[asterisk-users] delay in processing dtmf

[Danny Nicholas]

The default setting of tonedur in dahdi.conf is 80.  This means that each
DTMF key you press takes 80 ms to transmit across  the "line".  When mine
got changed to 300, it made my in-house calls take 2-5 seconds to connect
and outgoing calls 7-11 seconds.  Genconf_dahdi changes this if you
regenerate (I think).

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Waruna de
Silva
Sent: Monday, October 19, 2009 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] delay in processing dtmf

 

Hey thanks for that,

but can you give more details on that.




On Mon, Oct 19, 2009 at 11:46 PM, Danny Nicholas <danny at debsinc.com> wrote:

You might want to play with tonedur in dahdi.conf.  This IME effects SIP
calls as well.

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Waruna de
Silva
Sent: Monday, October 19, 2009 1:06 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] delay in processing dtmf

 

Hi,
I'm new to this list

I'm developing asterisk application where users can call and control volume
up and down in music player. 
Problem I'm getting is if users press 222228 in fast speed, system will
process all those 2s and then process 8, so there is few seconds ( around
4-5) processing key press 8 , therefore users will feel unresponsiveness in
system.(in other words users will press those six keys in 2.5 seconds and it
will take 5 seconds to process them),its not application's problem. may be
within asterisk.

this only happen when users call from physical phone, if they use softphone
eg: Xlite then this delay wont happen. 

think this is because network latency in telephony networks. Are  there any
work around or possible solutions? Please anyone know, give me
some idea.

Thanks in advance
Waruna.


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f8f15a6b/attachment.htm