Problem when calling to numer that does not exist


First of all i want to thank you for this great piece of software :smile:

I will begin with my hardware specification:

  • server asterisk ( installed on simple core i5 computer
  • softphone X-Lite 3.0 bulid 30942

Server works with vicidial. And it works well except one problem.

I can dial manualy from my softphone.
I dial to a number that doesent exist (it was disconnected from land line operator).

And there we have our problem. When i call to this number (or any number that is disconnected) i want immediate disconnect from asterisk (Everyone is busy/congested at this time, and status 1 CHANUNAVAIL). One every three or four times it works that way.

But most of the time i must wait 16 second to disconnection.

Also when i remove “r” from my dialplan i can hear a women voice from land line operator “this number does not exist” - she speaks exacly 16 second and there is diconnection.

My dialplan looks like this:
exten => _7ZXXXXXXXX,1,AGI(agi://
exten => _7ZXXXXXXXX,n,Dial(${NETIA3}/${EXTEN:1},tTor)
exten => _7ZXXXXXXXX,n,Hangup

My voip operator said that it is problem with" 183 session progress" and with “early media”.

What can i do to solve this problem? It is necessary that when i dial to number that does not exist i get imediate disconnection.

Thanks in advance for your time :smile:

Buy out the public network operator and change the configuration of their switches.

The use of in-band early media voice messages, rather than out of band signalling is a choice of the network operator. You cannot force them to signal the condition immediately or even at all.

If you hear three tones, at the start of the message, these may encode the reason, but Asterisk doesn’t have facilities for decoding SIT.

Unfortunately this not an option :smile:

But why one time it works good and another time it doesent?

My VOIP operator send me this message:
“In the first 16 seconds there is send an Announce from land line operator with communicate 183 session progress. Probably as a rosult of this communicate your equipment (my asterisk server) issues signal calling 183.Please verify the configuration of early media”

They are confirming that it is the network that is not sending the reason.

It would not be normal for a SIP user agent to drop a call on receiving 183, as the in-band progress can be ringback tone, as well as failure indications. Asterisk has no provision to do this.

Note that Asterisk won’t actually forward the early media unless you call Progress before you call Dial.

So the problem will not solve if i use this dialplan?:

exten => _7ZXXXXXXXX,1,AGI(agi://
exten => _7ZXXXXXXXX,n,Progress
exten => _7ZXXXXXXXX,n,Dial(${NETIA3}/${EXTEN:1},tTor)
exten => _7ZXXXXXXXX,n,Hangup

I don’t know what the agi script does.

The Progress call should mean that you hear the message telling you why the call fails.

The Hangup is redundant.