[Asterisk-Users] DTMF tones not long enough on out going call
s
Low, Adam
ALow at Prioritytelecom.com
Fri Aug 22 12:03:55 MST 2003
Hmmm, interesting I was not aware of this. My experience evolves from voice switches (Nortel DMS, Lucent EXS) rather than PBX's. I am using inband DTMF within my setup as I had problems (no DTMF recognition on PSTN calls) with RFC2833 when dialing through an AS5300 and onto a DMS100.
What is the reason for this? Is the PBX actually cancelling out the DTMF tone after it itself recognises the tone? Does it also effect the DTMF tones received over a B channel from the PSTN?
-----Original Message-----
From: Adam Roach
To: 'asterisk-users at lists.digium.com'
Sent: 22/08/03 20:08
Subject: RE: [Asterisk-Users] DTMF tones not long enough on out going call s
I'll point out that the same applies in general to many
commercial PBXes. I can verify from years of personal
experience, for example, that the Ericsson MD110 (probably
the most popular PBX in Europe) exhibits precisely
the same behavior.
/a
> -----Original Message-----
> From: Eric Wieling [mailto:eric at fnords.org]
> Sent: Friday, August 22, 2003 11:17
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] DTMF tones not long enough on out going
> call s
>
>
> Have someone using a SIP device with RFC2833 signaling call you, now
> have the press and hold down one of the dialing keys. You'll hear a
> short tone then nothing.
>
> On Fri, 2003-08-22 at 11:05, Low, Adam wrote:
> > Maybe its just me but I find this question a little
> confusing, the tone duration should have no impact on tone
> recognition and typically in my experience the duration of
> the tone is defined by how long the user holds down the button !?
> >
> > > -----Original Message-----
> > > From: James Sizemore [mailto:james at deny.org]
> > > Sent: 22 August 2003 17:33
> > > To: asterisk-users at lists.digium.com
> > > Subject: [Asterisk-Users] DTMF tones not long enough on out
> > > going calls
> > >
> > >
> > > DTMF tones are not long enough on out going calls, when I'm
> > > using either
> > > "info" or rfc2833. Does anyone know if the tone length value
> > > is in rtp.c
> > > or chan_sip.c ?
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> > ********* DISCLAIMER *********
> >
> > This message and any attachment are confidential and may be
> privileged or otherwise protected from disclosure and may
> include proprietary information. If you are not the intended
> recipient, please telephone or email the sender and delete
> this message and any attachment from your system. If you are
> not the intended recipient you must not copy this message or
> attachment or disclose the contents to any other person
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> BTEL Consulting
> 850-484-4535 x2111 (Office)
> 504-595-3916 x2111 (Experimental)
> 877-552-0838 (Backup Phone)
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
********* DISCLAIMER *********
This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
More information about the asterisk-users
mailing list