They won't catch, no (because of priority), but they do match, which is enough to trigger the 3 second timeout instead of the 8 second. So, if you pickup and dial 1, then you will only get 3 seconds (instead of 8) to type in the next digit before it considers it done. The issue I am describing is compounded by the fact that the patter is _X. instead of _X but the core issue is the same - only getting 3 second inter-digit timeouts instead of 8.
-Justin -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursday, July 11, 2013 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay The catch alls do not catch 1+ or 3+ calls. Look carefully at it. Therefore there will not be a delay. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, July 11, 2013 3:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay Right, but when you type any of those, there's only a 3 second inter-digit timeout because EVERYTHING is a match of the catch-all. There is no excessive delay, but instead a delay so short that I'm getting complaints. If I implement your suggestion and change the code in the channel driver, then there would be an 8 second delay all the time, even when dialing a number like 3001, which IMHO is excessive (and what I was referring to in the previous post). So, again: my two options as before: 1) Have the timeout be so short (3 seconds) that users complain (but they get a fancy message). 2) The timeouts are reasonable (8 seconds), but when they're wrong the users get a busy signal (no fancy message). Plus we can add a third option: 3) Alter chan_dahdi.c to increase matchdigittimeout to 8 seconds, then: The timeouts on invalid extensions are reasonable (8 seconds), but timeouts are valid extensions are excessive (8 seconds), and we get a fancy message. It's a shame that reasonable timeouts and a nice message are mutually exclusive. -Justin -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursday, July 11, 2013 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay This issue is simple dialplan management, which applies to any PBX. This is something every PBX admin has to deal with. Here is an example using 4-digit extensions in the 3xxx range and outside calls are dialed with a leading 1 so the PBX knows it is an outside call. There should be no excessive delay when dialing extensions or PSTN numbers in the setup below. Calls should match when the last digit is dialed for those calls. For invalid numbers there will, of course, be a delay. exten => _1NXXNXXXXXX,1,DoYourOutsideDialing exten => _3XXX,1,DoYourInsideDialing exten => _[24-9].,1,DoErrorHandling exten => _X,1,DoErrorHandling -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, July 11, 2013 1:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay No, I understand - maybe I'm not explaining myself well. Yes, I can change the source so that pattern-matched input delays 8 seconds instead of 3, but then the users have to wait 8 seconds for every number they dial (even internal 3 digit calls). I think what I really want is for the catch-all pattern to not trigger the shorter timeout. It seems to me that if 3/8 second timeouts are standard and a catch-all for fancy messages is commonplace, then the two should work together without too much trouble, but instead they are currently mutually exclusive. I realize that a code change will be required to accomplish standard 3/8 second wait times AND be able to get a fancy message (I'll be submitting an issue to jira - I'm thinking add a special 'no pattern matched' extension like i or t). For the time being, I have the catch-all disabled at the site and things are running smoother. Thanks Eric for your help on this - you helped me to track down the cause of the issue and provided a work-around, which is much appreciated. -Justin -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursday, July 11, 2013 9:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay You seem to be confused. If you want to change the dialing timeouts on Asterisk analog channels, then you need to change the source code. Now your dialing timeout problem is fixed. I did that about 10 years ago to handle slow dialing users on asterisk analog ports. Then add a catchall pattern for bad numbers and your congestion tone is fixed. done! -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, July 11, 2013 12:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay So my only two options then are: 1) Have the timeout be so short that users complain (but they get a fancy message). 2) The timeouts are reasonable, but when they're wrong the users get a busy signal (no fancy message). It's a shame that reasonable timeouts and a nice message are mutually exclusive. --Justin -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursday, July 11, 2013 7:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay I imagine setting up a catch-all extension pattern is your best option. That is what most seem people do. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Wednesday, July 10, 2013 4:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay Okay, so I is no good. Does anybody else have a work-around for this? -Justin -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, July 10, 2013 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay "I" has the same limitations as dialplan timeouts, you have to be in a Background or WaitExten or similar for them to work. These items are designed for IVRS. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Wednesday, July 10, 2013 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay It seems likely that this is exactly what is happening. I'd rather not change the code though, but rather fix the dialplan. I'm thinking using the 'i' extension would work just the same - would there be a reason to use a wildcard pattern match instead of i? -Justin -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, July 10, 2013 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay >From chan_dahdi.c, don't know if it applies to your situation or not. /*! \brief Wait up to 16 seconds for first digit (FXO logic) */ static int firstdigittimeout = 16000; /*! \brief How long to wait for following digits (FXO logic) */ static int gendigittimeout = 8000; /*! \brief How long to wait for an extra digit, if there is an ambiguous match */ static int matchdigittimeout = 3000; -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@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: [bad-number] include => bad-number-custom exten => _X.,1,Noop(bad-number, timeouts: absolute: ${TIMEOUT(absolute)} digit: ${TIMEOUT(digit)} response: ${TIMEOUT(response)}) exten => _X.,n,ResetCDR() exten => _X.,n,NoCDR() exten => _X.,n,Progress exten => _X.,n,Wait(1) exten => _X.,n,Progress exten => _X.,n,Playback(silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer) exten => _X.,n,Wait(1) exten => _X.,n,Congestion(20) exten => _X.,n,Hangup 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). -Justin ________________________________ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@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 On Mon, Jul 8, 2013 at 12:14 PM, Justin Killen <jkil...@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: Internal 3 digit numbers 91 XXX XXX XXXX (for backwards compatibility) 9 XXX XXXX (also for compatibility) XXX XXXX 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? Richard -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
