If I use meetme conference room, can I invite another user during a
conversation?
In which way?
Matteo
===
Matteo Piazza, Junior Researcher
CREATE-NET
Via Solteri, 38 - 38100 Trento - Italy
email: [EMAIL PROTECTED]
Tel: +39-0461-408400ext:308
Matteo Piazza wrote:
If I use meetme conference room, can I invite another user during a
conversation?
In which way?
Use a .call file (search for sample.call in the asterisk source directory for
an example).
You can then copy the file into /var/spool/asterisk/outgoing to make the call.
--
On Wed, 2005-11-09 at 21:59 +1300, Matt Riddell wrote:
Matteo Piazza wrote:
If I use meetme conference room, can I invite another user during a
conversation?
In which way?
Use a .call file (search for sample.call in the asterisk source directory for
an example).
You can then copy the
trixter aka Bret McDanel wrote:
On Wed, 2005-11-09 at 21:59 +1300, Matt Riddell wrote:
Matteo Piazza wrote:
If I use meetme conference room, can I invite another user during a
conversation?
In which way?
Use a .call file (search for sample.call in the asterisk source directory for
an
On Wed, 2005-11-09 at 23:00 +1300, Matt Riddell wrote:
trixter aka Bret McDanel wrote:
For anyone doing this it may not always be /var/spool/asterisk/outgoing,
especially with non linux installs. check your asterisk.conf file for
astspooldir. It should be in that directiory/outgoing :)
I don't have app_meetme.so file neither in /usr/lib/asterisk/modules,
nor /usr/src/asterisk/apps. How to get it?
--
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Bartosz Piec
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Hi all
when i try to the conference the number i am getting
the following error in asterisk console. i am using
the g729 codec in asterisk and my sip devices but i
can able make the call between the device.
error:
Nov 7 16:07:49 NOTICE[3190]: channel.c:1703
ast_set_write_format: Unable to find
You will need to buy some g729 licences...
PaulH
- Original Message -
From: nr k [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, November 07, 2005 9:42 PM
Subject: [Asterisk-Users] meetme conference getting error using codec g729
Hi all
when i try
Hi,
Seem to be a G729 licences issue.
Have you buy G729 licences ?
Regards,
Le lun 07/11/2005 à 11:42, nr k a écrit :
Hi all
when i try to the conference the number i am getting
the following error in asterisk console. i am using
the g729 codec in asterisk and my sip devices but i
can
Hi all
I am having Asterisk 1.0.9. now i configured the
meetme conference with conference number 1234 and also
i add the extension 1234 in extension.conf.if i call
to 1234 asterisk says it's invalid conference number.
i am having both sccp and sip devices.
[room]
; Usage is conf = confno[,pin]
I am having Asterisk 1.0.9. now i configured the
meetme conference with conference number 1234 and also
i add the extension 1234 in extension.conf.if i call
to 1234 asterisk says it's invalid conference number.
i am having both sccp and sip devices.
[room]
; Usage is conf = confno[,pin]
Have you checked your zaptel interface. If you don't have hardware then use ztdummy.
I guess you would have.
~Vamsi
On 11/6/05, nr k [EMAIL PROTECTED] wrote:
Hi allI am having Asterisk 1.0.9. now i configured themeetme conference with conference number 1234 and alsoi add the extension 1234 in
Hi
I configured the meetme number in the area where i
specified the other extensions but still i am having
pbm. herewith i am sending the error i got in the
asterisk console.
Nov 6 19:07:35 WARNING[4952]: chan_zap.c:770 zt_open:
Unable to open '/dev/zap/pseudo': No such device or
address
Nov
nr k wrote:
Hi
I configured the meetme number in the area where i
specified the other extensions but still i am having
pbm. herewith i am sending the error i got in the
asterisk console.
Nov 6 19:07:35 WARNING[4952]: chan_zap.c:770 zt_open:
Unable to open '/dev/zap/pseudo': No such device or
If my memory serves you need some kind of zaptel device for meetme to
work -- I think even dummy will do, but you need something.
on Sunday 11/06/2005 nr k([EMAIL PROTECTED]) wrote
Hi
I configured the meetme number in the area where i
specified the other extensions but still i am having
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Meetme Conference-reg
If my memory serves you need some kind of zaptel device for meetme to
work -- I think even dummy will do, but you need something.
on Sunday 11/06/2005 nr k([EMAIL PROTECTED]) wrote
Hi
I
Hi all
i am having Asterisk 1.0.9. now i configured the
meetme conference with conference number 1234.I have
both sccp ande sip device.if i use the codec ulaw i
can able make call between sip and sccp devices and
also put meetme conference.if i use g.723.1 codec i
have pbm in conference and call
Hi,
I would like to know the possibility of sending DTMF to other users in a meetme.
I'm looking at inviting a participant from within the conference, here
the participant is another conference bridge. So we need to send PIN to
this conference bridge. How can I bypass the IVR detect menu and send
Hello,
We wrote a small AGI script to do just this. We just drop it's exten
into the conference room and pass it the digits you want played and it
will play the audio files of DTMF digits to all participants in the
meetme room. It works great for us and we've been using it for over 2
years now.
.
Thanks,
~Vamsi
-- Forwarded message --
From: Matt Florell [EMAIL PROTECTED]
Date: Nov 4, 2005 10:03 PM
Subject: Re: [Asterisk-Users] Meetme: Sending DTMF to other users in a
conference
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Hi All,
* noob here :)
What I'm trying to do is have a meetme number that
streams a recording.
Let's say there was a company pressconference
live that people could join and then later, a cleaned-up version was
avaialable.
Live is fine, I just set up a conference where
everyone comes in
Hi,
I have been doing some tests with app_meetme, all the clients i used
were SIP clients, and i have noticed that MeetMe continues to decode
the channels of the clients even if they are just connected to a
conference as listeners or muted. This really affects the performance
of Asterisk since
There is an Meetme command option 'a' for admin. I tried using this option
and noticed that it allows users to login with the user pin as well as the
admin pin. In my dialpan I have:
exten = 700, 1, Meetme(500,Mas)
And in meetme.conf, I have:
conf = 500,1234,
After dialing extension 700,
I've noticed that when I use the MeetMe app, it shows the Zap/pseudo
context
in a fax context. Anyone know what could cause this??
from my extensions.conf
exten = conf,1,Answer
exten = conf,2,SetMusicOnHold(rachmaninov)
exten = conf,3,Macro(isadmin)
exten = conf,4,GotoIf(${CONFADMIN}?5:7)
I have install Flash Operator Panel but Asterisk show
this message:
WARNING[3564]: pbx.c:1650 pbx_extension_helper: No
application 'Meetme' for extension (conferences, 101, 1)
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open /etc/asterisk/modules.conf and add the following:
load app_meetme.so
save and close file; reload asterisk
Fabio Montemaggiore wrote:
I have install Flash Operator Panel but Asterisk show
this message:
WARNING[3564]: pbx.c:1650 pbx_extension_helper: No
application 'Meetme' for extension
I have MeetMe setup to pick an empty conference room, and it works great.conf,10,MeetMe(,aMXq)However, I also set the flag (X) to exit the conference room with any key, but I need a wayto get back to the room with the user having to key in the extension numbers.I'm familiar with this MeetMe
Hello,I have MeetMe setup to pick an empty conference room, and it works great.conf,10,MeetMe(,aMXq)However, I also set the flag (X) to exit the conference room with any key, but I need a wayto get back to the room with the user having to key in the extension numbers.I'm familiar with this MeetMe
I think I configured the MeetMe right. Since I am using SIP for
inbound calls I followed the
instruction, for 2.6 kernel, from this web page:
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
When I call the MeetMe number I get the greeting to enter in your
conference room. I do and get
I think I configured the MeetMe right. Since I am using SIP for
inbound calls I followed the
instruction, for 2.6 kernel, from this web page:
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
When I call the MeetMe number I get the greeting to enter in your
conference room. I do
kurt x wrote:
exten = _15551232432,2,Meetme
exten = _15551232432,3,Hangup
Try exten = _155512324232,2,MeetMe(|Msicp)
Doug
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Hi,
Thank you very much for your suggestion this was what i nedded.
Best Regards
Accursio Avona
The question is, how can i indicate the marked user?
A quick search of the archives reveals:
Example:
meetme.conf
conf = 1000
extensions.conf
; ** Normal users enter the conference
Hi all,
I'd like to use the w option of the meetme application.
>From tiki i read:
'w' wait until the marked user enters the conference
All other connected users will hear MusicOnHold until the marked
user enters.
The question is, how can i indicate the "marked user"?
thank's in
On Tue, September 13, 2005 11:53, Accursio Avona said:
Hi all,
I'd like to use the w option of the meetme application.
From tiki i read:
'w' -- wait until the marked user enters the conference
* All other connected users will hear MusicOnHold until the marked
user enters.
Hi all,
I'd like to use the w option of the meetme application.
From tiki i read:
'w' — wait until the marked user enters the conference
* All other connected users will hear MusicOnHold until the marked
user enters.
The question is, how can i indicate the marked user?
thank's in
Accursio Avona wrote:
Hi all,
I'd like to use the w option of the meetme application.
From tiki i read:
'w' — wait until the marked user enters the conference
* All other connected users will hear MusicOnHold until the marked
user enters.
The question is, how can i indicate the
Is there a way to have a Meetme room dial an extension? For example, is
there a way to use the meetme as the channel in an originate command?
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Yes, just use 'Channel: Local/[EMAIL PROTECTED]' as the Channel (where
8600100 is the meetme room exten and default is the context) in the
manager API Action.
Here's an example:
ACTION: Originate
Channel: Local/[EMAIL PROTECTED]
Exten: 1234
Priority: 1
Context: default
you can also swap the
On Aug 24, 2005, at 11:21 AM, Doug Lytle wrote:[EMAIL PROTECTED] wrote: Hello,But does not go into how to mark a user. voip-info archives, and google didn't lead me to any clue, anddigging to app_meetme.c wasn't fruitful.Anyone have an example on how they marked a user in their dialplan? Create
On Aug 24, 2005, at 7:40 PM, Doug Lytle wrote:[EMAIL PROTECTED] wrote: On Aug 24, 2005, at 11:21 AM, Doug Lytle wrote: Create an extension that the user to be marked knows about, maybe even have it authenticate, mark the user and drop them into the conference.Doug If the Marked user isn't the
Hello,The MeetMe application offers this flag:'w' — wait until the marked user enters the conferenceAll other connected users will hear MusicOnHold until the marked user enters.But does not go into how to mark a user. voip-info archives, and google didn't lead me to any clue, anddigging to
[EMAIL PROTECTED] wrote:
Hello,
But does not go into how to mark a user. voip-info archives, and
google didn't lead me to any clue, and
digging to app_meetme.c wasn't fruitful.
Anyone have an example on how they marked a user in their dialplan?
Create an extension that the user to be
[EMAIL PROTECTED] wrote:
On Aug 24, 2005, at 11:21 AM, Doug Lytle wrote:
Create an extension that the user to be marked knows about, maybe
even have it authenticate, mark the user and drop them into the
conference.
Doug
If the Marked user isn't the first to enter the channel, then
I'm currently working out the config bugs on my * box and I'm noticing
that the meetme is very scratchy. As in not usable scratchy tho I can
hear the audio it sounds like when you talk through a fan.
Anyone have any ideas? Linux 2.6 with RTC installed. Using stable
release and SIP devices.
I installed icecast-2.2.0.tar.gz and ices-2.0.1.tar.gz and referenced
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Ices.
But I could not succeed to start ices-2.0.1 as follows;
-- Attempting call on Local/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1
(Retry 1)
-- Executing
Hi, Matt and Asterisk gurus
I encountered the same problem in my asterisk meetme.
Whenever the 3rd person joins the meeting, it creates echo in the meeting,
while 2 person meeting is fine.
I am wondering if you can give me more hint on how to configure the mixer to
have echo cancelled.
We are
Hi,
the option c for the announce of the user count does not work me in * 1.0.9.
exten = ,1,Wait(1)
exten = ,2,MeetMe(|Mdcs)
And how to handel the marked mode with option A? I can't find any sample config
for this.
Regards
___
Kib Eki wrote:
Hi,
the option c for the announce of the user count does not work me in *
1.0.9.
exten = ,1,Wait(1)
exten = ,2,MeetMe(|Mdcs)
That is correct. I was able to get it to work with CVS HEAD.
And how to handel the marked mode with option A? I can't find any
sample
Has anyone attempted to change the MeetMe enter and exit sounds. I see
that the raw values in the enter.h and exit.h files. If I want to change
the sounds is it as easy as converting the auto files to .raw and place
the text in the file? I don't believe there is a header in the raw
format.
Thanks
hello
how can i install meetme application without Zaptel
interface. and if this is not posible then how to
install zaptel module.
any helpful link
thanks in advance
Kamran
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Hi all,
Can anyone help me to make my soundcard (CONSOLE) to participate
in a meetme room automatically from my dialplan. I want the soundcard to join a
meetme room when someone else joins the room.
Thanks a lot!
==
Hi,
today i have taken a strong look at meetme.c
what i am trying to accomplish is the following:
it should be possible to access an menu from within the conference in
order to perform special tasks, eg. to dial another number so that the
called person is joined with the conderence.
my
Hi All...
I set up a conference bridge using MeetMe. It works nicely, except that it
seems that certain parameters I give it are ignored or else don't work.
Here is the line from my dial plan:
exten = 6500,1,absolutetimeout,0
exten = 6500,2,MeetMe,100|ciMpPs|1234
The MOH and * work, but
I have a conference set up through MeetMe and I
can record each call coming in with the Monitor command. What I would like to
move away from is having to then generate multiple files for the final output of
these calls.
On voip-info.org, there is an 'r' option to record
the conference.
Hi all.
What is the best hardware configuration to handle this following
scenario?
- 4 IVR menu with conference applications for each option;
- Only SIP/g711 user access
- 3500 simultaneous users(800 at the beginning)
- No ZAP channels
Where is the most important point of failure? CPU?
Of Denis Galvão -
iSolve
Sent: Thursday, July 07, 2005 2:24 PM
To: Asterisk Users
Subject: [Asterisk-Users] MeetMe hardware dimensioning
Hi all.
What is the best hardware configuration to handle this following scenario?
- 4 IVR menu with conference applications for each option;
- Only SIP/g711
Hi William.
On 07 de jul de 2005, at 18:39, William Boehlke wrote:
If your users are business people they ratio to 1100 simultaneous
business
calls and you will need 6-9 Lintel servers, again depending on the
conferencing load and the transcoding.
I think that I will be in this case. That
hallo,
i just experienced that all meetme rooms share the same voice data, if i
connect to 499, it could be heard in all other rooms (498,500, 501)
could sombody help me, why does asterisk send the voice out of all rooms if
i only connect to one?
thanks for help,
alex
meetme.conf
conf =
Hello,
I seem to have a strange problem, which appeared out of nowhere.
Did anyone see something like this?
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel:
Hello,
I seem to have a strange problem, which appeared out of nowhere.
Did anyone see something like this?
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal
Hello,
I seem to have a strange problem, which appeared out of nowhere.
Did anyone see something like this?
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel:
Hello list,
I wonder if someone might be able to clear up something for me.
I recently set up asterisk and have now managed to get the MeetMe
application up and running.
When I dial the extension to access the conference/MeetMe application, the
only prompt I hear is:You are currently the
i think that the thing that really matters here is wich version of
Asterisk are you using exactly. I dont know wich version the latest
debian package is using, and i dont know wich version from CVS your
friend has compiled.
Also, its needed to show the extensions.conf configuration of your
Hello,
Thanks for the message. The exact version numbers and conf info follow:
My Asterisk:
/usr/sbin/asterisk -V = Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k
Debian/AMD64 package file name/version:
pool/main/a/asterisk/asterisk_1.0.7.dfsg.1-2_amd64.deb
Extension.conf:
exten =
the line 187 of app_meetme.c of the CVS version says:
#define CONFFLAG_INTROUSER (1 14)/* If set, user will be ask
record name on entry of conference */
in Asterisk 1.0.7 from the sources that Debian download, this flag
does not exists, actually there are many others differences in the
yeah, but if you track your executed commands that only
tracks what you have done but if someone mutes/unmutes a
conferencee from a telnet manager session you will never
know.
you can add a manager event to report mute/unmute but then
you have to have a monitor session running all the time.
and
Doing further tests, I discovered that I can successfully do MeetMe
on both server B and server C, AS LONG AS all parties are SIP
extensions registered on the same server (e.g. server B or server C).
However, when I try to bring a call from server A into a MeetMe in
server B or server C,
Hello,
I seem to have a strange problem, which appeared out of nowhere.
Did anyone see something like this?
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel:
hi,
is there any way to figure out what the mute status
is of the meetme conference participants?
i personally can no see any difference on the output:
kamikaze*CLI meetme
Conf Num PartiesMarked Activity Creation
5000 0002 N/A00:00:40 Static
*
In my little experience with Meetme, i have not found how to know if
certain user is muted or not, so im keeping track of the commands i
execute from the web interface, so i know if its muted or not. Its not
so hard to add a manager event, check manager.c to know how to add
events.
On 6/22/05,
Absolutely. Here is the CLI output. I made two attempts. First, I
dialed inbound into an extension and then tried using meetme room
0201 from Server B, which didn't work. Then I dialed inbound into the
same extension and then tried using meetme room 0215 which resides in
Server A. Note
I decided to test a similar scenario against another machine (server
C). This machine behaves in a similar way as server B. It is also
running on Gentoo. When I try to transfer a call into a conference
room, it fails. Below is the CLI output of an inbound call coming
from server A into
I have two asterisk machines. One of them has a Digium board (server
A) and the other is simply using ztdummy (server B). Server A is
running on Debian and Server B is running Gentoo. Server A is running
Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 and Server B is running
Asterisk 1.0.7.
The
it would be very helpfull (IMHO) if you post the output of the
Asterisk console with a high verbosity level. Also, show us how the
important code in your extensions.conf
best regards
On 6/21/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I have two asterisk machines. One of them has a Digium
I would us Meetme for conferance SIP--SIP fist.
my Meetme.conf:
[rooms]
conf =
my extensions.conf:
exten = ,1,MeetMe()
But :
== Parsing '/etc/asterisk/meetme.conf': Found
Jun 16 10:33:22 WARNING[12100]: chan_zap.c:916 zt_open: Unable to open
'/dev/zap/pseudo': No such file
On Thu, Jun 16, 2005 at 10:46:30AM +0200, sylvain garcia wrote:
I would us Meetme for conferance SIP--SIP fist.
my Meetme.conf:
[rooms]
conf =
my extensions.conf:
exten = ,1,MeetMe()
But :
== Parsing '/etc/asterisk/meetme.conf': Found
Jun 16 10:33:22
Hi Peoples
I am having problems with meetme, in that it responds
with conf-invalid when I dial a conference number.
I notice that there is a note with regards to
ztdummy, and the need for that to be loaded. Is this still the case?
Is meetme dependent on this module? I do NOT
use
Yes, Meetme needs timing. You can install ztdummy.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
You also need to recompile asterisk after you compile and install
zaptel.
Kevin
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Yes, meetme requires a clock source. You could try ztdummy. I tried
using an FXO card as a clock source and observed that SIP calls connected
to the conference seemed to get out of sync. Basically, after perhaps 20
minutes or so in conference there was a 2 - 3 second delay between the
time that
Yes, meetme requires a clock source. You could try ztdummy. I tried
using an FXO card as a clock source and observed that SIP calls connected
to the conference seemed to get out of sync. Basically, after perhaps 20
minutes or so in conference there was a 2 - 3 second delay between the
time
I currently have my
Asterisk set up to "monitor"(record) all audio inmyconference
room on meetme. However, Asterisk will record an "in.wav" and
"_out.wav" file for each user that joins the conference. Is there a
way to set my extensions.conf file up so it only records when user when
Hi there
I am using Meetme. Now, I know it is possible to mute a user
in a conference, but is it possible to stop a user receiving audio at a
specific time (basically when they speak) and to do this through the Manager
API.
Looking at the Asterisk wiki it seems there might be some
I have MeetMe working fine except I have a little confusion over the i
option. Is that supposed to allow for announcing to the conference the
recorded name of the individual who is entering the conference?
If so, it's not working in my verions of Asterisk HEAD. I am using
ztdummy because I
Dan Morin wrote:
Thank you both for your responses. From looking at the app_conference
page in the wiki, it seems as though it is best used for one or two
speakers and everyone else just listening. Unfortunately, my company
likes to have conferences with 15 or so people and everyone can talk.
I was searching for help on how to handle the errors that are returned
from the MeetMe application.
for instance.
1) if a user tries to join a conference that is locked, allison says that
the conference is locked and then comes back to the dialplan, however it
does not continue down the
Title: Normal
So no one has any ideas about how to get MeetMe
to work with a codec other than ulaw?
Is anyone successfully doing it?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin
Sent: Tuesday, May 03, 2005 10:26
PM
To: Asterisk Users Mailing List -
---
-Original Message-
From: Dan Morin [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 04, 2005 1:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] MEETME core uses ulaw?
So no one has any ideas about how to get MeetMe to work with a codec other
or stable releases. I'd be curious to hear how it affects
performance as well.
MATT---
-Original Message-
From: Dan Morin [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 04, 2005 1:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] MEETME core
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MEETME core uses ulaw?
I was going to recommend the same to him last night, but then I
started digging into the code there and realized they were transcoding
back to LINEAR at their core as well. Now they're
Yeah, so Im an idiotsubject
should have been MeetMe not MOH.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin
Sent: Tuesday, May 03, 2005 10:26
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] MOH Core
uses ulaw...
All,
I seem to be confused :(
Meetme won't work with the message That is not a valid conference
number, please try again even with the simplest of configurations.
Having trawled the list archives, wiki and harrased people on
#asterisk I've come to a dead-end.
I compiled ztdummy last night from
Dear All :
How can I enable the announcement Feature of Meet-me rooms ?
So that when I enter the conference room , the system ask me
about my name ,, then announce all the existing people in the room about my entrance
..
Also when I go out of the conference an
announce should be played
Hi, I just recently
installed Asterisk 1.0.7 but I cannot figure out how to install the MeetMe
application. I don't think it installed with the standard 'make install'
command. If not, how do I accomplish this?
Thanks,
Matt
___
Asterisk-Users
Matt Schwartz wrote:
Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to
install the MeetMe application. I don't think it installed with the
standard 'make install' command. If not, how do I accomplish this?
MeetMe requires Zaptel. If you do not have Zaptel installed,
MeetMe is straight forward. Follow the steps for ztdummy and
there you go conferencing
Check out www.voip-info.org for more info
Cheers,
~Vamsi
On 4/18/05, Matt Schwartz [EMAIL PROTECTED] wrote:
Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to
install the
Matt Schwartz wrote:
Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out
how to install the MeetMe application. I don't think it installed
with the standard 'make install' command. If not, how do I accomplish
this?
Thanks,
Matt
Hi there
I am using Meetme and am connecting with clients that use
VAD. The clients have been built with RTC Client API. What Meetme seems to do
is cut users off from the conference if it does not receive any audio packets
from the user for 1 minute 45 seconds. The solution I have found
Hi
Has anyone explored possible ways of billing subscribers using Meetme?
I have 2 ideas in mind. First is to use background AGI and have it
send info to billing DB. Other is to have a background process running
all the time, continuously monitoring conferences being used and
kicking all users
While researching Areski's new Web-MeetMe management gui,
I found some odd (from what I expected) behaviour). Using
the CLI to set un/mute status works but does not update the
flags, or so it appears.
Starting with a fresh conference (1 user)
*CLI meetme list 3456
User #: 1 Channel: OH323/R61
It does seem to be a bug. If you look in app_meetme.c
you will see there is no code to do this update!
I added just the one line of code that respons to the
mute command from CVS HEAD and now it works fine.
you can take a look at the meetme app in [EMAIL PROTECTED]
0.8 it is based on Asterisk
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