/*! \brief Wait up to 16 seconds for first digit (FXO logic) */ static int firstdigittimeout =3D 16000; /*! \brief How long to wait for following digits (FXO logic) */ static int gendigittimeout =3D 8000; /*! \brief How long to wait for an extra digit, if there is an ambiguous match */ static int matchdigittimeout =3D 3000; -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Justin Killen Sent: Wednesday, July 10, 2013 3:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay So then, by saying "If the digits already dialed match an extension in the dialplan...wait 3 seconds...", then we're saying that asterisk has found a match, and the match is the bad-extension. Here is the bad-number context that is included: =20 [bad-number] include =3D> bad-number-custom exten =3D> _X.,1,Noop(bad-number, timeouts: absolute: ${TIMEOUT(absolute)} digit: ${TIMEOUT(digit)} response: ${TIMEOUT(response)}) exten =3D> _X.,n,ResetCDR() exten =3D> _X.,n,NoCDR() exten =3D> _X.,n,Progress exten =3D> _X.,n,Wait(1) exten =3D> _X.,n,Progress exten =3D> _X.,n,Playback(silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer) exten =3D> _X.,n,Wait(1) exten =3D> _X.,n,Congestion(20) exten =3D> _X.,n,Hangup =20 =20 =20 So then, what you're saying then is that if I was to remove this include, there would be no match in the dialplan and asterisk will wait for 8 seconds instead of 3? The next question then is how to accomplish this without using the wildcard (and how to change it in freepbx). =20 -Justin=20 ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Richard Mudgett Sent: Wednesday, July 10, 2013 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay =20 =20 =20 On Mon, Jul 8, 2013 at 12:14 PM, Justin Killen <jkillen at allamericanasphalt.com> wrote: I have an installation that has analog phones connected via T1 channel banks. I'm getting complaints from users that they will enter a partial number (eg 91213), then turn away to get the next few digits, and the system will start dialing before they have a chance to put in the rest of the dialing string. Is there a way to increase this delay? The only use these 4 dialing patterns: =20 Internal 3 digit numbers 91 XXX XXX XXXX (for backwards compatibility) 9 XXX XXXX (also for compatibility) XXX XXXX =20 The simple switch in chan_dahdi has two hardcoded timeout times for more digits. 1) If the digits already dialed match an extension in the dialplan but could match another extension if more digits are dialed then chan_dahdi will wait 3 seconds for more digits to arrive. 2) If the digits already dialed do not match any extension in the dialplan but more digits could match an extension then chan_dahdi will wait 8 seconds for more digits. The shorter timeout is so the caller won't have to wait too long if the caller intends to call the shorter dialplan extension. You need to look at the extension patterns in your dialplan to see where you have ambiguity between extensions. Are you using the '.' wildcard? =20 Richard =20 -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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