agent answering the call can greet accordingly? For example, only one receptionist exists who will personally answer all incoming calls to screen them, forward them, take messages, etc. accordingly. Bob’s number is 1111, and he wants all of his calls answered “Bob’s Office.” Barbara’s number is 2222, and she wants all of her calls answered differently. Each number would initially be coming in on its own FXO port. Can the number called in on be identified in some way to the agent, either through use of different lights on the phone, distinctive ring, or a text display?

My next couple of questions are hardware related.

Is there a significant performance or reliability difference between a PCI FXO interface, like the 4 port Digium cards, versus a standalone FXO gateway, like those offered by AudioCodes or others?

What minimum PC hardware should we plan on using to implement 8 lines and 20 phones to ensure good voice quality at all times?

I’ve seen some fax to pdf solutions for Asterisk. Is it possible to make a pdf of an incoming fax while also shunting the call to an FXS port with a fax machine? The purpose for this would be to keep a backup copy of all incoming faxes.

Finally, telephones…can they be mixed and matched to suit individual users’ preferences? Or is it best to stick with one brand/model across the board?

Thanks for your suggestions!

I can answer some of your questions.

When you get an inbound call, there’s a global variable that holds the number of the party that called you. It is called CALLERIDNUM. We use Cisco IP phones and the caller ID shows up on the LCD display.

FXO or PRI cards definitely take up resources on your asterisk server. The interrupts to service these devices really eat up CPU. However, with a decent CPU, 8 lines shouldn’t really hurt you.

Eight lines and 20 phones won’t take too much of a server, but servers are cheap these days, so I’d go for a 2 Ghz machine, which should do the trick.

I can’t help you on the fax questions but I will tell you that faxing using VoIP is tricky and my personal opinion is to avoid it if at all possible.

You can mix and match any phones you wish, but from a practicality standpoint, it is best to standardize on one phone. You’ll become very familiar with it and can easily swap them around the office if necessary.

looks like you got the big question out of the way

i am working with a local law firm on a 25 phone installation

we are working on some custom code tailored to this environment

  1. utilizing the dictation function and some equpiment from

  2. tagging calls as billable while in progress and reporting those to someone responsible for billing them

pm or email me if interested in discussing

gonna toss in my two cents-

First- congrats on searching and reading the Wiki. Many people don’t figure this out :smiley:

  1. sure you can identify by incoming line a few ways. Either way it is definately possible. Caller ID is held in a variable which can be manipulated however you want, ie-

exten => 1234,1,Set(CALLERID(name)=callforbob-${CALLERIDNAME})

would replace a caller id of John Smith with “callforbob-John Smith”. This can be applied to your dialplan in a number of ways to produce your result. It won’t be hard. The new caller id name is then passed on to whatever it’s sent to, ie the phone.

Faxing- I don’t think you can just monitor a fax session and create a printout from it. I could be wrong. One possible way to do this would be to have * recieve the fax, store it somewhere, then start up another call to transmit it to the fax machine. Or just buy a $150 network laser printer dedicated to faxes which * prints its faxes to. Asterisk could also email the fax to your receptionist. There are many options.

Gateways- it depends on how many lines you have. I wouldn’t put more than 24 analog ports in a server (MAYBE 48 with two TDM2400’s). 8 lines will be no sweat get Digium TDM400 or Sangoma A200 cards. You might consider the sangoma as their ‘remora’ design allows several cards to use only one PCI slot, they link together with a backplane. You can expand up to 20 ports on one pci slot (5 slot backplane). Unless you have a large number of ports I generally say go with the card as it’s one less thing to configure (no SIP link to worry about).

As for telephones there is no technical reason why you can’t mix and match. You can choose ‘dumb’ analog phones connected to FXS ports (cards or on ATAs), ADSI phones on zaptel cards, IP phones, wireless IP phones, softphones (use computer headset), PDAs, you name it asterisk can probably deal with it.
That said, I would recommend IP phones wherever possible, and I’d suggest sticking with at most 2-3 models. I suggest IP phones because they give you quick access to features such as hold, transfer, conference, etc; analog phones require hookflashes and feature codes. You must of course make sure your network is up to the task, but IP phones can be very reliable and work great. They are however more $$/user than plain analog phones. But you’re a law office so you can afford it :smiley: and trust me its worth it.
I suggest standardize on 2-3 models at most so you don’t confuse people. Most non-geek people want to learn something like this once (if that) and never again. Even a different IP phone layout can create subtle annoyances.

As for the phones you DO get- I recommend SNOM and AAstra (manufacturers). Both make great hardware, well built, good business quality handsets; both make models that support 802.3af PoE (so you don’t need power bricks everywhere), and more importantly both support text config files that are easy to deal with. SNOM 360 supports a sidecar with extra buttons for your receptionist.
Whatever you buy get 1-2 to play with and make sure you like them. is a decent reseller I’ve never had a problem w/ them.

For a powered switch check out the Netgear FS726 (if i remember that right), it’s a 24 port managed switch with 16 PoE ports and its only like $300.

What you should do to set up-
First configure your dhcp server to provide the TFTP server address (this would be your * box when you enable tftp, it’s not part of *). This will save you mucho time when you go live.
Set up config files. Both phones use one ‘master’ file and one specific file. IE with AAstra, there is aastra.cfg and (phonesmacaddy).cfg. The phone first grabs aastra.cfg, then the specific file… any settings in the specific file override the master one. So for example you would define the SIP server and maybe feature key paramaters in the master file, leaving the account, password and screen idle text to the specific one (make sense?)

This way when you set up the phone- you just plug it in and that’s it. The phone will get TFTP from the dhcp server, download a firmware update if you provide one, then get the config files and configure itself. Total time including a few reboots- 2 mins tops. No assistance required. Just plug it in and walk away.

Hope that helps!

Do you know if the 7914 busy lamp field is supported with asterisk? Have tyou used the SNOM phone with the BLF?

not sure about 7914 (i’d assume yes) but i know that snom and aastra both support BLF (I have used both).

I’d be cautious about Cisco phones as they can be difficult to configure and you have to keep paying for the support contract…

i’ve used the Snom 360 with sidecar and love them. the only thing you may object to is the power requirement … i think you need to use the PSU … which removes the PoE capabilities. i may be wrong, and it only needs the PSU for > 1 sidecars.

First of all, thanks for all of the advice and suggestions. That’s one of the things that has impressed me about the Asterisk community so far–no newbie sniping. :wink: And very helpful, knowledgeable people.

Actually, it was the Wiki that really prompted me to seriously consider an Asterisk solution. :smiley: The more I read, the more enticed I was. Before I came to work here, I worked for a larger company where we put in a $100K Intertel system that had a lot (but not all!) of the features that Asterisk seems to have, that I didn’t think I could get in a system within our budget.

It’s a shame to hear Cisco phones are such a pain. They have a great look and a lot of appeal. I’m glad to hear the SNOM and sidebars work with Asterisk–I’ve looked at that phone already and had put it on my list to check out.

Reviewing some of the forum posts, I’m realizing I need to consider QoS issues. We currently have a small 10/100 switched network, five PCs and a couple of network printers. We will be adding several PCs, and probably a couple of printers. Should I check out switches that support QoS? I’d hate for a call to drop every time someone decided to print. :wink:

Here’s a question antithetical to open source: what type of commercial support is available for an Asterisk system if I leave the company? I’m hesitant to put in a system that no one can manage once I’m gone. I think remote management would probably be suitable for most issues that might arise in the future, if there are any firms that offer it. I know Digium has a commercial version of Asterisk, but I’m not sure I want to go with that from the start, and I suspect they wouldn’t be willing to support any custom tweaks I might put in.

Another loaded question: are there any reputable VoIP telephone service providers we could switch to in order to have a completely IP solution? We would need to port existing numbers, and the carrier would need to be reliable. When we first installed the Intertel system at my former employer, CLEC startups were all over the place, but kept dropping on us like flies, or couldn’t deliver what they promised. Most of the VoIP providers I see mentioned on the Wiki look geared towards individuals or small businesses who may be more tolerant if their phone company vanishes one afternoon. :smiley:

Some people have more trouble with Ciscos than others. I have heard of some people spending 8+hours trying to get them to simply firmware upgrade, and I have heard of others that have never had any problems. Personally I avoid them on principle- i dislike having to jump thru hoops and pay extra for firmware and documentation.

QoS- if you use VoIP providers, you will NEED QoS controls on your WAN link(s). This is not optional. Also make sure you have the BW to cover it :smile:
For your internal network- it’s often a good idea to partition the phones onto a separate VLAN. This can be given elevated switch priority. If you only have 1x ethernet per office and are wiring the computers thru the phones, the phones can usually tag their own traffic.
Check out the switch i mentioned- netgear fs726tp. I’ve used it a few times with good success and its cheap. 24 ports, 16 of them PoE.

For commercial support- there are a handful of companies that support * installs (many are on the Wiki). There are also a large number of independent consultants that will provide support for a fee, many lurk around on this board, the Wiki and the Asterisk chat room ( #asterisk). There are a few lists on the Wiki of paid support providers.
If you buy ABE (asterisk business edition) then it comes with support from Digium. As I understand it they will support you decently well, but keep in mind ABE is a binary (no source code) so you can’t compile anything in even if you want to. You can also pay digium for support as I recall for general issues…

As for VoIP providers it depends on how much you use. If you use like 100k minutes/mo then the big guys (CLECs, XO, Level3, etc) will talk to you and you can probably get a good deal. For less than that there are a ton of companies, but like you say many are small and not geared entirely for business.

One thing to consider when hiring an ITSP (voip provider) is what kind of ITSP you will get. There are two major choices- ‘lines’ or wholesale.

A provider like Viatalk or Broadvoice sells you lines. Each ‘line’ is a DID (phone number) with usually unlimited outgoing minutes. You usually get two channels (two concurrant calls, so 3way calling works) per ‘line’ and your outbound caller id is locked to your DID number (although it can be blocked ala *67). Note that with two channels on BYOD- you can have two useful calls at a time, IE if you are on the phone a 2nd person can make a call. This isn’t a 3way call but it works.

A provider like voicepulse connect ( is a wholesale provider. They sell you DIDs and minutes separately. You pay a lowish fee for the DID and that gets you unlimited incoming calls. You then PAYG for outgoing minutes, usually 1-2c/min. You can also set your own caller ID for outgoing calls, although there is often a sanity check done on this (so you can’t call from 911). The channel limit is higher or nonexistant. VPConnect for example gives you 4 inbound 4 outbound (8 total) per account and you can buy more if you need them.

Thus you should consider your needs carefully- if you use less than 4000-5000 outbound minutes per month, or you use alot of concurrant calls, or most of your calls are inbound, wholesale is definately a better option.

I beg to differ on Cisco phones. We have about 300 of them in the field and have relatively few problems with them. We also use Polycom and while they are less expensive, we do see more failures with them. I don’t think you can go wrong with a Cisco. If it is taking 8 hours to get it configured, then you are doing something wrong.