Asterisk IVR support

Asterisk Asterisk Support
1

I am a new user to asterisk and I want to setup an ivr in my office, but before making changes I want my few questions to be answered.

  1. I have a single PSTN line in my office, one with the RJ-11 connector. How will I connect this to my asterisk installed computer?
  2. Will the single PSTN line be able to handle multiple calls?
  3. I want the IVR to serve data from remote database. Is it possible? If it is, can it provide different results to multiple users at the same time?
  4. What is the best open source bundle like AsteriskNOW to my problem?

Please Help.

2

[quote=“sushilthe”]I am a new user to asterisk and I want to setup an ivr in my office, but before making changes I want my few questions to be answered.

  1. I have a single PSTN line in my office, one with the RJ-11 connector. How will I connect this to my asterisk installed computer? [/quote]
    You need a fxo gateway (there are usb gateways or SIP gateways)or a Internal Card(like digium, sangoma or openvox cards).

[quote]
2. Will the single PSTN line be able to handle multiple calls?[/quote]
No, analog lines only handle 1 call per line/wire.

[quote]
3. I want the IVR to serve data from remote database. Is it possible? If it is, can it provide different results to multiple users at the same time?[/quote]
Global answer, yes but you need to create that.

[quote]
4. What is the best open source bundle like AsteriskNOW to my problem?[/quote]
If you dont know much about asterisk, yes FreePBX based distro will help you a lot.

3

If PSTN lines/analog lines cannot handle multiple calls, then will the queue() in asterisk help me achieve that?

4

No. Analogue lines cannot signal for more than one call at once. Basically an incoming call puts ringing voltage across the two wires, combined with about 48 volts DC. When Asterisk answers the call, it loads the line down to a lower voltage. The speech is then sent over the wires, bascially the same as for a headphone cable. When Asterisk stops loading the line, and the voltage goes back up, the call clears.

If you want to queue incoming calls for an analogue line, your network operator will need to do that…

5

A PSTN phone line is an analog phone line. You need a FXO card which will have an RJ-11 connector to terminate it in to your Asterisk PC and digitize the audio. It will look similar to a modem card if you want to think of it like that.

A single PSTN line can only handle 1 phone call. So if this is the only way calls can come in to your office you will be limited to one caller. Asterisk doesn’t give you extra phone lines. If you want to avoid hardware cost and have internet connectivity you can look at using SIP as a way to bring more lines in to your office.

Basically there are two types of IVR. You can create menus for users to navigate menus directly in the Asterisk dialplan. When you need to dip databases you can use AGI to write an IVR in any language. Think of AGI is to Asterisk as CGI is to Web Servers. Want to dip a database… it’s just like your using PHP only instead of echo “balance is $dollars” you have to use agi->PlayBack(“BalanceIs”), agi->SayDigits($dollars);

Depends on your comfort level. You can use stock asterisk in just about any distro to do this. For example in Ubuntu you can apt-get install asterisk. But it won’t have a flashy web interface. Which might be a good thing for you to learn the basics. Get it installed, get some documentation, and keep posting, others will help you. Good luck.

6

Can multiple calls be handled if i get the SIP or VOIP account? The SIP/VOIP also uses the same PSTN line. Will it be possible? How will i connect it to computer for handling multiple calls?

7

Depends on the carrier You choose. You have to look at the amount of (inbound/outbound) channels the provider gives You. As far as I understand Your demands, You’d like to have a SIP-Account combinend with one ore mor DIDs to call You. So You have to look for a provider not only selling You VOIP-channels but also one ore more DIDs.

And: If You have at least more than one available channel connected to Your asterisk, You’ll be able to handle multiple calls at the same time (as much as You have channels).

No, that’s a mismatch: Your existing analog line and any additional SIP-Line aren’t the same and will definitely have different DIDs (for the experts: except ENUM, but I think thats not relevant here :unamused: ). If You want to keep Your analog DID and use it in a Multi-Channel environment You may

a) ask Your telephone-provider to set up a digital multi-channel termination (BRI,PRI a.so.) to You
b) port/transfer Your DID from Your telephone-provider to a multi-channel VOIP-provider

However: The different technologies may not share the same DID.

For the analog line see posts before. If You upgrade a analog line to something else (BRI, PRI, T1, E1) answer is still the same except that You need another card / gateway than for a single analogue line (where a single FXO-port fits).

To connect a VOIP/SIP-provider You just need to connect the computer hosting Your PBX to the Internet. As VOIP uses Your Internet-connection the amount of parallel calls is - beside the characteristics of the contract with Your provider - dependent on the characteristics (Up- and Downspeed) of Your Internet-connection. Normally You would need 80 Kbit/s in either direction for each call (codec alaw/ulaw, compressed codecs use less, g722-HD-codec or some video-codecs use more)

8

I think he means he has xDSL over tthe same copper pair has his phone line. The number of voice calls that will fit into an ADSL connection are limited by its uplink speed.

Note that some ITSPs sell services intended for IP phones, not for for PABXes, and may limit to one call, as a commercial policy, not for technical reasons.

(If the SIP is being carried over a baseband modem connection, you will get less than one full telephone quality call over the line, but maybe a couple of voice only, reduced quality ones, assuming that the other end supporsts suitable codecs.)

Generally, though, I think the OP needs to do a lot more background reading, particularly of Asterisk the Definitive Guide, as they don’t have enough basic knowledge on telecoms to attempt what they are trying to do.

9

This is getting off topic in terms of Asterisk in particular but Open Source, Linux and Asterisk has been very kind to me so I’m going to pass long what I know. The question your asking is hard to answer without knowing how many phone lines you need.

If you are in the 2-3 phone lines cause you want some sort of after hours service, or basic call tree for in a small company to get to the right person then get 2-3 lines installed at your business and get up to a 4 port FXO card from Digium.

If you are trying to get a modest IVR system that might support 5-10 callers and perhaps there is an Asterisk PBX to support the office get a fractional T1 from an internernet provider where some of the 24 channels are internet some are phone lines and get a T1 card from Digium.

If you are trying to support 24 callers get a T1 / PRI, if you need 48 get 2 T1 (they come in 24 lines per T1, not exactly but don’t want to explain it to you why not).

If you are trying to get 100+ phone lines in to a call center and can get 4-5MB of internet bandwidth, from multiple providers, you can ge tSIP (digitial voice over ip) phone lines from most phone companies but you have to buy the bandwidth and then pay for SIP “sessions” (think of them as phone lines).

We have 300+ SIP phones from two providers and two internet carriers for redundancy. If you looked around there isn’t a single traditional “phone line” in the place. Crazy idea but our phone system gets to the phone lines over the internet, and you see no dedicated “phone hardware” anywhere.

10

I’ll add this too because it was confusing to me because Astersk never addresses “telecom” per say. It assumes you know these things.

When you get in to the multiple phone line world, the phone line is not dedicated to a particular number. In other words a T1 line that has 23 phone line bundled in it is not restricted to 23 phone lines. 281-501-1000 does not ring line 1, 281-501-1001 does not ring line 2, etc…

When you step up to this class of connection with the phone company they can send the phone number that was dialed down any of the lines. 281-501-1000 can be dialed, and sent down any of the 23 phone lines in a T1.

In Asterisk you don’t really care cause you just match on the digits dialed.

exten => 2815011000,1,Answer()
exten => 2815011000,2,Dial(SIP/1000)

It doesn’t matter that it came in on line 1,2,3,4, or 23 of the T1. Now outbound it makes a bit of a difference, in that you cannot dial out on line 1 if it is in use. So what you do there is you treat it as a group and dial out on any available line.

In other words while you could dial DAHDI/01, DAHDI/02, DAHDI/03 to address the individual lines you would not typically do that. You would treat them as a group and dial DAHDI/g1 and let Asterisk figure which line was free.

So same thing in reverse that the phone company does not treat T1 line 1 as 281-501-1000, T1 line 2 as 281-501-1001, you tell the phone company what digits were dialed and what even line was available. They don’t care if it came in on line 1 or 23, it just has to be an open line.