Hello,
I am trying to use a dynamic_features during a MeetMe conference without
any luck. The dynamic_features defined macro works great during a normal
call, but is ignored while on a MeetMe conference.
extensions.conf
[macro-RaiseHand]
exten => s,1,DumpChan(1)
features.conf
RaiseHand =>
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
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
was only getting x callers per server but with confbridge I now get y
callers per
Hi,
is there a way to put a conference participant in talk only mode (not
listening) using CLI or AMI like mute/unmute ?
MeetMe in Asterisk 1.8
Thanks for any hint.
--
Daniel
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Hello,
How to move 2 of 3 users in the MeetMe conference to the newly created
MeetMe conference? Dialplan example is welcome.
Best,
Igor
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New to
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?
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 using meetme list command it shows Minus
in activity column.
Any Idea.
meetme list
Conf Num Parties
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
Of bilal ghayyad
Sent: Tuesday, October 01, 2013 12:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] meetme conference password and time limitation
Hello;
We need to have admin page, so the administrator can create passwords to be
used to join the meetme conferences and can
Hello;
We need to have admin page, so the administrator can create passwords to be
used to join the meetme conferences and can determine the allowed time ..
Well, the admin interface can be done easy (I do not know if there is something
ready), and the password and the time limitation can be
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Tuesday, October 01, 2013 12:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] meetme conference password and time limitation
Hello;
We need to have admin page, so the administrator can create
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
--
_
--
Hello fellow asterisk users,
I've been facing a problem when using MeetMe's admin functionality to
unmute users in a conference using *Asterisk 1.6.2.11*.
I've tried:
1) MeetMeUnmute (AMI)
2) MeetMeAdmin(AMI)
3) MeetMeChannelAdmin(AMI)
and also tried via console : asterisk -rx 'meetme unmute
Sent from my Verizon Wireless 4G LTE DROID
Dan Austin dan_aus...@phoenix.com wrote:
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New to Asterisk? Join us for a live introductory webinar every
Hello,
Can anyone tell me the format for meetme list concise command, so that I
know what field is what (separated by '!'s)
Thanks
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New to Asterisk? Join
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
Discussion
Subject: [asterisk-users] meetme list concise
Hello,
Can anyone tell me the format for meetme list concise command, so that I know
what field is what (separated by '!'s)
Thanks
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** **
Dan
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *[Digital^Dude] ®
*Sent:* Thursday, August 15, 2013 4:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users
: 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
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?
--
thiagoc
O povo não deveria temer o governo. O governo é quem deveria temer o povo.
V de Vingança
--
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
hello list ,
i want to use meetme with asterisk1.4 i check in this forum and i found
this code :
exten = 508,1,MeetMe(1000,ipdM)
when i use this code in my server i can say my name and i press 1 in order
to enter in the conference ; but i want to asks the customer to press an
number and
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
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
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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
Hello,
using Asterisk 1.8.12.2
I am having trouble with exiting the conference room by entering a
single digit.
option X of the Meetme()-application should do this.
I have following in extensions.conf :
/exten = _1000X,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite)//
//exten =
- 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
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 one of the
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
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 differently so the
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. One was slightly behind the other... Any way to get
them more in sync?
Is it the delay from starting
- 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
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
Hi Group,
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.
Thanks for help
Daniel
--
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
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 should unmute this person.
Is it possible to have a meetme conference timeout (and go to the next line in
the dialplan) if there is only one participant left?
Thanks,
Matt
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_
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03, 2012 11:41 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] meetme timeout if only one participant
Is it possible to have a meetme conference timeout (and go to the next line
in the dialplan) if there is only one participant left?
Thanks,
Matt
We use MeetMe with res_timing_dahdi as the timing source, and once a while we
get the following error which then causes Asterisk to crash/restart (with safe
Asterisk).
ERROR[6518] res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0
sample timer ticks
According to the following
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 tune in via a browser or streaming through
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
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 channel is not updated, and the only way to get
: [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
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 for C, C enter the some DTMF inputs and A should be on hold.
after finish C input will complete again they want talk each other .This
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
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
conference, once user hangs up I would like to continue executing the rest
of the dialplan. But when caller hangs up from the conference
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
Hello list
i have one question related to meetme,i have to providers with the first one
i put the number with 9 digit 520XX and all works without issue, with
the second i put just the last 3 numbers 500 with meetme there is nothing
but when i put the last 3 numbers like below i can call
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.)
--
_
--
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 far as
reading the 1.2 to 1.4 upgrade document.
s far as I can see,
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
Is there a way to place a hangup time on a dynamic Meetme conference. I am
using Page() with a Meetme conf and I have had a few instances where
someone from a wifi voip phone looses ip while doing a page and the page
never hangs up. I have to kill it. I need to somehow limit the page so
after
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
Ok, I've been running applications on 1.4 for quite some time using
meetme to hold a person, while the person on the other end of the call
accepts, etc. I was playing status messages to the calling party using a
context like this:
[status-one-en]
exten = 100,1,Playback(my_status_message)
exten =
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
i have been trying to find a way to accomplish the following but have not
found a method in which to do so:
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
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
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
;;exten=_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=_8[1-9],n,Hangup
And that works.
Also changed the hints:
;;exten = 81,hint,MeetMe:81
exten =
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
I have a server running 1.6.2.13 that uses realtime for most
configurations. Everything works fine except for meetme. When I use
Meetme with Realtime any options specified in the dial plan are ignored.
For example:
exten = 1557,1,Meetme(905,icM(somemusic))
With realtime I just
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
--
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
--
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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 ,
Is it possible to have two meetme room in asterisk 1.6 which each one have a
different language? I mean, one room the annoucement is in Portuguese an
another in english?
Today I can change over the sip.conf and it is valid for all room.
regards!
Att,
Flavio Roberto Miranda
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
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