Analog lines and Freepbx

General Help
1

Hi everyone,

I need your recommendations.

My current office in Malaysia has 5 analog lines (4 for phone calls and 1 for fax). 1 out of the 4 phone numbers is the main hunting number. We used a super old panasonic pbx and has no plans to use it anymore. Our current workforce is around 8 - 10 people. 4 at office and others mostly mobile workers.

I plan to propose to my boss to setup asterisk + freepbx and he likes the idea of VOIP. I also believe FreePBX comes with Fax too, which makes things easier. However he is okay with the change, just the numbers can’t be changed.

3 questions.

  1. I have an existing server (quite new) which I can use (hopefully this spec is good enough?):-
  • 1 x Intel Xeon Quad-Core E3-1230v2 3.30Ghz processor
  • 5.0GTI, 6MB Intel Smart Cache, DMI, Hyper-Threading, Intel
    Turbo Boost 2.0, Intel Virtualization (VT-x) & (VT-d) Technology - Intel C202 PCH chipset
  • 4GB ECC DDR3 1333Mhz memory, max 32GB
  • 1 x 1.0TB 3.5" Enterprise SATA 3Gb/s, 64MB, 7.2K rpm
  • Intel C202 SATA2 RAID supporting level 0, 1, 5, 10
  • 1 x 1GbE Intel Gigabit Ethernet LAN port via
    PCIe x4 NIC

  1. But how about the 5 analog lines (4 phone numbers & fax line)? I assume I need to plug into some PCI card which serves both phone and fax purpose? or some ATA box? Any recommendations which works with Asterisk & FreePBX but reasonable cost?

  2. For the fax, I assume FreePBX is using HylaFax. Is the fax workable and good enough for small workforce? We receive around less than 100 faxes a month. We like the idea of HylaFax open source aka AvantFax (email to fax, fax to email, client to fax)… as you can fax anywhere, not necessary at the fax machine. Same to reading your faxes if someone fax to our office’s fax number. Do I need to buy another new server to allow communications between asterisk + freepbx and fax? or will it be the same box?

Any help from this community? Thanks.

2

You have 20 times more server power than you need. I wouldn’t waste that server on the PBX, just get a small Atom box.

The folks that write Asterisk make the cards www.digium.com as does a company named Sangoma, those are excellent cards also.

FreePBX has it’s own fax engine, SpanDSP and does not use Hylafax or Avantfax. FreePBX does not have outbound email to fax capability however one of the add on commercial modules, Fax Pro does add PDF to Fax capability.

What kind of phones are you thinking of using?

3

Thanks for replying back.

As for your answers:-

a. You have 20 times more server power than you need. I wouldn’t waste that server on the PBX, just get a small Atom box.

So I just need to buy an intel atom home computer will do? any recommended specific specs which I can benchmark with?

b. The folks that write Asterisk make the cards www.digium.com as does a company named Sangoma, those are excellent cards also.

For digium, are you referring to either this
http://www.digium.com/en/products/telephony-cards/analog/4-port
or
http://www.digium.com/en/products/telephony-cards/analog/8-port

and for sangoma, are you referring to this?
http://www.sangoma.com/products/a200-2-24-port-expandable-analog-voice-board/
or
http://www.sangoma.com/products/a400-2-24-port-expandable-analog-voice-board/

am i correct?

So installing the above cards on Centos 6x & Asterisk will be straightforward too?

c. FreePBX has it’s own fax engine, SpanDSP

So even without the fax pro, is SpanDSP stable and good enough? I assume the incoming fax can be pushed into gmail right?

d. What kind of phones are you thinking of using?

2 possibilities:-

softphone - bria 3, or open source soft phone
voip phone - yealink ip phone, or digium phone

so what do you recommend best for freepbx?

e. Last question, which version of freepbx should I consider? Which is stable for office use and not for testing?

Any help? Thanks.

4

a. You have 20 times more server power than you need. I wouldn’t waste that server on the PBX, just get a small Atom box.

I would go with a server grade such as the Supermicro, simply to get a better power supply. Any decent 1U appliance type server. The less moving parts the better.

So I just need to buy an intel atom home computer will do? any recommended specific specs which I can benchmark with?

b. The folks that write Asterisk make the cards www.digium.com as does a company named Sangoma, those are excellent cards also.

For digium, are you referring to either this
http://www.digium.com/en/products/telephony-cards/analog/4-port
or
http://www.digium.com/en/products/telephony-cards/analog/8-port

and for sangoma, are you referring to this?
http://www.sangoma.com/products/a200-2-24-port-expandable-analog-voice-board/
or
http://www.sangoma.com/products/a400-2-24-port-expandable-analog-voice-board/

am i correct?

Exactly correct

So installing the above cards on Centos 6x & Asterisk will be straightforward too?

Yes, very simply, however if your requirements are not special why would you not use our distro? It is put together by the team developers and fully vetted. It is also supported commercially if you ever need assistance quickly.

c. FreePBX has it’s own fax engine, SpanDSP

So even without the fax pro, is SpanDSP stable and good enough? I assume the incoming fax can be pushed into gmail right?

Yes, out of the box FAX to email, works fine.

d. What kind of phones are you thinking of using?

2 possibilities:-

softphone - bria 3, or open source soft phone
voip phone - yealink ip phone, or digium phone

so what do you recommend best for freepbx?

Phones are a personal preference, as long as it is supported and has a body of users you in good shape. Certainly the brands you mentioned are excellent and the Digium phones have deep feature integration.

e. Last question, which version of freepbx should I consider? Which is stable for office use and not for testing?

FreePBX and Asterisk are vetted with hundreds of thousands of installations world wide. The stable is stable and the beta is cutting edge. Once beta gets to release candidate stage I start rolling it out to friendly users.

Any help? Thanks.

I hope I have, your questions are thoughtful, concise and easy to answer.

5

Hi Scott,

Thanks so much for replying back. You rocks :slight_smile:

Just few more questions.

a. for the server, am i looking something like this specs? http://www.supermicro.com.tw/products/system/1U/5017/SYS-5017A-EP.cfm

For the above box, I assume I can plugged in either the digium or sangamo card (8 ports, example).

but seriously, this box can cater concurrent 4 analog calls, at the same time outgoing calls via SIP providers, receive faxing, and able to support 20 people? No lags or slow down?

if it is possible, i am surprised atom processor can do this though. But 4GB DDR3 RAM really sufficient? I used an intel atom netbook before this and it is really slow :slight_smile:

b. Assume I don’t plan to get the fax pro module, but to install hylafax+ … is it recommended to install hylafax+ on the same box with Freepbx distro or separate box? If it is separate box, should i have ethernet cross cables between servers for the communication to happen? I believe Freepbx server will have the Analog Card attached to it, somehow i need to pass the fax calls across. Any idea how to get this to work?

I really like the idea of email-to-fax from AvantFax. It is an interesting idea though :slight_smile: Unless fax pro can do email-to-fax?

c. Assume I get the fax pro module (assume my boss doesn’t care much about the email-to-fax thing)… is there any developer API which I can call from my 3rd party application to talk to fax pro module? Example, in my open source portal, i need to grab the latest 10 incoming faxes from freepbx fax pro module. Something like that? Can this be done?

d. Since the beta distro is not on release candidate yet, I will go with stable. Should I go with the Stable-2.210.62-6 32 bit or 64 bits? Does it make any difference if I buy the above server?

e. Since all incoming calls come from the analog lines, I have no choise to use them. But for outgoing calls, could I use other than the analog line.

example,

if detect local number -> some local SIP provider which offer cheaper calls rates
if detect mobile number 012,019 -> some gsm gateway to call out
if detect some code -> dials out to skype account (i saw this feature in pbx in flash)
if detect australia number -> some SIP to australia that offer cheapest calls
if detect X country number -> some SIP to the X country
else if all numbers failed (assume SIP failed to connect or something like that), I will use back the analog lines for outbound calls.

would that work? and how can i configure that in freepbx?

f. Related to point e above, is it possible to not to show any numbers when I call my customer? It will confuse them that we have so many numbers calling our customers. The customer only needs to know our main hunting number for incoming will do. Any idea to make this work? Or do I have to talk to each SIP provider to get them to show Private or no number instead of the actual number, something like that?

Any help? Thanks :slight_smile:

6

Just few more questions.

a. for the server, am i looking something like this specs? Supermicro | Products | SuperServers | 1U | 5017A-EP

God yes, that’s the new 64 bit Atom. Linux/Asterisk are very efficient. I run two PRI’s and 120 residence rooms off of the earlier Atom processors.

For the above box, I assume I can plugged in either the digium or sangamo card (8 ports, example).

The Sangoma for sure, check the length on the 8 port Digium. The Openvox 8 Port card (see link below) is a better physical fit. It works find with our distro.

http://www.openvox.cn/en/products/fxofxs-cards/a810e.html?page=shop.product_details&flypage=flypage.tpl&product_id=86&category_id=1

but seriously, this box can cater concurrent 4 analog calls, at the same time outgoing calls via SIP providers, receive faxing, and able to support 20 people? No lags or slow down?

Yes, yes yes. How many more times are you going to ask? More than enough power. Asterisk was designed when a PIII 1Ghz was a screaming machine. That quad core Xeon of yours would do over 1000 concurrent calls. Go search Asterisk dimensioning for more background.

if it is possible, i am surprised atom processor can do this though. But 4GB DDR3 RAM really sufficient? I used an intel atom netbook before this and it is really slow :slight_smile:

Oh my God, did you really ask again?

b. Assume I don’t plan to get the fax pro module, but to install hylafax+ … is it recommended to install hylafax+ on the same box with Freepbx distro or separate box? If it is separate box, should i have ethernet cross cables between servers for the communication to happen? I believe Freepbx server will have the Analog Card attached to it, somehow i need to pass the fax calls across. Any idea how to get this to work?

I know nothing about Avantfax or Hylafax. The key is that they work with Linux and Asterisk so of course you can run them on the same box. Configuring them will be up to you.

I really like the idea of email-to-fax from AvantFax. It is an interesting idea though :slight_smile: Unless fax pro can do email-to-fax?

No but you may want to ask Schmooze (the author) if that feature is on the road map.

c. Assume I get the fax pro module (assume my boss doesn’t care much about the email-to-fax thing)… is there any developer API which I can call from my 3rd party application to talk to fax pro module? Example, in my open source portal, i need to grab the latest 10 incoming faxes from freepbx fax pro module. Something like that? Can this be done?

No, not at the present time. I am sure if you shoot a developer a ticket they would give you hooks. Make sure you are fluent in PHP or don’t bother asking.

d. Since the beta distro is not on release candidate yet, I will go with stable. Should I go with the Stable-2.210.62-6 32 bit or 64 bits? Does it make any difference if I buy the above server?

If you get the Supermicro go 64 bit and run 8G of RAM. RAM is cheap. Like any modern OS, the more RAM you toss at it the better it runs.

e. Since all incoming calls come from the analog lines, I have no choise to use them. But for outgoing calls, could I use other than the analog line.

Not only can you we suggest it. SIP trunks offer lower cost and more features. If you are in KP you will find many good local providers.

example,

if detect local number → some local SIP provider which offer cheaper calls rates
if detect mobile number 012,019 → some gsm gateway to call out
if detect some code → dials out to skype account (i saw this feature in pbx in flash)
if detect australia number → some SIP to australia that offer cheapest calls
if detect X country number → some SIP to the X country
else if all numbers failed (assume SIP failed to connect or something like that), I will use back the analog lines for outbound calls.

would that work? and how can i configure that in freepbx?

This is just a matter of routing. I suggest you take a look at the documentation and load the distro up on a spare machine or a VM so you can see what it does.

f. Related to point e above, is it possible to not to show any numbers when I call my customer? It will confuse them that we have so many numbers calling our customers. The customer only needs to know our main hunting number for incoming will do. Any idea to make this work? Or do I have to talk to each SIP provider to get them to show Private or no number instead of the actual number, something like that?

Most SIP providers allow you to set outbound CID. You can’t do that on analog though you might be able to block on outbound with a prefix code

Any help? Thanks :slight_smile:

7

Thanks Scott :slight_smile: You rocks 100x times :slight_smile:

I will probably get the supermicro reseller in Malaysia and recommend them to look at the sangamo or openvox specs, and see whether it can fit into the 1U specs.

I will also get the atom 64 bit processor as well as 8GB of RAM.

Just 2 more questions.

  1. Most SIP providers allow you to set outbound CID. You can’t do that on analog though you might be able to block on outbound with a prefix code

Assume my analog number is this 03-88819393 (example). If I use other SIP providers and they allow me to set outbound CID, could I use the analog number 03-88819393 as the outbound CID? So to the receiver, it seems like it is coming from my analog number, and they will remember that. Is that possible?

  1. Just added this question.

Since I have total 5 analog lines. 4 for phone numbers and 1 for fax number. Assume I use SpanDSP or FaxPro module, at the the time…

Main Pilot analog line - being used
Other 3 analog line for phone - not being used
Fax Line - being used

If I want to send fax, could the outgoing fax take advantage of the other 3 analog lines which are avaialble to be used? Can this be done? Or I have to specifiy only the 1 fax analog line?

Any help? Thanks.

8

Assume my analog number is this 03-88819393 (example). If I use other SIP providers and they allow me to set outbound CID, could I use the analog number 03-88819393 as the outbound CID? So to the receiver, it seems like it is coming from my analog number, and they will remember that. Is that possible?

Yep, most carriers let you send any CID you want. To the called party it looks like you called from the landline.

Since I have total 5 analog lines. 4 for phone numbers and 1 for fax number. Assume I use SpanDSP or FaxPro module, at the the time…

Main Pilot analog line - being used
Other 3 analog line for phone - not being used
Fax Line - being used

If I want to send fax, could the outgoing fax take advantage of the other 3 analog lines which are avaialble to be used? Can this be done? Or I have to specifiy only the 1 fax analog line?

Yes, you don’t have to designate any line for outbound FAX. In fact if your SIP provider supports t.38 you can send FAX over SIP.

9

Thanks Scott :slight_smile: I will give freepbx distro a try :slight_smile:

10

Hi Scott,

By the way, I need your help on this.

I contacted example OpenVox reseller for these cards

http://www.openvox.cn/en/products/fxofxs-cards/a400e.html?page=shop.product_details&flypage=flypage.tpl&product_id=18&category_id=1 (PCI Express slot)

http://www.openvox.cn/en/products/fxofxs-cards/a400p.html?page=shop.product_details&flypage=flypage.tpl&product_id=19&category_id=1 (PCI Slot)

I assume either specs above will work with the FreePBX stable distro right?

But they asked me how many FXO and FXS modules needed. Example, 2 FXO and 1 FXS.

Hmm… I am not that sure.

I have 4 Analog Lines (3 for phone lines and 1 for fax lines - caters both inbound/outbound calls).

Phone Line 1 - Main pilot number
Phone Line 2 - this number can be interchanged for phone or fax subject to availability
Phone Line 3 - this number can be interchanged for phone or fax subject to availability
Fax Line - Main number

I have no plans to go beyond 4 analog lines.

Any idea what will be the right specs which I can propose to them? Any help?

11

Rather than just give you the answer, I’m going to direct you to simply Google “FXO FXS”. You’ll find all about these terms, and why they’re important to analog telephony.

Bill

12

yeah, you are right :slight_smile: i found my answers here

http://my.digium.com/en/docs/misc/fxs_fxo_desc/
http://forums.digium.com/viewtopic.php?p=153225

basically i need 4 FXO port analog card

But something confuse me.

PSTN FXS -> 4 analog lines -> 4 ports FXO analog card (attached into the server)

basically freepbx distro can receive and create outgoing calls.

So my question is…

Can I setup a VOIP phone e.g. Digium or Yealink phone and point to the Asterisk IP server? Can this work?

This diagram confuses me a little http://my.digium.com/images/graphics/fx_demo.gif and seems like i need to connect phone line from the analog card to physical phone???

Any idea? Thanks.

13

That drawing is showing using an analog phone, so it would need an FXS port. The PBX would switch the call from the FXO to FXS, that’s the X in PBX. Since you want to use IP phones the PBX will switch from FXO to SIP. The job of the PBX is to terminate and route calls.

14

Thanks Scott :slight_smile: you rocks :slight_smile:

I’m getting the 4 FX0 analog card now :slight_smile: cheers

15

is there a reason why you just don’t switch to efax for all your faxing needs?

16

what do you mean by efax in this context?

isn’t that spandsp or faxpro able to achieve electronic fax?

17

Yes they are, I don’t understand why he brought up a third party service that is not even available in Malaysia.

18

hi scott, thanks again.

Need your advice before buying :slight_smile:

i checked with few suppliers and below were their proposal.

Note: I am going with PCI-e and yes the server which I purchase can support PCI-e. I assume PCI-e is a better choice as it is faster than PCI.

Digium 1AEX404EF 4 FXO (with hardware echo cancellations) = RM2019.00

Digium 1AEX404BF 4 FXO (without hardware echo cancellations) = RM1290.00

Sangoma A200 Analog Card 4 FXO (with hardware echo cancellations) = RM2145.00

Sangoma A200 Analog Card 4 FXO (without hardware echo cancellations) = RM1295.00

OpenVox A400E04 4 FXO (no hardware echo cancellations) = RM637.00

Questions:-
a. OpenVox is really cheap here compare to others. Is it as good as Digium and Sangamo?
b. Is hardware echo cancellations matter here especially for Analog lines? Budget is not an issue for me. If is needed, Digium vs Sangamo, which one better?

Any help? Thanks.

19

if cost isn’t an issue, my preference is sangoma, then digium. you’re much better off getting the card with e/c.

20

Openvox delivers a lot of value. Sangoma has a very high quality driver stack. Buying from Digium helps support Asterisk. It’s a tough call.

Not knowing the analog line quality and how much LBO you need in your part of the world it is hard to comment on the need for EC. The cards have prescription levels adjustments that if your line quality doesn’t fluctuate wildly will work fine was the hybrid is properly adjusted (levels set to null out any echo). The echo canceller provides additional protection against echo.