Hi guys thank you for such a great software, and for such a big helpful community. I’ve just started learning Asterisk and more Free Alternative to PBX, so just landed up here… and decided to work on a small project.
I’ll have 10 users in the network, all will be connected to Asterisk Server.
Using VoIP Phones | ATA Adapters and PC . It will an Intercom project, like a VoIP Intranet project where every user can talk to each other inside the network. But some of them or all of them have or will have permissions to call.
Now, the Questions …
How many Minimum Phone Lines I should have, so that everyone can make calls simultaneously.
Which is the Best TDM Card / Hardware recommended
Should I go with VoIP or ATA ? (leave aside the PC)
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I’ll be posting more Question as I get response to my Questions. Thanks again to you guys.
[quote]* How many Minimum Phone Lines I should have, so that everyone can make calls simultaneously.
Which is the Best TDM Card / Hardware recommended
Should I go with VoIP or ATA ? (leave aside the PC)
[/quote]
I can’t really recommend how many phone lines you will need. That really depends on your calling volumes. You’ll need to take a look at your users and their calling patterns. Once you determine this you’ll have a better idea where to start. I would highly recommend that whatever you do you plan ahead and give yourself some room for growth on your system.
I have only used the TDM cards sold by Digium. They have performed very well for me. These are very easy to configure and support comes with the card in case you need it. There are other brands that probably work very well, but I can’t vouch for them, sorry.
I personally prefer to go with SIP. In my experience it takes less CPU to process SIP calls (when transcoding is minimized), so you can have more active calls on one box than with TDM cards. I also like the extra signalling features that are available with SIP like caller id. These are not always available on analog lines from your phone company or may cost extra. Lastly the taxes on SIP are usually lower, sometimes as much as ~20%
There are some problems with SIP. Firewall issues can be a pain. The quality of your Internet connection will dramatically effect your call quality. i.e. unless you have QOS on your internet connection (and most people don’t) there is a good chance you will get some calls that will be choppy. Also make sure you get service from a carrier that is willing to work with you and has good support. That will save you a ton of time and headaches.
Hey, Dan nice to see such a big reply, I thought may be I’ve to wait for a couple of days… but NO… pat came the reply… Thanks.
I’m working on this project & the clients are my 9 neighbors, Yes I’ll be connecting 10 people where I live using VoIP phones and PC’s – and according to your advice I’ll be dropping the plan for using the Analogue Phones.
Next thing, I want to make sure, every house is able to communicate to each other for FREE.
And they don’t find the line busy when they need to call outside the NETWORK. I’m presuming that 6 PSTN lines would do the trick, and if all the lines are busy they would hear a busy tone.
Next thing I don’t exactly know how asterisk takes care of Incoming calls, but I don’t want any person to miss their call. Either the calls should wait in the queue or after particular rings/time they should be transfered to the Voice Mail system.
Talking about the TDM Card, Which Model I should go for (Digium Only), I mean talking about TDM 400 I think that supports 3 lines from PSTN and 1 line out to the Network, I really don’t know about this …
Well for now I would stop here… Please correct my misconceptions & Possibilities…!!
[quote]I’m working on this project & the clients are my 9 neighbors, Yes I’ll be connecting 10 people where I live using VoIP phones and PC’s – and according to your advice I’ll be dropping the plan for using the Analogue Phones.
[/quote]
As an alternative to using PC’s you may want to look at a SIP telephone. It works just like your regular analog telephone, except has an RJ-45 plug instead of an RJ-11 plug. Grandstream has several that work well with Asterisk. They also seems to have the most reasonably priced phones.
I prefer using SIP when setting up a “internal” network to make free calls. It works well and there is a ton of hardware (phones, ata’s) available. It also seems to work over wireless 802.11a/b/g networks. Which may help you out.
Incoming calls will hit your Asterisk machine over the method you have chosen to receive. So if you have phone lines from your local company connected to a TDM card, you would configure the zaptel driver (zaptel.conf & zapata.conf) for the phone lines and configure your extensions.conf file for the lines. The zaptel driver handles the TDM signalling and Asterisk through you extensions.conf file handles the call processing.
You would probably want to look at the Digium 8xx model which can have at max eight fxo and / or fxs cards.
