Re: [asterisk-users] Shorten time between DTMF
Which EXACT parameter did you change in asterisk.conf? Changing DTMF duration for DAHDI is done in chan_dahdi.conf. SIP DTMF duration and inter-digit duration is generally set on the phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of CDR Sent: Friday, June 06, 2014 1:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Shorten time between DTMF I already shortened the DTMF duration, but I need to change the time elapsing between them. The first thing I achieved by changing a parameter in asterisk.conf, but how do I conquer the second goal? -- _ -- 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
[asterisk-users] Shorten time between DTMF
I already shortened the DTMF duration, but I need to change the time elapsing between them. The first thing I achieved by changing a parameter in asterisk.conf, but how do I conquer the second goal? -- _ -- 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
Re: [asterisk-users] Using macros in extensions.lua?
On Fri, Jun 6, 2014 at 1:48 AM, Dennis Guse wrote:
> Hi,
>
> I have defined a dialplan in lua and now would like to use "dial" with the
> macro M to implement some logic, when the callee-channel gets created.
>
> Working old style would be (extensions.conf)
>
> [default]
> exten => _X,1,dial(SIP/1,,M(mymacro^parameter))
>
> [macro-mymacro]
> exten => s,1,verbose(${ARG1})
>
> How to implement the same functionality using pbx_lua?
>
> Details: Asterisk 11.7 on Ubuntu 14.04
>
> Kind regards
>
> Dennis Guse
>
> Here's how I do it for pre-dial handlers...
extensions.handlers = {
["addheader"] = function(c,e)
channel.PJSIP_HEADER('add', "Alert-Info"):set(";info=custom1")
end;
}
extensions.local_default = {
[""] = function(c,e)
app.dial('PJSIP/'..e,nil,'b(handlers^addheader^1)')
end;
}
--
_
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Re: [asterisk-users] Problem reload queue dynamical members
Josh, thanks for the feedback. That problem can also occur with dynamic members, would not be just for those who work with realtime? tks 2014-06-06 10:14 GMT-03:00 Josh Metzger : > On Fri, Jun 6, 2014 at 9:03 AM, Eduardo Leones < > edua...@ypytecnologia.com.br> wrote: > >> >> Guys, I have a problem. I have a queue on asterisk 1.8 that members are >> added dynamically via the AMI QueueAdd. When you run the CLI a >> "reload app_queue.so" all members who were in the queue disappear. This is >> a bug or some parameter that I do not know? >> >> Would have another way to do the reload queue without any risk to members >> who are already in it? >> >> > It depends on which exact version of 1.8 you're running, but this appears > to be a bug that was fixed in April of this year. From the changelog for > 1.8: > > 2014-04-01 16:48 + [r411584] Joshua Colp > > * apps/app_queue.c: app_queue: Fix a bug where realtime members > would be deleted during reload causing waiting callers to get > ejected. This patch causes realtime queue members to remain in > queues during the reload process. Previously these members would > be removed causing any waiting callers to be ejected from the > queue with a reason of "EXITEMPTY". ASTERISK-23547 #close > ASTERISK-23547 #comment Patch > app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo > Rossi (license 6409) Review: > https://reviewboard.asterisk.org/r/3404/ > > > -Josh > > -- > _ > -- 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
Re: [asterisk-users] Problem reload queue dynamical members
On Fri, Jun 6, 2014 at 9:03 AM, Eduardo Leones wrote: > > Guys, I have a problem. I have a queue on asterisk 1.8 that members are > added dynamically via the AMI QueueAdd. When you run the CLI a > "reload app_queue.so" all members who were in the queue disappear. This is > a bug or some parameter that I do not know? > > Would have another way to do the reload queue without any risk to members > who are already in it? > > It depends on which exact version of 1.8 you're running, but this appears to be a bug that was fixed in April of this year. From the changelog for 1.8: 2014-04-01 16:48 + [r411584] Joshua Colp * apps/app_queue.c: app_queue: Fix a bug where realtime members would be deleted during reload causing waiting callers to get ejected. This patch causes realtime queue members to remain in queues during the reload process. Previously these members would be removed causing any waiting callers to be ejected from the queue with a reason of "EXITEMPTY". ASTERISK-23547 #close ASTERISK-23547 #comment Patch app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo Rossi (license 6409) Review: https://reviewboard.asterisk.org/r/3404/ -Josh -- _ -- 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
[asterisk-users] Problem reload queue dynamical members
Guys, I have a problem. I have a queue on asterisk 1.8 that members are added dynamically via the AMI QueueAdd. When you run the CLI a "reload app_queue.so" all members who were in the queue disappear. This is a bug or some parameter that I do not know? Would have another way to do the reload queue without any risk to members who are already in it? tks Ed -- _ -- 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
Re: [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)
On Fri, Jun 06, 2014 at 08:28:07AM -0400, John Novack SCII wrote: > A J Stiles wrote: > >On Thursday 05 Jun 2014, Mojtaba wrote: > >>My scenario is (2) > >After doing some tests with my own hardware, I'm now convinced that this is > >actually normal behaviour: As far as Asterisk is concerned, a call is deemed > >"answered" as soon as the hardware seizes the line. It is only "not > >answered" > >if the line is not available. > > > >Which makes sense, because an analogue line has no D-channel. Once the trunk > >is acquired successfully, there is no way for a machine to know the state of > >the call beyond then. Such supervisory information as there is -- a regular > >cadence during ringing, possibly a burst of white noise and then a human > >voice > >-- is geared towards interpretation by human beings. > > > >Moerover, since the tones are different in every country (and sometimes, > >between different telephone exchanges in the same country; at one time, the > >UK > >was using three sets of supervisory tones depending whether you were on an > >old-fashioned "clicky-clicky" exchange, an intermediate-generation analogue > >electronic exchange or System X) it would not be a trivial task to make > >sense > >of them. > > > > > >I think if you want full supervisory information, you are going to need to > >use > >some sort of digital telephony technology (ISDN or GSM). > > > This is well known behavior for many years, since the inception of > Asterisk/Zaptel > I wonder why tests had to be run! > The OP issue was answered several days ago > His issue was obvious and well stated until another poster confused the issue! Just to keep it clear for anyone who stumbles on this thread in the future, this can sometimes work if you set the callprogress=yes option in chan_dahdi.conf if your country/provider/exchange is supported. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- 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
Re: [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)
A J Stiles wrote: On Thursday 05 Jun 2014, Mojtaba wrote: My scenario is (2) After doing some tests with my own hardware, I'm now convinced that this is actually normal behaviour: As far as Asterisk is concerned, a call is deemed "answered" as soon as the hardware seizes the line. It is only "not answered" if the line is not available. Which makes sense, because an analogue line has no D-channel. Once the trunk is acquired successfully, there is no way for a machine to know the state of the call beyond then. Such supervisory information as there is -- a regular cadence during ringing, possibly a burst of white noise and then a human voice -- is geared towards interpretation by human beings. Moerover, since the tones are different in every country (and sometimes, between different telephone exchanges in the same country; at one time, the UK was using three sets of supervisory tones depending whether you were on an old-fashioned "clicky-clicky" exchange, an intermediate-generation analogue electronic exchange or System X) it would not be a trivial task to make sense of them. I think if you want full supervisory information, you are going to need to use some sort of digital telephony technology (ISDN or GSM). This is well known behavior for many years, since the inception of Asterisk/Zaptel I wonder why tests had to be run! The OP issue was answered several days ago His issue was obvious and well stated until another poster confused the issue! John Novack -- _ -- 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
Re: [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)
Thank you very mush for your good replying. M.Esfandiari.S -- _ -- 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
Re: [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)
On Thursday 05 Jun 2014, Mojtaba wrote: > My scenario is (2) After doing some tests with my own hardware, I'm now convinced that this is actually normal behaviour: As far as Asterisk is concerned, a call is deemed "answered" as soon as the hardware seizes the line. It is only "not answered" if the line is not available. Which makes sense, because an analogue line has no D-channel. Once the trunk is acquired successfully, there is no way for a machine to know the state of the call beyond then. Such supervisory information as there is -- a regular cadence during ringing, possibly a burst of white noise and then a human voice -- is geared towards interpretation by human beings. Moerover, since the tones are different in every country (and sometimes, between different telephone exchanges in the same country; at one time, the UK was using three sets of supervisory tones depending whether you were on an old-fashioned "clicky-clicky" exchange, an intermediate-generation analogue electronic exchange or System X) it would not be a trivial task to make sense of them. I think if you want full supervisory information, you are going to need to use some sort of digital telephony technology (ISDN or GSM). -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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
[asterisk-users] Using macros in extensions.lua?
Hi,
I have defined a dialplan in lua and now would like to use "dial" with the
macro M to implement some logic, when the callee-channel gets created.
Working old style would be (extensions.conf)
[default]
exten => _X,1,dial(SIP/1,,M(mymacro^parameter))
[macro-mymacro]
exten => s,1,verbose(${ARG1})
How to implement the same functionality using pbx_lua?
Details: Asterisk 11.7 on Ubuntu 14.04
Kind regards
Dennis Guse
Quality and Usability Lab
Telekom Innovation Laboratories
TU Berlin
Ernst-Reuter-Platz 7
D-10587 Berlin, Germany
Tel: +49 30 8353 58874
Fax: +49 30 8353 58409
E-mail: dennis.g...@telekom.de
Web: www.qu.tlabs.tu-berlin.de
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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