Am 08.02.2010 21:15, schrieb Philippe Sultan:
Philippe, what exactly is a playback channel? Is it a pseudo participant
playing back the announcements?
Yes. Announcements are played to the conference members by creating a
channel of type 'Bridge' which streams the sound files.
thanks
On Tue, Feb 9, 2010 at 6:24 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
I wonder what mute should mean. Does it mean that the participant will
not receive any media, or that media sent by the participant will be
ignored, or both?
Please post your discoveries to:
Am 09.02.2010 15:35, schrieb David Backeberg:
On Tue, Feb 9, 2010 at 6:24 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
I wonder what mute should mean. Does it mean that the participant will
not receive any media, or that media sent by the participant will be
ignored, or both?
On Tue, Feb 9, 2010 at 10:26 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
Answering myself: muting means that the participants voice is ignored.
Thank you for updating the wiki and the list.
I looked into this when I was having problems with early 1.6.0.*
MeetMe(), specifically the
Hi!
IIRC there was an announcement some time ago that it is possible now to
make conferences without the need for DAHDI anymore - but I can not
remember the name of this feature anymore, and google didn't solved my
problem.
Thus, any references to this new system are appreciated.
thanks
8 feb 2010 kl. 12.29 skrev Klaus Darilion:
Hi!
IIRC there was an announcement some time ago that it is possible now to
make conferences without the need for DAHDI anymore - but I can not
remember the name of this feature anymore, and google didn't solved my
problem.
Thus, any
Hi Klaus,
The module is app_confbridge, and the application is ConfBridge. I had
been using it for a while because it's really easy to use : you don't
need any configuration file, and you get cool announcements upon
conference events from a playback channel.
The options work pretty much like
And by the way, app_confbridge is included in the 1.6.2 series (at least).
On Mon, Feb 8, 2010 at 1:49 PM, Philippe Sultan
philippe.sul...@gmail.com wrote:
Hi Klaus,
The module is app_confbridge, and the application is ConfBridge. I had
been using it for a while because it's really easy to
Hi Philippe!
Am 08.02.2010 13:49, schrieb Philippe Sultan:
Hi Klaus,
The module is app_confbridge, and the application is ConfBridge. I had
been using it for a while because it's really easy to use : you don't
need any configuration file, and you get cool announcements upon
conference
Am 08.02.2010 13:49, schrieb Philippe Sultan:
Hi Klaus,
The module is app_confbridge, and the application is ConfBridge. I had
been using it for a while because it's really easy to use : you don't
need any configuration file, and you get cool announcements upon
conference events from a
Philippe, what exactly is a playback channel? Is it a pseudo participant
playing back the announcements?
Yes. Announcements are played to the conference members by creating a
channel of type 'Bridge' which streams the sound files.
thanks
klaus
Further, is there somewhere a documentation
would somebody be able to recommend a good package that works with Asterisk
please ... not commercial as will be mainly used for my home office.
Best Regards,
--
This message has been scanned for viruses and
dangerous content and is believed to be clean.
SplatNIX IT Services :: Innovation
On Mon, 29 Sep 2008, Jim Boykin wrote:
Thanks Gordon Mike for the response.
What accuracy are you getting from zaptest/dahdi_test (and system info).
Two more questions:
1) Does ztdummy requires change into kernel? I am running 2.6.9 kernel.
2) What about CPU load?
I run a custom
We plan to use asterisk for conferencing. As I understand, it requires
either a separate hardware like x100p clone or ztdummy. What are the
pro cons of x100p vs ztdummy. Any other hardware suggestions for
conferencing? It should be able to handle few simultaneous
conferences.
Thanks
Jim
On Sun, 28 Sep 2008, Jim Boykin wrote:
We plan to use asterisk for conferencing. As I understand, it requires
either a separate hardware like x100p clone or ztdummy. What are the
pro cons of x100p vs ztdummy. Any other hardware suggestions for
conferencing? It should be able to handle few
Go for it.
ztdummy is not an issue.
I have used ztdummy with 220 simultaneous participants in 18
different conference groups.
At one time, I had 60 machines running simultaneously in a FARM all
of which were carrying
the same 18 conference groups with over 200 participants active on
each
Thanks Gordon Mike for the response.
What accuracy are you getting from zaptest/dahdi_test (and system info).
Two more questions:
1) Does ztdummy requires change into kernel? I am running 2.6.9 kernel.
2) What about CPU load?
Thanks
Jim
On Mon, Sep 29, 2008 at 5:02 AM, Mike Trest [EMAIL
Ajey Gore wrote:
I figured that asterisk can do conferencing if we have zap interface. The
BRI cards I have they do not have Zaptel. How do I enable conferencing on my
server?
What do you mean when you describe cards as having or not having Zaptel?
--
Alex Balashov
Evariste Systems
Web:
Hi Ajey,
which kind of BRI are you using?
Giorgio Incantalupo
Ajey Gore wrote:
I figured that asterisk can do conferencing if we have zap interface. The
BRI cards I have they do not have Zaptel. How do I enable conferencing on my
server?
Regards
Ajey
You can do conferencing without the zap interface. just modprobe
ztdummy. Its good for small conferences.
On Mon, 2008-04-14 at 23:39 -0500, Ajey Gore wrote:
I figured that asterisk can do conferencing if we have zap interface. The
BRI cards I have they do not have Zaptel. How do I enable
On Mon, 14 Apr 2008, Ajey Gore wrote:
I figured that asterisk can do conferencing if we have zap interface.
It can do conferencing without a zap interface too.
The
BRI cards I have they do not have Zaptel.
And?
How do I enable conferencing on my server?
Well you could start by reading
Hi Faraz,
yes, you can use ztdummy but it cannot completely replace Digium cards.
It depends from your hardwareI had troubles with some kind of
serversso beware.
Giorgio.
Faraz R. Khan wrote:
You can do conferencing without the zap interface. just modprobe
ztdummy. Its good for small
I figured that asterisk can do conferencing if we have zap interface. The
BRI cards I have they do not have Zaptel. How do I enable conferencing on my
server?
Regards
Ajey
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nhadie wrote:
Hi Matt,
I tried
/usr/local/src/zaptel-1.2.22.1# ./zttest -v
and it just freezes at this.
Opened pseudo zap interface, measuring accuracy...
no more outputs, when i cancelled this is what i got.
--- Results after 0
On Wed, Jan 09, 2008 at 03:36:06PM +0800, Nhadie wrote:
Hi Matt,
I tried
/usr/local/src/zaptel-1.2.22.1# ./zttest -v
and it just freezes at this.
Opened pseudo zap interface, measuring accuracy...
no more outputs, when i cancelled this is what i got.
--- Results after 0 passes
Hi Tzafrir,
cat /proc/zaptel/*
Span 1: ZTDUMMY/1 ZTDUMMY/1 1
Kernel: 2.6.18-5-686 #1 SMP
Zaptel: zaptel-1.2.20.1
OS: Debian GNU/Linux 4.0
i downgraded my zaptel from 1.2.22.1 to 1.2.20.1 but still the same.
thanks again
regards,
nhadie
Tzafrir Cohen wrote:
On Wed, Jan 09, 2008 at
Hi Steve,
I see. I have this now,
*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudodefault en
*CLI load chan_zap.so
Unable to load module chan_zap.so -- on the log file it says, it as
already loaded that's why it's unable to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nhadie wrote:
Hi Steve,
I see. I have this now,
*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudodefault en
That means the zap channel should be ok.
One thing you could do is go
I will be out of the office on Wednesday, January 9, 2008. If this is an
emergency, please call Customer Service at (877) 791-7700. Thank you.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To
Hi Matt,
I tried
/usr/local/src/zaptel-1.2.22.1# ./zttest -v
and it just freezes at this.
Opened pseudo zap interface, measuring accuracy...
no more outputs, when i cancelled this is what i got.
--- Results after 0 passes ---
Best: 0.00 -- Worst: 100.00 -- Average: 100.00
does
Hi Matt,
it seems i don't have that command.
*CLI zap show channels
No such command 'zap' (type 'help' for help)
*CLI
! abort add ael agent agi
cdr databasedebug dnsmgr dontdump
dundi
extensions feature group help
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nhadie wrote:
Hi Matt,
it seems i don't have that command.
:)
You'll need to make sure that:
1. You have zaptel compiled
2. You compile Asterisk *after* zaptel is compiled and installed
3. You have either modprobed zaptel + ztdummy or made the
Then it's time to build zaptel, then rebuild asterisk
later,
PaulH
On Tue, 2008-01-08 at 13:16 +0800, Nhadie wrote:
Hi Matt,
it seems i don't have that command.
*CLI zap show channels
No such command 'zap' (type 'help' for help)
*CLI
! abort add ael
On Tue, Jan 08, 2008 at 06:45:14PM +1300, Matt Riddell wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nhadie wrote:
Hi Matt,
it seems i don't have that command.
:)
You'll need to make sure that:
1. You have zaptel compiled
2. You compile Asterisk *after* zaptel is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tzafrir Cohen wrote:
(This tests both cases right away: gives different error messages in the
different cases)
Sweet :)
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great
Anyone got any experiences of good quality VoIP conferencing phones?
I've used Polycom analogue units in the past, and I see that they have a
SIP version (the IP4000) - but it is better/worse/as good as an analogue
version?
(ie. would I be better off with an analogue version into a TDM card
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gordon Henderson
Sent: Friday, February 09, 2007 9:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Conferencing Phones ...
Anyone got any experiences
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Friday, February 09, 2007 8:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Conferencing Phones ...
Anyone got any experiences of good quality VoIP conferencing phones?
I've used
Hi All,
I have a problem in configuring in asterisk.
I configure asterisk meetme.conf and extension.conf, but when i transfer
call to conference it give me this message and asterisk kill it self.
Ouch ... error while writing audio data: : Broken pipe
If any one knows
List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Conferencing with multiple servers
On Tue, 2006-06-20 at 15:22 +0100, Wildheart wrote:
Hi,
I am trying to join 2 asterisk servers together using a
sip channel.
This is so, if a user joins a conference on box
Hi,
I am trying to join 2 asterisk servers together using a sip channel.
This is so, if a user joins a conference on box A and another user
joins a conference on box B, providing they are in the same conference
room, the two conferences are joined via the sip channel. We only want
to join the
On Tue, 2006-06-20 at 15:22 +0100, Wildheart wrote:
Hi,
I am trying to join 2 asterisk servers together using a sip channel.
This is so, if a user joins a conference on box A and another user
joins a conference on box B, providing they are in the same conference
room, the two conferences
-Original Message-
From: Patrick [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 20, 2006 12:05 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Conferencing with multiple servers
On Tue, 2006-06-20 at 15:22 +0100
I'm currently writing some code to support conferencing in Asterisk without
using the Meetme application. The conference runs in its own thread and every
new inbound or outbound channel that is created is passed to it. This thread
runs the conference loop reading and writing frames to each
On Mon, 7 Feb 2005, Juan Jose Comellas wrote:
I'm currently writing some code to support conferencing in Asterisk without
using the Meetme application. The conference runs in its own thread and every
new inbound or outbound channel that is created is passed to it. This thread
runs the
Hi All,
This are going nice with my setup, but I m now trying to play with
conferencig. My setup uses a isdn4linux card with no zaptel drivers.
When trying configs for conferencing I m getting this errors bellow .
What do I need to do to install this pseudo drivers ?
Thanks in advance,
Paulo
Dear Paulo ,
You need *ztdummy* for
conferencing (Meetme).
-Jefferson Carvalho
Paulo Adriano wrote:
Hi All,
This are going nice with my setup, but I m now trying to play with
conferencig. My setup uses a isdn4linux card with no zaptel drivers.
When trying configs for conferencing I m getting this
I want to build a conferencing system and I'm looking for suggestions on
which application will make a better starting point.
A conference consists of: up to 20 callers, a small number (1-3) of agents
and a small number of supervisors (0-3). Multiple conferences will be
active on the same host
On Mon, 2003-10-06 at 20:47, Brian West wrote:
Works fine on my 7960 with 5.3 firmware.
bkw
Im having a similar problem with my 7960 when I receive two incoming
calls I cannot join them.
Hello,
I am trying to conference two or more calls on a Cisco 7940 phone. When I have
Im having a similar problem with my 7960 when I receive two incoming
calls I cannot join them.
ya you can't join them. That sucks.. but you can park one call, go back
to call number 1. Press conf. Dial the parking orbit.. then press join!
bkw
On Tue, 2003-10-07 at 12:09, Brian West wrote:
Im having a similar problem with my 7960 when I receive two incoming
calls I cannot join them.
ya you can't join them. That sucks.. but you can park one call, go back
to call number 1. Press conf. Dial the parking orbit.. then press
I dont see it as a bug.. I see why it don't work.. and why people think it
should. Enable # transfers.. and setup call parking to get around this.
Also if you conf very much look at app_meetme.
bkw
On Tue, 7 Oct 2003, Juan J. Sierralta P. wrote:
On Tue, 2003-10-07 at 12:09, Brian West wrote:
Brian,
Would you be kind enough to give me a brief overview of why it doesnt work. I also
appreciate the work aorund. This is something I will have to educate my soon to be
users on. We do a lot of conferencing of calls as a matter of facilitating clients'
immediate needs.
For now I will
On Tue, 2003-10-07 at 15:38, Brian West wrote:
I dont see it as a bug.. I see why it don't work.. and why people think it
should. Enable # transfers.. and setup call parking to get around this.
Also if you conf very much look at app_meetme.
I did it, problem that I have now is the
Hello,
I am trying to conference two or more calls on a Cisco 7940 phone. When I have one
inbound call and one outbound (I initiate the second call by pressing conference) I
get the join button at the bottom of the screen and I can conference.
When I initiate both calls or I receive both
Works fine on my 7960 with 5.3 firmware.
bkw
On Mon, 6 Oct 2003, Babak Pasdar wrote:
Hello,
I am trying to conference two or more calls on a Cisco 7940 phone. When I have one
inbound call and one outbound (I initiate the second call by pressing conference) I
get the join button at the
The key functionality isn't in Asterisk right now, but someone has
notified us that they've written it. We're waiting to get the patch and
disclaimer from them to incorporate the changes. In the mean time you can
use Authenticate application:
exten = 8600,1,Authenticate(4321)
exten =
Hello !
I made some changes in the extension.conf to make *
ask for a key when a person enters a conference room.
But it is not asking for any key and takes the user
directly to the conference room. I am using sip
softphones as end points connected to sip for
conferencing. Any pointers for the
58 matches
Mail list logo
