What Harsware i need to connect astrisk with analog phones?


#1

Hello every body,

H’m new to asterisk i have installed asterisk on my linux system and tested it with x-light phones.

Now i want to make PBX system for our company with near 35 employees .

I have analog phones with Landline .

My server is : HPProlaint DL80G6 with OS:Centos 5.6

as i read i need analog interface card to make this working but i don’t have enough information.

Can any one Help me pleaaaaaaaase !!!


#2

You have 3 options:

  1. T1 cards attached to a channel bank. Expensive, since you’d need at least 2 ports of T1 plus at least 2 channel banks to get 35 extensions.

  2. 2 of the Digium 24-port analog cards. Again, expensive. About $3000 USD total.

  3. Analog Terminal Adapters for each of the phones. Cheapest and most expandable way to go, but requires an Ethernet/IP connection at each phone. Some examples of the commonly used ATAs are the Cisco ATA-186 and Grandstream HT-486.


#3

[quote=“jpsharp”]You have 3 options:

  1. T1 cards attached to a channel bank. Expensive, since you’d need at least 2 ports of T1 plus at least 2 channel banks to get 35 extensions.

  2. 2 of the Digium 24-port analog cards. Again, expensive. About $3000 USD total.

  3. Analog Terminal Adapters for each of the phones. Cheapest and most expandable way to go, but requires an Ethernet/IP connection at each phone. Some examples of the commonly used ATAs are the Cisco ATA-186 and Grandstream HT-486.[/quote]

Thank you so much Sir for your reply but i have another question please.

Do ATAs are the Cisco ATA-186 and Grandstream HT-486 require VOIP account ??

I don’t wan’t to use voip i want to take advantage of my Landline telephone service provider

Waiting your reply.

Thank you again


#4

another question please …

if i’m taking about near 10 employee for asmall company what will i need ??


#5

[quote=“Bayan”]Do ATAs are the Cisco ATA-186 and Grandstream HT-486 require VOIP account ??

I don’t wan’t to use voip i want to take advantage of my Landline telephone service provider [/quote]

The SIP connection is terminated on Asterisk; you do not need to pay an external service to “provide accounts”.

You definitely need to read Asterisk: The Future of Telephony, or The Definitive Guide.


#6

[quote=“david55”][quote=“Bayan”]Do ATAs are the Cisco ATA-186 and Grandstream HT-486 require VOIP account ??

I don’t wan’t to use voip i want to take advantage of my Landline telephone service provider [/quote]

The SIP connection is terminated on Asterisk; you do not need to pay an external service to “provide accounts”.

You definitely need to read Asterisk: The Future of Telephony, or The Definitive Guide.[/quote]

OK sir i want my phones to be able to call any number such as my mobile number …

if i have a phone connected to asterisk can i use it to call any mobile or number ??


#7

Not if that is all that is connected. Please read the books. They can be downloaded.


#8

It is better you get a trunk from some voip provider that allows dialing to mobiles and landline with minimal charge. If you use fxo card with asterisk then you can use a landline phone cable to your fxo card and dial out mobile or land line number. If landline phone provider allows calls to mobile. You can achieve this with something like this.

exten => _X.,1,Dial(DAHDI/1/${EXTEN})
exten => _X.,n,Hangup()

Remember, fxo lines are very lousy with answer and hangup detection and you will need to do some tuning and experiments. At the end I agree with david that you must ready asterisk TFOT book.


#9

[quote=“BrennaKessler”]
exten => _X.,n,Hangup()[/quote]

This is not needed with any currently supported version of Asterisk. Asterisk will terminate successful calls if either party hangs up, and will terminate unsuccessful ones when it runs out of dialplan steps.


#10

Thank you all …

ihave another question >>

how many internal extensions (inside the company) can Asterisk have? Does it depend on the server (hardware) card?


#11

For VoIP, depends on the number of simultaneous calls, and the amount of audio processing done within Asterisk.


#12

I really thank you Mr david55 for your help …

i’m new to asterisk .

we attend to connect asterisk to our phone line and analog phones in our company.
as i read we should use analog interface card such as TDM400P.

can you help with more explanation and again if this depend on the server (hardware) card ??


#13

You may consider the MP-118 FXO from audiocodes, about 600$.
will connect up to 8 POTS lines.

I have one and ONCE IT’S CONFIGURED (I mean it’s a pain to configure)
it works really well. No need to reset once in a while, it’s just plain solid.

http://www.voipsupply.com/blog/setting-up-an-audiocodes-mp-114118-fxo-with-asterisk-and-freeswitch

Martin Politick.


#14

The TDM400P is EOL’d; it was replaced by the TDM410. There are numerous reasons why the TDM410 is superior to the TDM400.


#15

To cut down price, you can use soft phones if employees have PC with mic and speaker or headphone at their desk. Least cost option is to use Voip adapters with normal telephone handsets. For example Linksys SPA8000 adapter - which has 8 FXS ports and can connect 8 telephone sets costing around 250$. Moving to voip is smart choice after all :smile:


#16

My scenario looks like:

we have a company with 20 employees, we do not mind having IP or analog phones internally, and we need to have up to 8 simultaneous lines to the outside preferably regular phone lines, what would be the best hardware configuration in terms of server (CPU, RAM, etc.), Digium card, phones, etc.? Is there anything we need to watch for, or to be concerned about as far as Asterisk is concerned with such configuration?


#17

For 8 ports, look at Digium’s 8-port TDM800 (PCI) or AEX800 (PCIe) cards.

That plus 20 phones doesn’t require anything substantial on the hardware side - go find the slowest PC or server in your IT closet and it’ll do.