You can use the caller controlls in the conference.conf.xml to implement your
own features something like mute or kick. Or do you want to mute mute other
conference members like a moderator can do this.
BR
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Von: Artem Shiyanov shiya...@gmail.com
An:
? If yes, what should i update?
Thanks again.
BR
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Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rden...@tng.de http://www.tng.de
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Register: Amtsgericht Kiel, HRB 6596 KI
Supervisory board
Hello,
there is no more problem. I installed 1.0.4 and everthing is fine. :-)
BR
- Ursprüngliche Mail -
Von: Rudolf Denert rden...@tng.de
An: freeswitch-users freeswitch-users@lists.freeswitch.org
Gesendet: Dienstag, 4. August 2009 13:14:43 GMT +01:00
Amsterdam/Berlin/Bern/Rom
(514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
Am 2-Aug-09 um 8:18 AM schrieb Rudolf Denert:
Hello,
sorry but I don't understand your answer.
BR
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Von: Michael Jerris m...@jerris.com
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Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097
] Language of the speech
we don;t have any german sound files?
On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:
Hi again,
does anybody know why my freeswitch doesn't play German speech-files?
Here is my construct:
I'm generating a random number in my lua script:
rand = math.random(11, 1000
()
sofia/external/anonym...@anonymous.invalid SET [default_language]=[de]
But I don't here any German speech only the English on
Does anybody of you have an idea? Thanks a lot again.
Best regards
--
--
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone
Hello,
I´m getting always the message:
[ERR] mod_conference.c:5463 conference_new() Unable to install caller controls
group 'test'
I made a new caller-controls in the conference.conf.xml. It has the name
test. I also implemented the line:
param name=caller-controls value=test
in my
Hallo everybody!
I would like to play soundfiles in a existing conference.
The procedure is this:
Someone calls the number of the conference. Then this person types the pin in
to his phone. The next step is that he has to say the name for example John.
This file is saved in a special folder.
be in
the correct context? Review the default config again.
/b
On Jun 17, 2009, at 8:42 AM, Rudolf Denert wrote:
Context features not found
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in context features
2009-06-17 15:38:21 [WARNING] mod_dialplan_xml.c:263 dialplan_hunt() Context
features not found
Does anybody has an idea?
Greetz
--
--
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
Hello,
I can give the all-clear! It was my mistake ( ...and it was a silly one :-/ ) I
had to applications that interfere each other.
They are:
!--action application=start_dtmf/ --
!--action application=start_dtmf_generate data=true/--
I don't know why I skip that in my
Hello!
Is there a possibility to detect or scan which DTMF mode is sent by the
calling CPE so that I can establish logical interrogation in my configuration?
Greetz
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and the inband of the DTMF... triple DTMF. :)
/b
On Jun 3, 2009, at 10:35 AM, Rudolf Denert wrote:
Is there a possibility to activate all DTMF detection modes (in-band, SIP INFO
RFC 2388) in the same dialplan or maybe in the same extension of the
dialplan?
Greetz
Brian West
br
Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: Re: [Freeswitch-users] Insufficient RTP stream
Without a sip trace its hard to tell.
/b
On Jun 3, 2009, at 10:29 AM, Rudolf Denert wrote:
How can I avoid the problem? It seems that RTP auto adjust generates the error.
Maybe I can
to an suspicious
IP. There are over 5 packets in number. Not till then there is the correct
destination IP (see packet 386). But this could be the fact that the freeSWITCH
produces an error.
Does anybody have an idea?
Greetz
--
--
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D
you see a message about RTP auto adjusting the IP.
On Wed, Jun 3, 2009 at 6:18 AM, Rudolf Denert rden...@tng.de wrote:
Hello,
I have a question about the RTP stream. I made two calls with different
devices. One had no problems, the other call made some difficulties. I put an
extraction
: Dienstag, 2. Juni 2009 21:29:19 GMT +01:00
Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: Re: [Freeswitch-users] Prolems with DTMF
On Tue, Jun 2, 2009 at 11:37 AM, Rudolf Denert rden...@tng.de wrote:
Hello,
I'm not sure which one is it. But I think I send the digits in RFC 2833. All
Hello,
I have problems with sending DTMF. The freeswitch server recieves the digits
from my telefon (SIP account) if I activate DTMF with:
action application=start_dtmf/
in my dialplan.
The line
--param name=dtmf-type value=rfc2833/--
in sofia.conf.xml is active, too.
But than I have the
: [Freeswitch-users] Prolems with DTMF
On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert rden...@tng.de wrote:
Hello,
I have problems with sending DTMF. The freeswitch server recieves the digits
from my telefon (SIP account) if I activate DTMF with:
action application=start_dtmf/
in my dialplan
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