Help configuring asterisk for incoming calls

Hi, sorry if this is a duplicate to some other topic. I’ve been given the task of getting an Asterisk server to receive incoming calls in the event our main phone system crashes. If our phone number is through a local telco and not a voip provider, how do I configure sip.conf and extensions.conf? Right now, our main phone number is active…so if our main phone system dies, wouldn’t I have to have the telco route the incoming phone calls to my asterisk server?

I’m a little confused.

how does the lines connect to your main system? T1 … individual pots lines?

What are the phones at the desks analog or some digital system? If they are phone system specific then you probably wont be able to use them with the Asterisk box in a failure. So how then do you want people to get or make calls?

I managed to stick in an asterisk box that had T1 cards accepting the calls from the telco…then bridge them out another T1 port which went to our old Nortel Meridian 1 system.

In the event of a failure, that box was able to play messages or direct calls to a SIP phone.

[quote=“mazzic”]how does the lines connect to your main system? T1 … individual pots lines?

What are the phones at the desks analog or some digital system? If they are phone system specific then you probably wont be able to use them with the Asterisk box in a failure. So how then do you want people to get or make calls?

I managed to stick in an asterisk box that had T1 cards accepting the calls from the telco…then bridge them out another T1 port which went to our old Nortel Meridian 1 system.

In the event of a failure, that box was able to play messages or direct calls to a SIP phone.[/quote]

first question you asked…I will find out; I’m new to this project.

All of our phones in the office are Nortel IP Phones…just plugin to Ethernet ports in each cubical. I know I won’t be able to use them. The goal for this project…if the main phone system goes down, we just need some way that incoming calls will still ring to 1 phone…don’t care about extensions for all the employees or anything yet. Simply, when someone dials our main line, and the phone system is down…Asterisk server comes into play and a single phone rings until we can get the main phone system back up.

That help?

Still need more info. You have a nortel system, how do the lines connect to it? T1, PRI, Pots lines? Once you know that you can figure out how to do it and what you will need to do it. Pots lines are the easiest and cheapest way. Also which Nortel IP phones, they may work with asterisk…
voip-info.org/wiki/view/Nortel+Phones

[quote=“riddlebox”]Still need more info. You have a nortel system, how do the lines connect to it? T1, PRI, Pots lines? Once you know that you can figure out how to do it and what you will need to do it. Pots lines are the easiest and cheapest way. Also which Nortel IP phones, they may work with asterisk…
voip-info.org/wiki/view/Nortel+Phones[/quote]

I found out that we have a T1 PRI connecting to our main phone system. At this point, let’s assume that the nortel phones do not work…and that we are going to configure a software phone, or a compatible hardware phone with the Asterisk server. Is this enough info to get me started?

I would assume but you still need to know the T1 settings from your provider. You will also need to order a T1 card for your asterisk box, and configure it. You will need to know what type of encoding, and how many channels, also if you have DID’s and what they are…
Here is one example of a card you would need, telephonydepot.com/Catalog/D … ium-TE121B

I would look into your options on cards and see which you like. The easiest way is once the other pbx has a problem you move the T1 from the Nortel pbx to the asterisk system. You should be able to have a softphone work with it all. I would start reading downloads.oreilly.com/books/9780596510480.pdf
good luck!

One of our other engineers already installed a T1 in the Asterisk box and configured it.

More information, someone please help :smile:

The T1 is already configured and such. I’ve been told by another engineer that you can connect with a sip client and make calls out.

I know what my DID’s are and basically what I need to accomplish is to connect a voip phone to the system…configure Asterisk for incoming calls for all of our DID’s to ring to the voip phone. We are using DAHDI and not Zaptel.

I looked at /etc/asterisk/chan_dahdi.conf and there is a [channels] section with this under it:
context=default

Does that mean that my extensions.conf file needs to have [default] as a context with the applications/instructions beneath it? And, if the goal is to ring to 1 voip phone when someone calls my DID’s, should the context look something like:
[default]
exten => s,1,Answer
exten => s,n,Dial(SIP/phone)

? =\ Sorry, I’m still a little confused about this.

Yes, that sounds right.

You don’t need the Answer line though. You really don’t want to answer the call…do the dial and try passing the call to the sip phone. If they answer, then the call will be answered.