On Thu, 11 Jul 2013 13:53:27 -0700 Justin Killen <jkil...@allamericanasphalt.com> wrote:
> 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. Well, if you want an 8 second inter-digit timeout, you can do that by changing the DAHDI source. If you don't have any ambiguity in your extensions, you'll never have anyone waiting 8 seconds after they've finished dialing, because once they've dialed a valid number (which would match only one extension), it continues instantly without any timeout at all. So it looks like you'll need both fixes--and then you can have it all. > -----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? 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