DTMF tones don't work on Asterisk


#1

Hi,

I tried to use SC with a conference bridge on my Asterisk (app_konference). The app has a feature to select the video using DTMF dialtones.

This works with Linphone - but unfortunately with SC it doesn't react. I can't tell if Asterisk can deal with inband DTMF, especially on the Speex codec.

How does SC send DTMF? Inband - as tones or with the signalling?

Thanks for help
Conrad


#2

Hi Conrad,

we support two types of DTMFs, sending by SIP INFO (this corresponds
to dtmfmode=info in asterisk) and sending DTMFs using RFC4733
(dtmfmode=rfc2833 in *). By default RFC4733 is enabled, to disable you
must uncheck it in Audio configuration its named telephone-event and
then SIP INFO is active.
Make sure you have the proper settings in asterisk for your account.

Hum, I've just discovered that when switching off telephony-event we
still use it and SIP INFO is never used. So for now you can test by
configuring your asterisk account with dtmfmode=rfc2833. I will check
these days whats the problem with configuring telephony event.

Cheers
damencho

···

On Tue, Jan 25, 2011 at 8:59 PM, Conrad Beckert <conrad_videokonferenz@gmx.de> wrote:

Hi,

I tried to use SC with a conference bridge on my Asterisk (app_konference). The app has a feature to select the video using DTMF dialtones.

This works with Linphone - but unfortunately with SC it doesn't react. I can't tell if Asterisk can deal with inband DTMF, especially on the Speex codec.

How does SC send DTMF? Inband - as tones or with the signalling?

Thanks for help
Conrad


#3

Thank you for the explanation.
I had this issue of Sip Communicator losing sound (i posted about it in the old mailing list). Unticking the telephone-event solved the issue (and it wasnt just Asterisk, i had problems while caslling PSTN lines too).

I think this setup is confusing - moving the telephone event option to a separate option where is clearly stated what it does would avoid confusion for others too.

Thanks,

···

On Wed, 26 Jan 2011 10:31:11 +0200, Damian Minkov <damencho@sip-communicator.org> wrote:

Hi Conrad,

we support two types of DTMFs, sending by SIP INFO (this corresponds
to dtmfmode=info in asterisk) and sending DTMFs using RFC4733
(dtmfmode=rfc2833 in *). By default RFC4733 is enabled, to disable you
must uncheck it in Audio configuration its named telephone-event and
then SIP INFO is active.
Make sure you have the proper settings in asterisk for your account.

Hum, I've just discovered that when switching off telephony-event we
still use it and SIP INFO is never used. So for now you can test by
configuring your asterisk account with dtmfmode=rfc2833. I will check
these days whats the problem with configuring telephony event.

Cheers
damencho

On Tue, Jan 25, 2011 at 8:59 PM, Conrad Beckert > <conrad_videokonferenz@gmx.de> wrote:

Hi,

I tried to use SC with a conference bridge on my Asterisk (app_konference). The app has a feature to select the video using DTMF dialtones.

This works with Linphone - but unfortunately with SC it doesn't react. I can't tell if Asterisk can deal with inband DTMF, especially on the Speex codec.

How does SC send DTMF? Inband - as tones or with the signalling?

Thanks for help
Conrad

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#4

Hallo Damian,

I've put the dtmfmode=rfc2833 into my sip.conf and it works. Now I can flip video in my MCU :wink:

Conrad

-------- Original-Nachricht --------

···

Datum: Wed, 26 Jan 2011 10:31:11 +0200
Von: Damian Minkov <damencho@sip-communicator.org>
An: Conrad Beckert <conrad_videokonferenz@gmx.de>
CC: dev@jitsi.java.net
Betreff: Re: DTMF tones don\'t work on Asterisk

Hi Conrad,

we support two types of DTMFs, sending by SIP INFO (this corresponds
to dtmfmode=info in asterisk) and sending DTMFs using RFC4733
(dtmfmode=rfc2833 in *). By default RFC4733 is enabled, to disable you
must uncheck it in Audio configuration its named telephone-event and
then SIP INFO is active.
Make sure you have the proper settings in asterisk for your account.

Hum, I've just discovered that when switching off telephony-event we
still use it and SIP INFO is never used. So for now you can test by
configuring your asterisk account with dtmfmode=rfc2833. I will check
these days whats the problem with configuring telephony event.

Cheers
damencho

On Tue, Jan 25, 2011 at 8:59 PM, Conrad Beckert > <conrad_videokonferenz@gmx.de> wrote:
> Hi,
>
> I tried to use SC with a conference bridge on my Asterisk
(app_konference). The app has a feature to select the video using DTMF dialtones.
>
> This works with Linphone - but unfortunately with SC it doesn't react. I
can't tell if Asterisk can deal with inband DTMF, especially on the Speex
codec.
>
> How does SC send DTMF? Inband - as tones or with the signalling?
>
> Thanks for help
> Conrad
>


#5

Hi Conrad,

I recently tried to reproduce the issue and couldn't.
If you put dtmfmode=info in asterisk configuration and in Audio
configurations of SIP Communicator uncheck telehone-event, is it
working for you?

Thanks
damencho

···

On Wed, Jan 26, 2011 at 8:54 PM, Conrad Beckert <conrad_videokonferenz@gmx.de> wrote:

Hallo Damian,

I've put the dtmfmode=rfc2833 into my sip.conf and it works. Now I can flip video in my MCU :wink:

Conrad

-------- Original-Nachricht --------

Datum: Wed, 26 Jan 2011 10:31:11 +0200
Von: Damian Minkov <damencho@sip-communicator.org>
An: Conrad Beckert <conrad_videokonferenz@gmx.de>
CC: dev@jitsi.java.net
Betreff: Re: DTMF tones don\'t work on Asterisk

Hi Conrad,

we support two types of DTMFs, sending by SIP INFO (this corresponds
to dtmfmode=info in asterisk) and sending DTMFs using RFC4733
(dtmfmode=rfc2833 in *). By default RFC4733 is enabled, to disable you
must uncheck it in Audio configuration its named telephone-event and
then SIP INFO is active.
Make sure you have the proper settings in asterisk for your account.

Hum, I've just discovered that when switching off telephony-event we
still use it and SIP INFO is never used. So for now you can test by
configuring your asterisk account with dtmfmode=rfc2833. I will check
these days whats the problem with configuring telephony event.

Cheers
damencho

On Tue, Jan 25, 2011 at 8:59 PM, Conrad Beckert >> <conrad_videokonferenz@gmx.de> wrote:
> Hi,
>
> I tried to use SC with a conference bridge on my Asterisk
(app_konference). The app has a feature to select the video using DTMF dialtones.
>
> This works with Linphone - but unfortunately with SC it doesn't react. I
can't tell if Asterisk can deal with inband DTMF, especially on the Speex
codec.
>
> How does SC send DTMF? Inband - as tones or with the signalling?
>
> Thanks for help
> Conrad
>