On Mon, Apr 13, 2015 at 1:15 PM, Steve Edwards
asterisk@sedwards.com wrote:
I've got some Asterisk 11 (I could 'upgrade' if needed) hosts using meetme
and I'd like to switch to confbridge to service more callers.
Can anyone reply with their experience along the lines of 'using meetme I
On Thu, Jan 23, 2014 at 8:09 AM, Igor Dvorzhak idm...@gmail.com wrote:
snip
How to move 2 of 3 users in the MeetMe conference to the newly created
MeetMe conference? Dialplan example is welcome.
Maybe something like an AMI redirect?
Solved
On Wed, Jan 22, 2014 at 12:44 PM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Hello All,
Asterisk: 1.8.13.0
Dahdi : 2.6.2
Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686
i686 i386 GNU/Linux
OS : CentOS 6.4
When I show meetme room details
So this web-meetme applicationrequires to enable the real time in asterisk?
Where I can find documentation about web-meetme application?
Regards
Bilal
On Tuesday, October 1, 2013 6:57 PM, Dan Austin dan_aus...@phoenix.com wrote:
Look at Web-MeetMe ( http://sf.net/projects/web-meetme )
If
Look at Web-MeetMe ( http://sf.net/projects/web-meetme )
If you are on Asterisk 1.6.7 or later you have access to RealTime
MeetMe conference storage, otherwise you need to use a
script and Asterisk application included with the WMM download.
Dan
From: asterisk-users-boun...@lists.digium.com
exten = 123,1,Set(TIMEOUT(absolute)=3600)
exten = 123,n,MeetMe(blah,d)
if you are using freepbx and you want to set timeout for all conference rooms
go here -http://dn.forceit.ru/asterisk-conference-timeout
--
_
--
Sent from my Verizon Wireless 4G LTE DROID
Dan Austin dan_aus...@phoenix.com wrote:
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
I don't get what the 'F' option is for. Its not proper to exit a context
and then reenter the conference as admin
Isn't there any other way to do actions such as kick/mute/unmute users by
admin dtmf trigger?
On Fri, Jun 1, 2012 at 3:47 AM, Steve Edwards asterisk@sedwards.comwrote:
On
This list was accurate up to and including Asterisk 11
[0] = Caller #
[1] = Callerid Number
[2] = Callerid Name
[3] = Channel:
[4] = 1 for Admin, NULL for User
[5] = 1 for Monitor, Null otherwise
[6] = 1 for Muted, NULL for UnMuted
[7] = 1 for Resquests Floor, 0
Thanks Dan, I found the list arguments from app_meetme.c for asterisk 1.6.x
There doesn't seem to be any interface for [8] = Requests Floor.
How can we put initially muted users in the request to talk queue?
The provision of this parameter in the meet-me source indicates this is
doable... but I am
: Thursday, August 15, 2013 12:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] meetme list concise
Thanks Dan, I found the list arguments from app_meetme.c for asterisk 1.6.x
There doesn't seem to be any interface for [8] = Requests Floor.
How can we put
Thiago wrote:
I'm trying to limit the number of participants in a conference room
with the realtime option maxusers, but it doesn't work.
Asterisk version?
Any error messages?
Is the conference you are attempting to limit stored in a db (Realtime)?
Dan
--
On Fri, Jul 19, 2013 at 2:52 PM, Johan Wilfer li...@jttech.se wrote:
2013-07-19 15:35, Thiago Coutinho skrev:
Hi all.
I'm trying to limit the number of participants in a conference room
with the realtime option maxusers, but it doesn't work.
Someone have this option working properly?
Try
Hi Johan.
But the option maxusers should work too, right?
On Fri, Jul 19, 2013 at 2:52 PM, Johan Wilfer li...@jttech.se wrote:
2013-07-19 15:35, Thiago Coutinho skrev:
Hi all.
I'm trying to limit the number of participants in a conference room
with the realtime option maxusers, but it
2013-07-19 15:35, Thiago Coutinho skrev:
Hi all.
I'm trying to limit the number of participants in a conference room
with the realtime option maxusers, but it doesn't work.
Someone have this option working properly?
Try these:
https://wiki.asterisk.org/wiki/display/AST/Function_GROUP
On 06/02/2013 08:36 PM, Patrick Lists wrote:
Hi,
Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I
know for example if a conf ended normally or if someone gave a wrong
conf number or pin?
Thanks,
Patrick
There is no channel variable that provides that level of
On 06/03/2013 06:47 PM, Matthew Jordan wrote:
On 06/02/2013 08:36 PM, Patrick Lists wrote:
Hi,
Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I
know for example if a conf ended normally or if someone gave a wrong
conf number or pin?
Thanks,
Patrick
There is no
Hello,
what is the equivalent parameter of X in the ConfBridge()-command ?
How can you exit ConfBridge by pressing a digit ?
Concerning MeetMe() :
Verbosity is 25 and I still don't see anything on the console or in
the logs when pressing '0' (zero).
Kind regards,
Jonas.
On 02/20/2013
- Original Message -
From: Jonas Kellens jonas.kell...@telenet.be
I've tried now from Cisco SPA 508G and from Yealink T-28 to exit
Meetme() by pressing '0' (zero) but no success.
As I said, to log in I need to give password 12340 and that goes very
well ! Once inside the conference
- Original Message -
From: Jonas Kellens jonas.kell...@telenet.be
But nothing happens when pressing 0 (zero).
Why not check the logs in /var/log/asterisk/full ?. Make sure you have the
full log enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type
messages going to it.
Hello,
I don't really see anything when pressing '0' (zero). It's like the '0'
(zero) does not reach Asterisk.
However the password to enter the conference does reach Asterisk well.
Kind regards,
Jonas.
On 02/20/2013 03:32 PM, Rusty Newton wrote:
- Original Message -
From:
- Original Message -
From: Jonas Kellens jonas.kell...@telenet.be
Hello,
I don't really see anything when pressing '0' (zero). It's like the
'0' (zero) does not reach Asterisk.
However the password to enter the conference does reach Asterisk
well.
Please don't top post
Hello,
what is the equivalent parameter of X in the ConfBridge()-command ?
How can you exit ConfBridge by pressing a digit ?
Concerning MeetMe() :
Verbosity is 25 and I still don't see anything on the console or in the
logs when pressing '0' (zero).
Kind regards,
Jonas.
On 02/20/2013
Jerry Geis wrote:
I am running asterisk 1.4.43 on a really small network for testing, all
on same switch.
I launch a meetme between my server and 5 asterisk clients that
are all on 10 foot network cables all connected to the same switch.
The meetme is fine everything is in sync
Then I reboot
By not in sync do you mean that there is a delay between when the
speaker speaks and when the client hears it?
There's always going to be some amount of delay. It takes time to encode
the audio, send it, mix it (in this case), receive it, decode it, and
have it pass through a jitterbuffer
Jerry Geis wrote:
I think I have a race condition.
I am running something like this in my dialplan
call agi to bring my list of devices into my MeetMe
Playback beep
start MeetMe()
So in fact the meetme is not started before I bring the list
of devices into the meetme.
How can I do this
Can you clarify what you mean by MeetMe to be active? What MeetMe
options are you using and what is your configuration like? With the
proper combination of options it shouldn't matter who gets into the
conference bridge first. This is what Page essentially does, with the
difference being that
- Original Message -
From: Jerry Geis ge...@pagestation.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 1, 2012 5:01:43 PM
Subject: [asterisk-users] MeetMe
I am using Meeting on 1.4.43 with a handfull of
On Mon, 1 Oct 2012, Jerry Geis wrote:
I am using Meeting on 1.4.43 with a handfull of devices, like 10 to 20
in a Meetme.
I can tell a difference (as two of the devices are close to each
other) that they are not fully in sync.
You would have to measure how many ms they are 'out of sync' to
Nothing happens at the same time, unless you're broadcasting information
over some transport that supports multicast sends. There's always going to
be some interspersing of transmissions, if for no other reason than each
participant's channel in the conference has to be serviced after the media
- Original Message -
From: Jerry Geis ge...@pagestation.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 1, 2012 8:01:06 PM
Subject: Re: [asterisk-users] MeetMe not fully in sync
Mathew,
Makes sense does
On Thu, 31 May 2012, Daniel Knoll wrote:
is it possible to read the DTMF tones from a caller while he is in a
meetme conference? I would like to read the pressed key sequence and
call a command like MeetMeAdmin or System Commands. I'm using Asterisk
1.8.7.
I'm just a 1.2 Luddite, but...
Daniel wrote:
Hi Group,
is in MeetMe any option to identify the own number (from the view of a
caller)?
I would like to write an option to set on the telephone an request for voice,
if the room is muted. That request should display on our Conference Control
Website and an Admin
Don't think so. You can set up in the dialplan to skip meetme if the count
is 0 or use meetmeadmin to kick out the user when he/she is the last one.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton
Sent: Tuesday, April
On 07-03-12 11:44, David Klaverstyn wrote:
Hi All,
Can someone please tell me if it is possible and if so how do I go about
streaming a live conference to the internet for internet users to listen
to? I’d hope to be able to do thus dynamically as conferences are
created and internet users can
2012-01-20 20:09, Matt Hamilton skrev:
Hi,
Once in a while when a SIP channel connected to meetme conference is
hung up, I start getting the following error multiple times:
WARNING[14031]: app_meetme.c:3668 conf_run: Unable to write frame to
channel Local/100203@h
The status of the
as in use or hold.
It's really hard to duplicate it - it seems to happen more under heavier load
though.
Matt
Date: Sun, 22 Jan 2012 13:36:07 +0100
From: li...@jttech.se
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] meetme - Unable to write frame to channel
: [asterisk-users] meetme with IVR
Any one is help ?
Best Regards,
Mahesh Katta
On Mon, Jan 16, 2012 at 10:41 PM, mahesh katta maheshka...@flexydial.com
wrote:
Hi all,
please help me.
how we can configure between call add the IVR.
My scenarios is
A get the incomming call from C.In between
...@lists.digium.com] *On Behalf Of *mahesh katta
*Sent:* Tuesday, January 17, 2012 1:36 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] meetme with IVR
** **
Any one is help ?
Best Regards,
Mahesh Katta
On Mon, Jan 16, 2012 at 10:41 PM, mahesh
Any one is help ?
Best Regards,
Mahesh Katta
On Mon, Jan 16, 2012 at 10:41 PM, mahesh katta maheshka...@flexydial.comwrote:
Hi all,
please help me.
how we can configure between call add the IVR.
My scenarios is
A get the incomming call from C.In between them I need to one side IVR
play
In article CA+xMSg43sQ=ichydct27dvbjgwkmot3npab0fc2m_libsrh...@mail.gmail.com,
Karim Mardhani ka...@vertexcommunication.ca wrote:
Hi everyone,
I am trying to get Meetme to return back to the context from where it
joined the meetme. For example a user uses the following context to join a
hello,
when i use the number of the first provider like that
exten = 520870900,1,Answer
exten = 520870900,n,Wait(4)
exten = 520870900,n,Meetme
All works without problem,the issue just with the second provider i use just
the last 3 numbers for the outbound all works without issue, but whe i use
hi,
you are using pattern matching and not using the right syntax
like that.
exten = _520,1,answer
like that.
On 5 Oct 2011 21:47, salaheddine elharit salah.elharit...@gmail.com
wrote:
Hello list
i have one question related to meetme,i have to providers with the first
one
i put the
Doug Lytle wrote:
I've been searching the Jira issue tracker and found a ticket:
What I ended up doing was to copy the app_meetme.c out of the 1.4.30
source and compiled it into my current Asterisk setup. I now have PIN
prompts.
Doug
--
Ben Franklin quote:
Those who would give up
On Mon, Jul 11, 2011 at 8:22 AM, Doug Lytle supp...@drdos.info wrote:
Doug Lytle wrote:
I've been searching the Jira issue tracker and found a ticket:
What I ended up doing was to copy the app_meetme.c out of the 1.4.30 source
and compiled it into my current Asterisk setup. I now have PIN
That patch to 1.8 was a very simple change: modify one line, add another
line. Should be easy and straight-forward to replicate on 1.4.42. (Not using
1.4 anymore over here, otherwise I would've provided the patch.)
--
_
--
On Sat, Jul 9, 2011 at 11:31 AM, Doug Lytle supp...@drdos.info wrote:
I've just put into place an updated meetme server. I went from 1.4.20.1 to
1.4.42.
In testing, it would seem that dynamically created conferences will not
prompt for the PIN. I've read though the readme and even went as
Steve Totaro wrote:
I guess you could do it the old fashioned way until you open a ticket
I've been searching the Jira issue tracker and found a ticket:
https://issues.asterisk.org/jira/browse/ASTERISK-16747
Not being familiar with the new Jira system, I can't seem to find a
patch for the
Hi,
You can use
Meetme(1234,dL(1800))
where 1800 = 6 hours after 6 hours channel is hanf up
regards
Dhaval
On Mon, Apr 18, 2011 at 9:31 PM, Bryant Zimmerman brya...@zktech.comwrote:
Is there a way to place a hangup time on a dynamic Meetme conference. I am
using Page() with a
On Thu, Apr 21, 2011 at 4:03 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:
Hi,
You can use
Meetme(1234,dL(1800))
where 1800 = 6 hours after 6 hours channel is hanf up
regards
Dhaval
On Mon, Apr 18, 2011 at 9:31 PM, Bryant Zimmerman brya...@zktech.comwrote:
Is there a
hey just change following
[status-one-en]
exten = 100,1,Meetme (12345,qdM)
exten = 100,1,Hangup()
Channel: Local/100@status-one-en
CallerID: Rick 55
MaxRetries: 0
RetryTime: 15
WaitTime: 45
Application: Playback
Data: my_status_message
On Mon, Apr 4, 2011 at 10:38 PM, D. Rick
Check out the Random Application and the RAND function, Here is a
quick untested example for either.
exten = s,1,Answer
exten = s,n,Background(privacy-please-stay-on-line-to-be-connected)
exten = s,n,Random(33:${CONTEXT},s,FILE1) ; 33% Num1
exten = s,n,Random(33:${CONTEXT},s,FILE2) ; 33% Num2
On Thu, Feb 10, 2011 at 04:58:05PM -0800, John Jolly wrote:
i am trying to configure the meetme conference (asterisk 1.8) to play a *
random* sound file from a specific directory prior to it dropping the caller
into the conference itself.
Absent an Asterisk-specific solution, how about a
On Thu, 10 Feb 2011, John Jolly wrote:
i am trying to configure the meetme conference (asterisk 1.8) to play a
random sound file from a specific directory prior to it dropping the
caller into the conference itself. i am able to successfully get it to
play a specific file prior to entering the
In article 1296748085.2237.16.camel@shaft,
Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Is there an option on MeetMe that means the conference room is only
available if an admin user is logged in?
I've had a look the the application from the asterisk cli but I can't
really see what I'm
On Thu, 2011-02-03 at 16:39 +, Tony Mountifield wrote:
In article 1296748085.2237.16.camel@shaft,
Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Is there an option on MeetMe that means the conference room is only
available if an admin user is logged in?
I've had a look the the
On 12/21/2010 10:15 PM, sean darcy wrote:
On 12/21/2010 10:03 PM, sean darcy wrote:
On 12/21/2010 12:13 PM, Jeremy Betts wrote:
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com wrote:
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
On 12/21/2010 12:13 PM, Jeremy Betts wrote:
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
I'm trying to migrate from MeetMe to ConfBridge:
On 12/21/2010 10:03 PM, sean darcy wrote:
On 12/21/2010 12:13 PM, Jeremy Betts wrote:
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
I'm
Thanks all,
I realised after posting 2 things.. 1) I needed to also cover MOH
outside of meetme. And that 2) theres a bug in 1.4.18 where the
defaults aren't reloaded properly for MOH, and you have to do a server
stop/start to get them to reload.
Thanks,
Adrian
--
Adrian Marsh wrote:
Thanks all,
I realised after posting 2 things.. 1) I needed to also cover MOH
outside of meetme. And that 2) theres a bug in 1.4.18 where the
defaults aren't reloaded properly for MOH, and you have to do a server
stop/start to get them to reload.
Thanks,
Adrian
Discussion
Cc: Adrian Marsh
Subject: Re: [asterisk-users] Meetme and MOH
Adrian Marsh wrote:
Thanks all,
I realised after posting 2 things.. 1) I needed to also cover MOH
outside of meetme. And that 2) theres a bug in 1.4.18 where the
defaults aren't reloaded properly for MOH, and you
Hi Carlos,
you have to incllude the conference options (user ad admin) in the meetme
table and put schedule=yes in meetme.conf file
On the dialplan just call the conference like:
exten = 1557,1,Meetme(905)
Regards
- Bakko
--
On Thu, Nov 18, 2010 at 12:37 PM, Adrian Marsh
adrian.ma...@ubiquisys.com wrote:
Hi,
With a dynamic Meetme using: MeetMe(|DsMrc)
How do I control which context MOH uses, other than “default” ?
Asterisk: 1.4.15
Thanks,
Adrian
--
On Thu, Nov 18, 2010 at 12:37 PM, Adrian Marsh
adrian.ma...@ubiquisys.comwrote:
Hi,
With a dynamic Meetme using: MeetMe(|DsMrc)
How do I control which context MOH uses, other than “default” ?
Asterisk: 1.4.15
In 1.4.x you would use SetMusicOnHold(class) before you called your
Hi Flavio,
try with this funtion before the line with the english meetme application
Set(CHANNEL(language)=en)
and
Set(CHANNEL(language)=pr)
before the line with the portugues meetme application
Regards
- Bakko--
_
--
Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
From: asannu...@gmail.com
To: asterisk-users@lists.digium.com
Date: Sun, 17 Oct 2010 16:36:34 -0500
Subject: Re: [asterisk-users] Meetme
Hi Flavio,
try with this funtion before the line with the english
meetme application
Hi Flavio
is:
[conference]
exten = 1001,3,Set(CHANNEL(language)=pt_BR)
exten = 1001,4,MeetMe(1001,ipdM)
exten = 1001,5,Playback(vm-goodbye)
exten = 1001,6,Hangup
Regards
- Bakko--
_
-- Bandwidth and Colocation Provided by
It works!!!
Thanks a lot!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
From: asannu...@gmail.com
To: asterisk-users@lists.digium.com
Date: Sun, 17 Oct 2010 17:56:37 -0500
Subject: Re: [asterisk-users] Meetme
Hi Flavio
is:
[conference
Hi Kai-Uwe,
thank you for your answer. but it doesn't work.
i use this dialplan.
exten = 22,n,Answer()
exten = 22,n,NoCDR()
exten = 22,n,WaitExten(2)
exten = 22,n,Set(CHANNEL(musicclass)=music)
exten = 22,n,Set(CHANNEL(language)=de)
exten = 22,n,Read(roomid,conf-getconfno,6,1)
exten =
On Tue, Sep 7, 2010 at 3:11 AM, Daniel Knoll dan...@danielknoll.de wrote:
Hi Kai-Uwe,
thank you for your answer. but it doesn't work.
i use this dialplan.
c exten = 22,n,NoCDR()
exten = 22,n,WaitExten(2)
exten = 22,n,Set(CHANNEL(musicclass)=music)
exten = 22,n,Set(CHANNEL(language)=de)
Hi Paul,
i set Answer() .. just Cut the first, my fault.
is that the normal case, to treat errors like wrong conference Room?
Daniel
Am 07.09.2010 um 15:01 schrieb Paul Belanger:
On Tue, Sep 7, 2010 at 3:11 AM, Daniel Knoll dan...@danielknoll.de wrote:
Hi Kai-Uwe,
thank you for your
I use MeetMe(,Ms) in the Dialplan and if a Conference Room does't exist
Asterisk play (conf-invalid.slin)
If i use MeetMe(${room},Ms) (value from DTMF Read) and the Conference
Room doesn't exist Asterisk don't play (conf-invalid.slin) and Asterisk
Hangup the Call.
Use the i extension
On Mon, Aug 9, 2010 at 4:36 AM, Zhang Shukun bit...@gmail.com wrote:
hi, group
there are two module can used for meeting. MeetMe and
Conference(which is a plugin)
My question is :
which is better for large conference(maybe above 100 people in a meeting)?
There's at least one more
V.Joseph
From:
Danny Nicholas da...@debsinc.com
To:
'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Date:
07/21/2010 06:36 PM
Subject:
Re: [asterisk-users] Meetme Question
Sent by:
asterisk-users-boun...@lists.digium.com
Hi ,
I am trying to add
1) Ok, I'm using now self/peer on the feature map
2) there's no space in features.conf.
toca_macaco = 1,self/caller,Playback,tt-monkeys
But it's not working yet.
On Mon, Jul 26, 2010 at 11:33 PM, Tilghman Lesher tles...@digium.comwrote:
On Monday 26 July 2010 15:20:26 Felipe Figueiredo wrote:
Guys,
I put the option X and used the MEETME_EXIT_CONTEXT and it's working
thanks for the help!!!
=)
On Tue, Jul 27, 2010 at 9:07 AM, Felipe Figueiredo
felipe.figueired...@gmail.com wrote:
1) Ok, I'm using now self/peer on the feature map
2) there's no space in features.conf.
toca_macaco =
asterisk-users@lists.digium.com
Date:
07/28/2010 01:27 AM
Subject:
Re: [asterisk-users] MeetMe
Sent by:
asterisk-users-boun...@lists.digium.com
Guys,
I put the option X and used the MEETME_EXIT_CONTEXT and it's working
thanks for the help!!!
=)
On Tue, Jul 27, 2010 at 9:07 AM, Felipe Figueiredo
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe
Figueiredo
Subject: [asterisk-users] MeetMe
toca_macaco = 123, peer, Playback,tt-monkeys
But, if, inside the room, I press 123 the sound file tt-monkeys it's not
executed.
Danny,
didn't work... I didn't find other option to make meetme accpet dtmf but
F.
On Mon, Jul 26, 2010 at 5:25 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felipe Figueiredo
I think there is a mis-communication here; If you changed features.conf so
that toca_maccao = 123 . is now toca_maccao = 9, then if you press 9,
monkeys should play.
--
_
-- Bandwidth and Colocation Provided by
That was exactly what I did...but didn't work if I insert the option
p in the meetme on the dialplan, I can leave the room pressing #, so dtmf
is working fine
On Mon, Jul 26, 2010 at 6:07 PM, Danny Nicholas da...@debsinc.com wrote:
I think there is a mis-communication here; If you
On Monday 26 July 2010 15:20:26 Felipe Figueiredo wrote:
Hi guys,
i'm trying to use the featuremap of features.conf inside the app meetme,
but it's no working.
like:
_5XXX = {
Set(DYNAMIC_FEATURES=toca_macaco);
MeetMe(${EXTEN},F); //F forces the meetme to pass DTMF
Hi ,
I am trying to add an operator assistance feature to meetme , when the user
dials '0' ,support / help desk personnel should be added to the live
conference for live support / troubleshooting.
How can i do this ? Can I edit the meetme * menu and add a new menu item '
Press '0' for
On Sat, Jun 12, 2010 at 10:50:07PM +0200, Mickael Monsieur wrote:
because... I use it! But I do not use MeetMe with!
What is the importance of providing binary packets if the conference (MeetMe
app) is impossible without compiling ??
What version (of Debian? Of Asterisk?)
Official Debian
try using confbridge in lastest asterisk version
On Sat, Jun 12, 2010 at 11:30 AM, Daniel Knoll dan...@danielknoll.de wrote:
Hi Guys,
sometimes if one caller or many callers are in a meetme Room and a new one
join the room,
then he or another caller into the same room where kickt from the
it's not so easy, because i use a mysql database (realtime) to write the room
number from a webapp into the table.
also i extend the meetme table for my web application :-/
any other things at least to show more logs from meetme or dahdi ?
Daniel
Am 12.06.2010 um 17:39 schrieb Thomas Perron:
On Fri, Jun 11, 2010 at 04:39:46PM +0200, Mickael Monsieur wrote:
What is the interest to supply binary of Asterisk, under debian for example,
while to use MeetMe it is necessary to COMPILE Asterisk ??? :-))
Mickael.
And you don't use the existing DEB package because?
--
because... I use it! But I do not use MeetMe with!
What is the importance of providing binary packets if the conference (MeetMe
app) is impossible without compiling ??
2010/6/12 Tzafrir Cohen tzafrir.co...@xorcom.com
On Fri, Jun 11, 2010 at 04:39:46PM +0200, Mickael Monsieur wrote:
What is
If that's the case what I usually do is just stop asterisk, delete the
contents of /usr/lib/asterisk/modules/ (back it up first!) and compile the
new version (don't run make samples if you want to preserve your old .conf
files).
When using extra modules (like G729 codec) be sure to follow their
On Mon, 2010-04-19 at 11:19 -0500, Alyed wrote:
If that's the case what I usually do is just stop asterisk, delete the
contents of /usr/lib/asterisk/modules/ (back it up first!) and compile
the new version (don't run make samples if you want to preserve your
old .conf files).
When using
You are right if going from 1.4.X to 1.6.2.X or similar that's the best, but
if not moving from revision, then I don't think you need to remake the
sample files.
Alyed
2010/4/19 Carlos Chavez cur...@telecomabmex.com
On Mon, 2010-04-19 at 11:19 -0500, Alyed wrote:
If that's the case what I
I guess what you meant, is you don't have a physical card to provide the
timing needed by Meetme. Then, if you are looking for dahdi to use kernel
timer, then you need not to upgrade Aterisk but Dahdi to 2.3.0
Alyed
2010/4/18 Thomas Perron thomas.per...@gmail.com
I read that I need to run
You got him wrong.
He actually want to know the steps to upgrade to version 1.6.2 so he do can
a conference bridge using confbridge instead of of meetme because he does
not have dahdi installed.
He just want to know how to upgrade from an older version to version 1.6.2
On Sun, Apr 18, 2010 at
Hi,
Vinicius, did you actually solve the choppy audio issue by compiling
Gordon's kernel? I have the same problem on the exact same Alix platform
(using kernel 2.6.31, though).
regards,
Darko
ps. Sorry everyone if this mail does not get threaded right. I've just
joined the mailing list and not
In article 3de056a31003010645x2c4481fbr5b05923d88614...@mail.gmail.com,
David Backeberg dbackeb...@gmail.com wrote:
On Mon, Mar 1, 2010 at 6:42 AM, Emrah e...@ekanet.net wrote:
I am trying to get the usernum of a user when dialing in to a MeetMe
conference. Is there somehow a possibility to
On Mon, 1 Mar 2010, Emrah wrote:
I am trying to get the usernum of a user when dialing in to a MeetMe
conference. Is there somehow a possibility to save the usernum of a
MeetMe participant into a variable? Everything should be done through
the DialPlan, no manager and no *cli.
I use 1.2,
On Mon, Mar 1, 2010 at 6:42 AM, Emrah e...@ekanet.net wrote:
I am trying to get the usernum of a user when dialing in to a MeetMe
conference. Is there somehow a possibility to save the usernum of a
MeetMe participant into a variable? Everything should be done through
the DialPlan, no manager
Hi!
Thanks a lot for your answer.
The problem with the command you mentioned is... When do I call it? If two
people happen to enter the conf at the sametime,
I have a feeling there may be some little confusion there...
Do you think I could use the agi-background option with meetme?
I am using
1 - 100 of 702 matches
Mail list logo
