---------- Forwarded message ----------
From: Henry Coleman <[email protected]>
Date: Thu, May 12, 2011 at 12:48 PM
Subject: Re: [on-asterisk] DTMF, will it ever work?
To: [email protected]
In general settings (freepbx) the default is no t/T but their is no k/K
mentioned.
IMHO I would assume that it is not used.
H
On Thu, May 12, 2011 at 12:01 PM, Roger <[email protected]> wrote:
> Thanks, simon. So you use purely asterisk. I was wondering how to remove
> them in gui, for example, freepbx or elastix.
>
> Yajie
> Sent from my BlackBerry device
>
> -----Original Message-----
> From: "Simon P. Ditner" <[email protected]>
> Sender: [email protected]
> Date: Thu, 12 May 2011 11:12:45
> To: <[email protected]>
> Subject: Re: [on-asterisk] DTMF, will it ever work?
> I was explicitly using the tk/TK options in the Dial() command, so I
> removed them and live without the features
>
> On 4 May 2011 23:36, Roger <[email protected]> wrote:
> > hi simon,
> > sorry, can tell me how did remove the t and k option? and how did you
> confirm you are now on packet2packet mode? on Wiki page, those are
defined
> > T -Allow the calling party to transfer the called party by sending the
> DTMF sequence defined in features.conf. This setting does not
perform policy
> enforcement on transfers initiated by other methods.
> > K -Allow the calling party to enable parking of the call by
sending the
> DTMF sequence defined for call parking infeatures.conf.
> >
> > "Simon P. Ditner" <[email protected]> wrote:
> >
> >>I've done this test on 3 different platforms -- AMD Geode, Intel Atom
> >>270, and VirtualBox on x86 and AMD64. The virtualization had no impact
> >>on the outcome.
> >>
> >>When I remove the 'T' and 'K' options to Dial(), and the Packet2Packet
> >>bridging kicks in, I can dial as fast as my fingers will move.
> >>Otherwise, I'm limited to about 2 digits / second. My captures from
> >>wireshark using asterisk 1.8 show every other digit being missed when
> >>dialing at a normal rate with T or K present with a single active
> >>call, so Asterisk's in-line dtmf monitoring seems to muck things up
> >>pretty badly.
> >>
> >>On 3 May 2011 13:07, <[email protected]> wrote:
> >>> I had this with 1.4 a while back, version 1.4.21 or so fixed
this if I
> recall the version correctly, but I'll double check. Also, there was
a patch
> for the RTP.c (I think), but again I'll double check. We were having the
> DTMF issue with calling voicemail and ivr's at various other offices.
> >>>
> >>> By any chance is your system virtualised?
> >>> -----Original Message-----
> >>> From: "Simon P. Ditner" <[email protected]>
> >>> Date: Thu, 28 Apr 2011 22:47:19
> >>> To: [email protected]<[email protected]>
> >>> Cc: [email protected]<[email protected]>; Simon P. Ditner<
> [email protected]>; Asterisk Users Group<[email protected]>
> >>> Subject: Re: [on-asterisk] DTMF, will it ever work?
> >>>
> >>> I see this across a wide range of conditions, asterisk 1.4 thru 1.8,
> sip with rfc2833, using snom, polycom, and aastra handsets.
> >>>
> >>> When I sniff the traffic, I see the rfc2833 rtp go into asterisk
in the
> right order, but it comes out of asterisk in the wrong order.
> >>>
> >>> ---
> >>> Simon P. Ditner
> >>>
> >>> On 2011-04-28, at 10:29 PM, [email protected] wrote:
> >>>
> >>>> While we're at it, which device are you using? I've found some
handle
> better than others.
> >>>> ------Original Message------
> >>>> From: [email protected]
> >>>> To: Daniel Silver
> >>>> To: Simon P. Ditner
> >>>> To: [email protected]
> >>>> To: 'Asterisk Users Group'
> >>>> ReplyTo: [email protected]
> >>>> Subject: Re: [on-asterisk] DTMF, will it ever work?
> >>>> Sent: 28 Apr 2011 22:28
> >>>>
> >>>> And what DTMF mode are you trying to use on both the SIP Trunk
and the
> device? With RFC2833 on both side and Packet to Packet bridging you
> shouldn't have any issues.
> >>>>
> >>>>
> >>>> ------Original Message------
> >>>> From: [email protected]
> >>>> To: Simon P. Ditner
> >>>> To: [email protected]
> >>>> To: 'Asterisk Users Group'
> >>>> ReplyTo: [email protected]
> >>>> Subject: Re: [on-asterisk] DTMF, will it ever work?
> >>>> Sent: Apr 28, 2011 22:27
> >>>>
> >>>> Which version of asterisk are you using?
> >>>> ------Original Message------
> >>>> From: Simon P. Ditner
> >>>> Sender: [email protected]
> >>>> To: 'Asterisk Users Group'
> >>>> Subject: [on-asterisk] DTMF, will it ever work?
> >>>> Sent: 28 Apr 2011 22:22
> >>>>
> >>>> Is there some trick to getting asterisk working reliably when
calling
> >>>> into conference bridges and voicemail? No end of trouble with
DTMF on
> >>>> my home system regardless of which provider I'm using. Drives
my wife
> >>>> nuts when she's working from home.
> >>>>
> >>>>
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> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
---------------------------------------------------------------------
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> >>>> For additional commands, e-mail: [email protected]
> >>>>
> >>>>
> >>>> Sent from my BlackBerry device on the Rogers Wireless Network
> >>>
> >>>
> >>
> >>---------------------------------------------------------------------
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> >>For additional commands, e-mail: [email protected]
> >>
> >
>
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>
>
--
* Henry Coleman*
[image: Picture]
--
* Henry Coleman*
[image: Picture]
