Displaying 20 results from an estimated 30000 matches similar to: "MeetMe - new e and E flags?"
2008 Mar 04
1
Clustering Meetme over multiple boxes?
Has anyone here done any work on clustering Meetme conferences over
multiple Asterisk boxes? The scenario I am thinking of is where there are
two or more boxes connected to a set of PRIs that all answer to the same
PSTN number, and where it's not possible to know in advance on which box
a call would arrive. So it would be possible to have some calls on one
box and some on another, that should
2004 Aug 27
2
Someone please try MeetMe MOH with latest CVS and GS phone
I have today reported a bug with the latest channel.c (1.134) that
affects music-on-hold for the first user in a MeetMe room when calling
from a Grandstream BT102. The music is broken up about 5-10 times a
second. It doesn't happen when calling from Firefly. It is also fine
on both clients with 1.133 of channel.c. I am using the ALAW codec.
Mark at Digium can't reproduce the problem,
2011 Apr 20
2
No voice in MeetMe for SIP with
Thanks a lot Tony and Dhaval for your much appreciable suggestions.
Regards,
Rajib
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com
Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com
Date: Wed, 20 Apr 2011 13:55:25 +0530
From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>
2004 Aug 16
1
Is "Meetme" a generic term?
Just a trivial question: was the term "Meetme" invented for Asterisk
as something like a brand name for its conferencing? Or was it an
existing generic term for dial-in conferencing?
Cheers
Tony
--
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org
2005 Apr 27
6
Redirect two channels to each other?
I've been scratching my head trying to think of a way to do this, but
without success yet.
I'm using the Manager API. If I have two channels linked to each other
(i.e. direct connection), or even if they are independent channels,
I can transfer them both to the same extension by using Action: Redirect
and using Channel: for one and ExtraChannel: for the other. This is most
useful for
2007 Oct 03
6
Best config for 12 FXO system?
I have a client who wants a Meetme box with 12 FXO ports, to connect
to Analogue lines coming from an Ericsson PBX.
It looks like I could do this with four different hardware configurations:
a) three TDM04B cards (based on TDM400P)
b) one TDM04B and one TDM808B
c) one TDM804B (or TDM854B?) and one TDP808B
d) one TDM2403B (half filled TDM2400P)
Apart from considerations of cost and PCI slot
2014 Feb 20
2
G729 - what happens if licences used up?
I haven't been able to find the answer online, and am not currently
able to conduct an experiment to find the answer...
I understand that in a SIP call where G729 has been negotiated as the
preferred codec, a G.729 licence is not consumed until there is a need
to perform transcoding, e.g. play a non-g729 sound, or do voicemail,
or enter a Meetme, etc.
What happens when a SIP call in progress
2011 Apr 19
3
No voice in MeetMe for SIP with AGI_BACKGROUND
Hello List,
I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.
2005 May 18
1
Audio flutter on OH323 output?
Hi, I'm using OH323, mostly with success, to interface Asterisk to
a provider's switch (World Telecom INX). I have noticed a particular
effect, and I wonder whether anyone else has seen the same?
The effect is audio flutter (almost like the flutter one gets on
MF or HF radio sometimes) which only happens intermittently.
Audio coming into Asterisk is unaffected, as proved by using the
2004 May 18
0
MeetMe conference delay increasing
I've just noticed a strange behaviour with a MeetMe conference.
I have a pair of phones (GS BT102) on my desk, and dialled both of them
into a conference on speakerphone. If I spoke or made a sound, I heard
it replayed from both speakers together a split second later, as
expected.
I went away for about 15 minutes, leaving the conference running.
When I came back any sound I made came back
2006 Jan 17
2
MeetMe Listen Only flag (|m)
One of the features that I thought would be popular with the Web-MeetMe
suite is the ability to start all non-admin callers in a muted state and
selectively unmute them. For example any large conference that is
of an announcment nature with a Q&A session.
It's probably a feature I should have tested better, but I just
discovered
that a caller that is joined to a MeetMe with the |m flag
2006 Oct 25
0
Re: Meetme... No channel type registered for'zap'
> -----Original Message-----
> From: Tzafrir Cohen [mailto:tzafrir.cohen@xorcom.com]
> Sent: Wednesday, October 25, 2006 10:18 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Re: Meetme... No channel type registered
> for'zap'
>
>
> On Wed, Oct 25, 2006 at 10:06:02AM -0600, Douglas Garstang wrote:
> > > -----Original
2008 Jul 24
7
How to detect whether running on VMware?
Does anyone know how a program, script or shell user can best determine
whether the machine is running on bare metal or is a VMware guest?
Cheers
Tony
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org
2005 Oct 12
5
delays with IAX2 and Meetme
Hi there
I am using IAX2 softphones dialing into meetme conferences. I also have
jitterbuffer=yes, with typical jitterbuffer settings. The problem I am
having is that as soon as there is a delay from a participant, then the
delay continues until the participant hangs up and dials in again. When
dialing in again the delay seems to go.
It seems to me as though as soon as the server registers
2006 Jun 13
7
delay in MeetMe
Hi All!
I am facing some delay in conferencing.
Using DIAX for Voip calls ,no hardware used yet
I am using MeetMe to achieve conferencing and am having a lot of delays.
Can anyone tell me how to reduce the delay
Regards,
Amna Saleem
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2006 Oct 24
10
Meetme... No channel type registered for 'zap'
When I call meetme:
exten => 1000,1,Answer
exten => 1000,n,Meetme(|||d)
Asterisk is complaing with:
-- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack
-- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack
-- Playing 'conf-getconfno' (language 'en')
Warning, flexible rate not
2006 Oct 13
1
Digium TE410P LED problem
Has anyone else experienced a problem with the LED for span 1 on a TE410P
or TE405P?
I had a TE410P on which the span 1 LED would not light red, but once the
span was connected, it did correctly light green.
I RMAed the board to our UK distrbutor and received a replacement. However,
the replacement board displayed the same problem!
Wondering if it was related to the computer I was putting it
2005 Jun 16
5
meetme - conf-invalid
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 zaptel cards in my system,
and there for zaptel is not loading.
Can anyone shed some light
2013 Jun 19
1
fail2ban with standard Apache log format?
I want to use fail2ban on CentOS 6 to monitor Apache with the standard
default logfile format ("combined"). Has anyone here succeeded in doing so?
The format has the IP at the start of the line, followed by two dashes
(if no authentication) and THEN the timestamp. What I've read on the
fail2ban wiki seems to say that the timestamp must ALWAYS be at the start
of the line, followed by
2006 Apr 25
3
Background asynchronous AGI
I have been writing a lot of AGI programs in C with good success.
I would like somehow to have an AGI program continue in the background
while the pbx execution returns to the dialplan and continues. Is this
possible? I was thinking that perhaps I could fork or create another
thread within the AGI prog.
The reason I want to do so is in order to monitor external information
(e.g. credit limit and