Attention asterisk professionals. "help!"

I am having problems with False Answer Supervision (FAS) while using Asterisk and Digium FXO cards.

We are sending phone calls to Russia from a USA Gatway/VOIPswitch. The serverers in Russia use Asterisk utilizing Didium 4 port FXO cards. There are 3 cards per server (12 ports). We have twelve phone line attached.

The problem is when Asterisk selects/seizes a channel/port to dial out a call it gives immediate Answere Supervision to my client even though the call had yet to be dialed at the server in russia. it showes answered even though there is ring no answer. busy ton, etc etc.

I have been told there might mbe a patch or a solution for this.

$$ to anyone that can help. PLease contact me at ASAP…



It would appear that someone is proactively answering somewhere in the call setup process. What would be necessary is obtaining a view into the extensions.conf/dialplan of the Asterisk server in Russia that is handling the call through process.

The problem is that when you are using analog lines and you place a call the technology does not provide any way to detect that the party you are calling has answered.

Because of this the FXO cards are set to assume the line is answered immediately.

You can stop this by seting callprogress=yes in zapata.conf, but it is experimental and did not work for me

Here is some possible solution from another post

1-you can use a sip gateway ( such as cisco routers) and send your calls from asterisk to sip gateway by SIP protocol and cisco router send your call to PSTN network. ASTERISK <====>SIP gateway <====> PSTN

2- if you dont have SIP gateway , you can use NVlinedetect application code for this problem :


callprogress works to eliminate the “always answered” reporting, but it doesn’t solve the problem completely.

All calls either report ANSWERED or NOT ANSWERED. It doesn’t seem matter if the call was busy. Asterisk is always reporting 16 NCC. It sure would be nice if Asterisk would report 3 or 34 for accuracy.

Any insights?