On Thu, Sep 8, 2011 at 9:38 AM, Kevin P. Fleming <kpflem...@digium.com> wrote: > On 09/07/2011 11:06 AM, Daniel Tryba wrote: > >> The aim of the quest for overlap dialing is to let the user enter a >> number at their own pace but immediatly dial when all digits are >> received (just like plain old ISDN does). My trunk is a bunch of E1 PRIs >> in overlap mode. The following just works for any SIP client (without >> overlap dialing): >> exten => _X.,1,Answer() >> exten => _X.,n,Dial(${TRUNK}) > > Unless I'm mis-remembering, this was the point of adding the '!' dialplan > match character. If you use _X!, and you have your SIP endpoints configured > to send an INVITE as soon as the user has entered two digits (and you have > no other patterns in the context that could match), then the dialplan will > match against that and initiate a Dial() on your ISDN PRI. Since the number > is not yet complete, the SETUP message on the PRI won't result in the call > proceeding, and as the user of the phone presses additional digits they'll > be sent to Asterisk as DTMF, bridged over to chan_dahdi, and it will send > them as INFORMATION messages rather than as DTMF digits, because it knows > the outbound call is still in 'dialing' state. > > However, this is still going to 'mess with CDRs' as you put it, because the > only switch in the network that knows the complete number that was dialed is > the PSTN switch that your PRI is connected to. It seems possible that > chan_dahdi could 'update' the EXTEN on the current channel as the additional > digits are dialed so that the CDR contains the complete number, but I have > no idea whether it does or not. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org
Exactly Kevin. I remember now that I was using it for my http://etel.wiki.oreilly.com/wiki/index.php/Asterisk_Man_in_the_Middle in some setup/testing. -- ~ Andrew "lathama" Latham lath...@gmail.com http://lathama.net ~ -- _____________________________________________________________________ -- 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