I needed some help on my asterisk 1.8 system. I install asterisk to manage one of my endpoint to do call termination but i am encountering some problems.
I am able to make calls setting up inbound and outbound trunk but my problem is that from the time the codec establish billing start and this is consider as FAS another problem is that i am getting at interval IVR prompt from the carrier end.
How can i adjust or eliminate the FAS problem which mean from the time the call hit the carrier and gets a ring tone it started billing?
Also is it possible for me to setup a responsive system where by based on what the carrier IVR say. e.g press 1 to connect this call or press 5 to connect this call, can asterisk be able to do this automatically then upon when the party picks up then billing start.
I’ll appreciate if anyone can assist if it is possible or not 
Help!!!
There is no reasonable alternative for FAS in the Wikipedia disambiguation page, so I have failed to work out what you mean.
There is no provision in Asterisk to send DTMF as early media, only after the called party answers… If this network operator dialogue isn’t done as early media, called party answering will answer the calling party.
There would be no easy way of detecting the prompts from the network, so any DTMF would have to be sent by dead reckoning. You might be able to achieve it by using Originate to start the outgoing call and then Bridging it to the incoming call once it has been set up.
Thanks David - I know it is not something to achieved that easy but the latter part i am clueless - can you possible outline a scenario - please!
Call comes in. Issues Originate to a local channel, and to an extension that dials the target number. Local channel answers then waits an appropriate amount of time before sending the DTMF. Finally, it issues a channel redirect to the original call sending it into code which issues a bridge to called party channel. Use Wait on the original call and at the end of the dialplan that sends the DTMF, to stop them hanging up before the call is bridged.
If the DTMF is actually early media, you can use G on the main Dial to catch the answer and bridge the outgoing leg to the waiting incoming call.
You will need to pass variables around using local and global variables, and possibly carefully constructed extension names. You may also have to use ${IMPORT()}.
Is there a way you can assist me with the script. I dont mind pay you for your work i am not that savy yet with asterisk… I’ll really like to give it a try or maybe we can setup a login for you to do it on the asterisk itself. either way please let me know.