Hello, Maybe try that:
In your incoming isdn context: [isdn-incoming] exten => s,1,Set(TIMEOUT(digits)=3) exten => s,2,WaitExten(2) exten => s,3,Dial(SIP/operator...) exten => 10,1,Dial(SIP/10) exten => 20,1,Dial(SIP/20) So if a call comes in Asterisk waits, 2 seconds for further digits dialed and if so jumps to the right extension in the context. Overlapdial should be yes. yours christian gansberger www.accm.at On 3 February 2011 20:45, Cassius Smith <[email protected]> wrote: > Hello, > I have an installation in Austria; ISDN service provided by Austria Telekom. > The main number of the service is 6 digits. Incoming calls may contain 2 > additional digits, which I then use to route the call to the correct > extension. Telekom sends me all the digits. > My problem is that when someone tries to dial an 8 digit number to an > extension from an outside analog phone, AT sends the call before they finish > dialing all 8 digits. Is there a way to prevent this, or to catch the > additional 2 digits somewhere in the stream? The receptionist is unhappy > because she gets all the 6-digit calls and must then transfer. > Is this a p2p vs p2mp issue? > Thanks in advance, > Cassius Smith > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
