Advice on buidling an Interactive Voice Response (IVR) system using Asterisk

Question

What are some good resources that i can start with? I don't have any hardware yet, What advice can you give me if i want to start development/testing without hardware?

Answer 1

Diagram the prompts and have someone else test the flow. Try to use the same voice talent, if externally, make sure there is a contract in place so he or she remains available. We have been pretty happy with SipX and if you are not set on Asterisk you may want to check them out as well.

/Allan

Answer 2

Grab a SIP phone or IAX Phone - there are several free ones depending on your operating system.

You can also uses services such as connect.voicepulse.com to give your Asterisk machine a phone number without a physical hard line (all over IP).

Also make sure you have a microphone or headset - its hard to record voice prompts without one :-)

Answer 3

get the free ebook from o'reilly: Asterisk: The future of telephony.
lots of info at: http://www.voip-info.org download "asterisk appliance" distribution like asterisknow. It comes with os, dependencies and asterisk already setup, so you can focus on dial plan and onwards

Answer 4

suppose you have three computer. One pc have linux and this server. Other two pcs are windows system based and clients. install asterisk software in linux server. Others two pcs sip software is xlite software. then you configure extensions.conf and sip.conf.

extensions.conf [phones] exten => 3000,1,Dial(SIP/3000,30) exten => 3000,2,Answer() exten => 3000,3,Playback(sound_file_name) exten => 3000,4,Hangup()

exten => 3001,1,Dial(SIP/3001,30) exten => 3001,2,Answer() exten => 3001,3,Playback(sound_file_name) exten => 3001,4,Hangup()

sip.conf

[3000] type=friend secret=3000 host=dynamic allow=all

[3001] type=friend secret=3001 host=dynamic allow=all

Then finally u configure xlite soft phone.

Answer 5

I have deployed several IVR's with Asterisk and what I usually do is test them on my machine by setting up Asterisk with just SIP, having one SIP user which I set up on X-Lite. This user is set up with a test context where I have several extensions which redirect the call to some other extension (s or something else) on another context which belongs to one of the IVR configs I want to test.

In extensions.conf at the end I have several #include lines to import the IVR configs, I just comment out the ones I don't need and leave only what I'm about to test. Start Asterisk, run X-Lite and dial the proper extension and it will be just as if you call your IVR in deployment. If you depend on caller ID for authentication then remember to set the callerid field in your SIP user's entry.

Set the X-Lite's mic volume to zero, because you don't need to transmit voice at all, only DTMF and that's independent of the mic volume.

Record some prompts yourself with Audacity or whatever recording software you have at hand, doesn't matter how lousy they are, it's just so you can move forward and make demos/mockups. Put all the prompts that you make in a dir inside /var/lib/asterisk/sounds so that you can easily tell them apart from all the prompts Asterisk includes. Call them like so:

Playback(yourdir/yourprompt)
Read(YourVar,yourdir/yourprompt,5)

Or you can set the language (globally or per channel) to "yourdir" and then you just Playback(yourprompt) etc.

Shameless plug: Asterisk now includes AEL, but if you still want to use the old dialplan format, check out my Asterisk Dialplan Compiler.