Hi all,
After loading 1.6.0.1, I notice that I always have the "VOX" effect
on Meetme conferences whether I have the "o" option set in the dial plan
or not. Is anyone else seeing this?
Although I'm now running 1.6.0.1, I'm also seeing this on a system
still running 1.6.0beta9.
Tzafrir Cohen schrieb:
> On Thu, Sep 11, 2008 at 02:58:59AM -0700, Pezhman Lali wrote:
>> Dear,
>> I have some limitations to install zaptel because of kernel reinstalling.
>
> Zaptel does not require completely reinstalling the kernel. But it does
> require loading an extra kernel module.
>
>> a
On Thu, Sep 11, 2008 at 02:58:59AM -0700, Pezhman Lali wrote:
> Dear,
> I have some limitations to install zaptel because of kernel reinstalling.
Zaptel does not require completely reinstalling the kernel. But it does
require loading an extra kernel module.
> also there is'nt any zaptel device in
thanks for reply
is ztdummy installed Independence of kernel,?
--- On Thu, 9/11/08, Quan Nguyen, (NCS) <[EMAIL PROTECTED]> wrote:
From: Quan Nguyen, (NCS) <[EMAIL PROTECTED]>
Subject: RE: [asterisk-users] meetme without zaptel
To: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]&
You only need to install ztdummy.
It's usually straightforward if you have Linux kernel 2.6.
-quan
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pezhman Lali
Sent: September-11-08 5:59 AM
To: asterisk
Subject: [asterisk-users] meetme without zaptel
Dear,
I have some limita
Dear,
I have some limitations to install zaptel because of kernel reinstalling.
also there is'nt any zaptel device installed in the server.
but I need to install meetme, for conferencing .
can u help me ?
Best
Mani
___
-- Bandwidth and Colocatio
Hi,
I was wondering if there's any sense in increasing audiobuffer above the
minimal '2' in meetme, if every channel is already dejittered before
(Local/.../nj - as described at:
http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/)
Will it help in anythin
On Tue, Jul 15, 2008 at 04:43:44PM -0400, John covici wrote:
> > http://en.wikipedia.org/wiki/?-law_algorithm
>
> OK, thanks -- I was a bit puzzled because if I want to play audio over
> asterisk, it has to be 16-bit 8khz signed, so that is what I thought
> ulaw was -- thanks for the clarificatio
On Tuesday 15 July 2008 14:24:30 John covici wrote:
> on Tuesday 07/15/2008 Tilghman Lesher([EMAIL PROTECTED])
> wrote
>
> > On Tuesday 15 July 2008 13:32:12 Artie Gold wrote:
> > > Does anyone know of a replacement for meetme that provides native G729
> > > support? The transcoding back and for
OK, I guess I need to show my ignorance -- what is the difference
between ulaw and signed linear?
on Tuesday 07/15/2008 Tilghman Lesher([EMAIL PROTECTED]) wrote
> On Tuesday 15 July 2008 13:32:12 Artie Gold wrote:
> > Does anyone know of a replacement for meetme that provides native G729
> > su
That makes sense -- thanks!
--ag
On Tue, Jul 15, 2008 at 1:59 PM, Tilghman Lesher <
[EMAIL PROTECTED]> wrote:
> On Tuesday 15 July 2008 13:32:12 Artie Gold wrote:
> > Does anyone know of a replacement for meetme that provides native G729
> > support? The transcoding back and forth from/to 711 is
On Tuesday 15 July 2008 13:32:12 Artie Gold wrote:
> Does anyone know of a replacement for meetme that provides native G729
> support? The transcoding back and forth from/to 711 is eating too much
> processor for what we're doing...
Buy a hardware transcoder board. There is simply no way to mix c
Folks:
Does anyone know of a replacement for meetme that provides native G729
support? The transcoding back and forth from/to 711 is eating too much
processor for what we're doing...
Many thanks,
--ag
--
Artie Gold
F4W, Inc.
___
-- Bandwidth and Coloc
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
FaberK wrote:
>> FaberK wrote:
>>> On Mon, Jul 7, 2008 at 2:48 PM, Philipp Ott <[EMAIL PROTECTED]> wrote:
Hi!
FaberK schrieb:
> My question is, is it possible to cut off that request to"press one"?
>
I think you want to get
> FaberK wrote:
>> On Mon, Jul 7, 2008 at 2:48 PM, Philipp Ott <[EMAIL PROTECTED]> wrote:
>>> Hi!
>>>
>>> FaberK schrieb:
My question is, is it possible to cut off that request to"press one"?
>>> I think you want to get rid of the number-pressing. The only option to
>>> omit this seems to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
FaberK wrote:
> On Mon, Jul 7, 2008 at 2:48 PM, Philipp Ott <[EMAIL PROTECTED]> wrote:
>> Hi!
>>
>> FaberK schrieb:
>>> My question is, is it possible to cut off that request to"press one"?
>>>
>> I think you want to get rid of the number-pressing. The
On Mon, Jul 7, 2008 at 2:48 PM, Philipp Ott <[EMAIL PROTECTED]> wrote:
> Hi!
>
> FaberK schrieb:
>> My question is, is it possible to cut off that request to"press one"?
>>
>
> I think you want to get rid of the number-pressing. The only option to
> omit this seems to be option E - select an empty
Hi!
FaberK schrieb:
> My question is, is it possible to cut off that request to"press one"?
>
I think you want to get rid of the number-pressing. The only option to
omit this seems to be option E - select an empty pinless conference.
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
Regards
FaberK schrieb:
> but if I edit the sound file, remain that I have to press the "1"
> button to go ahead.
>
> Thanks to all.
>
> On Mon, Jul 7, 2008 at 1:57 PM, Philipp Kempgen
> <[EMAIL PROTECTED]> wrote:
>> FaberK schrieb:
>>
>>> we use meetme application with pin so when a customer joins he's
Hi,
but if I edit the sound file, remain that I have to press the "1"
button to go ahead.
Thanks to all.
On Mon, Jul 7, 2008 at 1:57 PM, Philipp Kempgen
<[EMAIL PROTECTED]> wrote:
> FaberK schrieb:
>
>> we use meetme application with pin so when a customer joins he's
>> prompted for his name.
>>
FaberK schrieb:
> we use meetme application with pin so when a customer joins he's
> prompted for his name.
> Then the voice say:"press one to accept the recording..."
> My question is, is it possible to cut off that request to"press one"?
Audacity. Edit the sound file.
Grüße,
Philipp Kempgen
--
Steve Totaro wrote:
> On Mon, Jul 7, 2008 at 6:58 AM, FaberK <[EMAIL PROTECTED]> wrote:
>> Hi folks,
>> we use meetme application with pin so when a customer joins he's
>> prompted for his name.
>> Then the voice say:"press one to accept the recording..."
>> My question is, is it possible to cut of
On Mon, Jul 7, 2008 at 6:58 AM, FaberK <[EMAIL PROTECTED]> wrote:
> Hi folks,
> we use meetme application with pin so when a customer joins he's
> prompted for his name.
> Then the voice say:"press one to accept the recording..."
> My question is, is it possible to cut off that request to"press one
Hi folks,
we use meetme application with pin so when a customer joins he's
prompted for his name.
Then the voice say:"press one to accept the recording..."
My question is, is it possible to cut off that request to"press one"?
Thanks to all
--
.:FaberK:.
_
There's so such thing as privacy on a phone call, at least not in the
United States since warrantless wiretapping. Then expecting security
on a conference call, which by definition is open to many parties, is
silly.
Depending on your state (in the US), you may need to disclose when and
if you do r
Hi:
I configured an asterisk server for conference call service but I have
a problem now :Does asterisk have an option to secure and warranty
meetme,in the other word,How can I play up users that their conference
won't hear by us in spite of asterisk can record meetme ?
I'd appreciate any help.
would look at your CPU and memory stats,
>> and run some tests and monitor that..
>>
>>
>> A.
>>
>>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of John
>> covici
>> Sent:
am, Dell 745) box. But it
> > >> will be setup-specific.. So I would look at your CPU and memory stats,
> > >> and run some tests and monitor that..
> > >>
> > >>
> > >> A.
> > >>
> > >>
> > &
run some tests and monitor that..
> >>
> >>
> >> A.
> >>
> >>
> >> -Original Message-
> >> From: [EMAIL PROTECTED]
> >> [mailto:[EMAIL PROTECTED] On Behalf Of John
> >> covici
> >> Sent: 08
would look at your CPU and memory stats,
>> and run some tests and monitor that..
>>
>>
>> A.
>>
>>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of John
>> covici
>> Sent
and run some tests and monitor that..
>
>
> A.
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of John
> covici
> Sent: 08 June 2008 16:34
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re:
Behalf Of John
covici
Sent: 08 June 2008 16:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MeetMe Limits
12 people is nothing -- I do 20 regularly -- however you may want to
have them come in as muted or tell them to mute themselves, because the
latency c
12 people is nothing -- I do 20 regularly -- however you may want to
have them come in as muted or tell them to mute themselves, because
the latency can cause very severe echoes if they are on a speaker
phone or cell phone.
on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote
> Actually I think they
Actually I think they will all be calling in using regular pstn phones
and cell phones.
Sam
Al Baker wrote:
> The 2 big questions are:
> -Are all participants using QoS end to end ?
>
> -Are all of them using the SAME CODEC. As the amount of Transcoding goes up,
> the work on the * box goes up
The 2 big questions are:
-Are all participants using QoS end to end ?
-Are all of them using the SAME CODEC. As the amount of Transcoding goes up,
the work on the * box goes up and can be a problem.
Sam wrote:
> I am thinking about using my asterisk server to host a conference with
> about 12 ot
I am thinking about using my asterisk server to host a conference with
about 12 other people from around the USA. Bandwidth issues aside, will
this work or will all the different latencies cause issues? Yea I know,
I could just "try it and find out" but it is going to take alot of time
to get
Anyway to disable the join conference notification on the party that is
joining, but not the parties already in the conference?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or upda
Hi All,
How can i enable time condition on meetme? below i would like to deny
callers if the time is not yet the scheduled time of the conference, but
it seems like its still goes to 600,2, hope anyone can help.
[meet-me-test]
exten => 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,600,3)
e
On Wed, Apr 30, 2008 at 5:14 AM, Nhadie Ramos <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> got some issues with meetme, i created meetme 8000, i have users 200 and 201
> dial to 8000, meetme works fine.
>
> but i need an outside user to join, so that user will dial-in via the sip
> trunk. sip trunk has
Hi all,
got some issues with meetme, i created meetme 8000, i have users 200 and 201
dial to 8000, meetme works fine.
but i need an outside user to join, so that user will dial-in via the sip
trunk. sip trunk has did e.g. , outside user dials-in to that DID,
asterisk rcvs it forward it
On Sat, Apr 19, 2008 at 11:36:23PM -0700, Steve Edwards wrote:
> On Sat, 19 Apr 2008, Nhadie Ramos wrote:
> > [meet-me-test]
> > exten => 600,1,GotoIfTime(11:00-12:00|*|20|Apr?meetnow)
> > exten => 600,n,Answer
> > exten => 600,n,Festival('Conference is not yet active, please dial in on
> > the as
On Sat, 19 Apr 2008, Nhadie Ramos wrote:
> [meet-me-test]
> exten => 600,1,GotoIfTime(11:00-12:00|*|20|Apr?meetnow)
> exten => 600,n,Answer
> exten => 600,n,Festival('Conference is not yet active, please dial in on the
> assigned time');
> exten => 600,n,Hangup
> exten => 600,n(meetnow),MeetMe(60
Hi Steve,
Thanks for the reply. I made it look like this:
[meet-me-test]
exten => 600,1,GotoIfTime(11:00-12:00|*|20|Apr?meetnow)
exten => 600,n,Answer
exten => 600,n,Festival('Conference is not yet active, please dial in on the
assigned time');
exten => 600,n,Hangup
exten => 600,n(meetnow)
On Sun, 20 Apr 2008, nhadie ramos wrote:
> How can i enable time condition on meetme? below i would like to deny
> callers if the time is not yet the scheduled time of the conference, but it
> seems like its still goes to 600,2, hope anyone can help.
>
> [meet-me-test]
> exten => 600,1,GotoIfTime(
Hi All,
How can i enable time condition on meetme? below i would like to deny
callers if the time is not yet the scheduled time of the conference, but
it seems like its still goes to 600,2, hope anyone can help.
[meet-me-test]
exten => 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,6
Hi All,
How can i enable time condition on meetme? below i would like to deny
callers if the time is not yet the scheduled time of the conference, but it
seems like its still goes to 600,2, hope anyone can help.
[meet-me-test]
exten => 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,600,3)
ext
I have been playing with meetme option b.
I was wanting to use a spool call file going to a local channel,
then bring a list of asterisk boxes into a meetme in talk only mode.
then run the AGI to play a wave file then hangup.
It seems to be doing all this but I hear no audio.
I seem to get pipe e
Jerry Geis wrote:
> Jerry Geis wrote:
>> I am running asterisk 1.4.18 trying to use MeetMe and option b.
>>
>> I am getting permissions denied failed to execute conf-background.agi
>> on the CLI
>>
>> lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi ->
>> /home/silentm/bin/conf-bac
Jerry Geis wrote:
> /home/silentm/bin/conf-background.agi
> my conf background is a symbolic link - then my permissions are :
>
> [EMAIL PROTECTED] src]# ls -l /home/silentm/bin/conf-background.agi
> -rwSr-Sr-- 1 root root 81 Mar 17 10:44 /home/silentm/bin/conf-background.agi
>
I'm guessing bec
In article <[EMAIL PROTECTED]>,
Jerry Geis <[EMAIL PROTECTED]> wrote:
> I am running asterisk 1.4.18 trying to use MeetMe and option b.
>
> I am getting permissions denied failed to execute conf-background.agi
> on the CLI
>
> lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi ->
>
Jerry Geis wrote:
> I am running asterisk 1.4.18 trying to use MeetMe and option b.
>
> I am getting permissions denied failed to execute conf-background.agi
> on the CLI
>
> lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi ->
> /home/silentm/bin/conf-background.agi
> my conf backg
I am running asterisk 1.4.18 trying to use MeetMe and option b.
I am getting permissions denied failed to execute conf-background.agi
on the CLI
lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi ->
/home/silentm/bin/conf-background.agi
my conf background is a symbolic link - then
In article <[EMAIL PROTECTED]>,
preethy varghese <[EMAIL PROTECTED]> wrote:
> I searched both AsteriskWin32 0.66b build from Asterisk 1.2.26.2 and
> AsteriskWin32 0.60b build from Asterisk 1.2.14 for Meetme application. But
> both are showing "The application is not registered" and that module
Hi,
I searched both AsteriskWin32 0.66b build from Asterisk 1.2.26.2 and
AsteriskWin32 0.60b build from Asterisk 1.2.14 for Meetme application. But
both are showing "The application is not registered" and that module
(app_meetme.so) is also not present.Plz anybody can tell me the reason for
th
Perfect! Thanks.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield
Sent: Tuesday, February 19, 2008 11:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MeetMe Admin Functions
In article <[EMAIL PROTECTED]>,
In article <[EMAIL PROTECTED]>,
Jeremy Mann <[EMAIL PROTECTED]> wrote:
>
> Is there any way that I can have an admin user hit * and then Mute all other
> users in a
> meetme conference? Sort of a moderator function?
>
> I know it can be done with MeetMeAdmin, but as I see it that requires a
>
Is there any way that I can have an admin user hit * and then Mute all other
users in a meetme conference? Sort of a moderator function?
I know it can be done with MeetMeAdmin, but as I see it that requires a
separate extension to dial, unless I've got the logic wrong?
If it can be done in a s
Hi,
I have asterisk installed in the xen virtual server.
I installed zaptel 1.4.2.1 and patched it to have ztxen module.
I loaded ztxen module but when I try to invoke or call to my meetme
application
I get the following warning and negative result of connecting to conference:
[Feb 5 17:46:13] WA
Ubuntu has a real time kernel in repository "apt-get install linux-rt" . So
you dont need to recompile . I think debian should also have one in
repository , or u can manually compile a real time enabled kernel . Here's
what is shows with real time patched kernel .
dmesg|grep ztdummy
[ 53.29307
Matthew J. Roth a écrit :
> Administrator TOOTAI wrote:
>
>> This is not true if you're using B410P cards. We always face timing
>> problem as we can't -Asterisk stability issues- add X100P or TDM400P
>> with those cards
>>
> Daniel,
>
> I thought that using an empty TDM400P as a timing s
Administrator TOOTAI wrote:
> This is not true if you're using B410P cards. We always face timing
> problem as we can't -Asterisk stability issues- add X100P or TDM400P
> with those cards
Daniel,
I thought that using an empty TDM400P as a timing source may no longer
be the best solution due to
Matthew J. Roth a écrit :
> [...]
>
> I settled on using an empty TDM400P as a timing source, because it is a
> simple solution that "just works." This may still be your best bet, but
> I'll defer judgment on that to the list because Asterisk has evolved
> quite a bit since I made that decision
I have,
I have ztdummy module loaded in the kernel
On Feb 1, 2008 11:59 AM, Gordon Henderson <[EMAIL PROTECTED]>
wrote:
> On Fri, 1 Feb 2008, Tomasz Zieleniewski wrote:
>
> > Hi,
> >
> > I have the following problem that when someone connects to my conference
> and
> > is the only member
> > musi
On Fri, 1 Feb 2008, Tomasz Zieleniewski wrote:
> Hi,
>
> I have the following problem that when someone connects to my conference and
> is the only member
> music on hold is played just for one second or less and then stops:
>
> [Feb 1 10:38:46] -- Started music on hold, class 'default', on c
Hi,
I have the following problem that when someone connects to my conference and
is the only member
music on hold is played just for one second or less and then stops:
[Feb 1 10:38:46] -- Started music on hold, class 'default', on channel
'SIP/sip.touk.pl-0083dad0'
[Feb 1 10:38:46] -- S
Tomasz Zieleniewski wrote:
> ztttest results show value below 99,98:
>
> [EMAIL PROTECTED]:~/src/zaptel-1.4$ ./zttest -v -c 5
>
> --- Results after 11 passes ---
> Best: 50.003 -- Worst: 49.612 -- Average: 49.931827, Difference:
> 49.931827
This is the first thing I would address. Get that aver
On Jan 30, 2008 5:48 PM, Matthew J. Roth <[EMAIL PROTECTED]> wrote:
> Tomasz Zieleniewski wrote:
> > I am using Debian OS kernel 2.6.22-3-amd64
> > and zaptel driver 1.4 with ztdummy module for meetme application.
> > I use meetme with SIP channels.
> >
> > I have such problem that when one conne
On Jan 30, 2008 10:35 PM, Dan Austin <[EMAIL PROTECTED]> wrote:
> Franklin wrote:
> > ztdummy can give you issues as a timing device.
> Yes and no. See below
>
> > Any way you could try using a Digium card just
> > as a timing device to see if this helps?
>
>
> Tomasz wrote:
> >> I am using Debia
Franklin wrote:
> ztdummy can give you issues as a timing device.
Yes and no. See below
> Any way you could try using a Digium card just
> as a timing device to see if this helps?
Tomasz wrote:
>> I am using Debian OS kernel 2.6.22-3-amd64
>> and zaptel driver 1.4 with ztdummy module for meetm
Tomasz Zieleniewski wrote:
> I am using Debian OS kernel 2.6.22-3-amd64
> and zaptel driver 1.4 with ztdummy module for meetme application.
> I use meetme with SIP channels.
>
> I have such problem that when one connects to the conference voice is
> "cut".
> Each voice sequence is disturbed.
>
>
n"
Sent: Wednesday, January 30, 2008 11:23:57 AM (GMT-0500) America/New_York
Subject: [asterisk-users] Meetme voice quality problems
Hi,
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme application.
I use meetme with SIP channels.
I h
Hi,
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme application.
I use meetme with SIP channels.
I have such problem that when one connects to the conference voice is "cut".
Each voice sequence is disturbed.
Does any one have similar issue and cou
Just set the variable ${MEETME_RECORDINGFORMAT} to the desired format
and voila its done.
Have fun.
on Tuesday 01/15/2008 Lees, James (UK)([EMAIL PROTECTED]) wrote
>
> Hello,
>
> Is there a way to change the format from the default?
>
> 'r' - Record conference (records as ${MEETME_RECORD
In article <[EMAIL PROTECTED]>,
Lees, James (UK) <[EMAIL PROTECTED]> wrote:
>
> Hello,
>
> Is there a way to change the format from the default?
>
> 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format
> ${MEETME_RECORDINGFORMAT}). Default filename is
> meetme-conf-rec-${CONF
On Tue, 15 Jan 2008, Lees, James (UK) wrote:
>
> Hello,
>
> Is there a way to change the format from the default?
>
> 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format
> ${MEETME_RECORDINGFORMAT}). Default filename is
> meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default f
Hello,
Is there a way to change the format from the default?
'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format
${MEETME_RECORDINGFORMAT}). Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. -
requires chan_zap.so
Many thanks
***
With the meetme "r" option, I know it can record a conference in wav
format. As this format eats up space like none other, is it possible to
have it record in gsm or wav/gsm format to keep it down in size?
Rob
___
-- Bandwidth and Colocation Provided by
Hello Tzafrir,
On Dec 3, 2007 11:01 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> Interesting. That explains why depmod was not run at package install
> time. So the next question is: why is that file missing? Do you have
> any guess?
It is because this is on the Xen domU where it only uses
On Mon, Dec 03, 2007 at 09:45:23PM +0800, GNUbie wrote:
> Hello Tzafrir,
>
> On Dec 3, 2007 8:50 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> >
> > What is the output of:
> >
> > uname -r
>
>
> 2.6.18-5-xen-amd64
>
> ls /boot/System.map-`uname -r`
>
>
> ls: /boot/System.map-2.6.18-5-xen-
Hello Tzafrir,
On Dec 3, 2007 8:50 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> What is the output of:
>
> uname -r
2.6.18-5-xen-amd64
ls /boot/System.map-`uname -r`
ls: /boot/System.map-2.6.18-5-xen-amd64: No such file or directory
# ls -a /boot/
. ..
This is because the Asterisk-1.4
On Mon, Dec 03, 2007 at 08:34:40PM +0800, GNUbie wrote:
> Hello Tzafrir,
>
> On Dec 3, 2007 7:24 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> >
> > You need to modprobe it. Or insmod zaptel first.
>
>
> # modprobe ztdummy
> FATAL: Module ztdummy not found.
>
> # modprobe zaptel
> FATAL: Mo
Hello Tzafrir,
On Dec 3, 2007 7:24 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> You need to modprobe it. Or insmod zaptel first.
# modprobe ztdummy
FATAL: Module ztdummy not found.
# modprobe zaptel
FATAL: Module zaptel not found.
# insmod /lib/modules/2.6.18-5-xen-amd64/misc/zaptel.ko
za
On Mon, Dec 03, 2007 at 04:11:50PM +0800, GNUbie wrote:
> Hello all,
>
> I am planning to setup a MeetMe conference functionality on
> Asterisk-1.4.13without having a Zaptel card. All users will be
> calling through SIP only.
> AFAIK, the said application needs a timer which makes use of the ztdu
Hello all,
I am planning to setup a MeetMe conference functionality on
Asterisk-1.4.13without having a Zaptel card. All users will be
calling through SIP only.
AFAIK, the said application needs a timer which makes use of the ztdummy
module. I have basically two (2) problems I am encountering her
Hello,
On Nov/06/2007, [EMAIL PROTECTED] wrote:
> Just remember if you don't have any Zaptel cards you are going to have
> to use ztdummy to run app_meetme. Ztdummy essentially requires Linux
> 2.6, which you should be using anyways.
yes, and this is the reason that we have setted up a new serve
In article <[EMAIL PROTECTED]>,
Carles Pina i Estany <[EMAIL PROTECTED]> wrote:
> On Nov/06/2007, Tony Mountifield wrote:
> > I have a number of systems that have a single Pentium 4 @ 2.8GHz (with HT),
> > 1GB RAM and a 4xE1 PRI card (TE410P), and they regularly have conferences
> > with up to 90 p
Just remember if you don't have any Zaptel cards you are going to have
to use ztdummy to run app_meetme. Ztdummy essentially requires Linux
2.6, which you should be using anyways.
On 11/6/07, Carles Pina i Estany <[EMAIL PROTECTED]> wrote:
>
> Hello,
>
> First of all: also thanks to Doug Lytle and
Hello,
First of all: also thanks to Doug Lytle and Steve Edwards. Just
answering one time to all of you.
I had the feeling that this computer, for 15 Meetme users, was more than
enough... but we wanted to avoid any last-minute surprises! Now we are
more sure that everything will work fine.
Ah y
On Tue, 6 Nov 2007, Steve Edwards wrote:
> On a dual Intel(R) Xeon(TM) CPU 3.40GHz, all calls are SIP, te410p for
> timing only, CentOS 4.5, Kernel 2.6.9-55.ELsmp, Asterisk 1.2.18, MySQL,
> various AGI's. According to "top" Asterisk is taking about 80mb and
> between 5% and 10% of a cpu.
This hos
On Tue, 6 Nov 2007, Carles Pina i Estany wrote:
> We would like to have a conference with 15 users aprox. We think that
> Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running.
Your configuration should be more than sufficient. I think (gut feel, no
hard stats) I get better audi
In article <[EMAIL PROTECTED]>,
Carles Pina i Estany <[EMAIL PROTECTED]> wrote:
>
> Hello,
>
> We would like to have a conference with 15 users aprox. We think that
> Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running.
>
> We wonder if somebody has some other experience, good
Carles Pina i Estany wrote:
> Hello,
>
> We would like to have a conference with 15 users aprox. We think that
> Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running.
>
>
I'd consider that over-kill (Means it'll work fine).
We've used a Pentium 3 866mhz with 512mb memory and h
Hello,
We would like to have a conference with 15 users aprox. We think that
Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running.
We wonder if somebody has some other experience, good or bad.
We will use Asterisk 1.2 (it is a small and short project for only
this).
Thanks!
-
> I am just wondering - it there any way how to protect a conference from
> being abused by someone?
> I know I can request pin, but that pin is then hardcoded in meetme.conf
> and normal user can not change it.
>
> I would like to establish an admin user who could set a pin for the
> conference t
Subject: [asterisk-users] Meetme - how to protect the conference?
Hi all,
I am just wondering - it there any way how to protect a conference from
being abused by someone?
I know I can request pin, but that pin is then hardcoded in meetme.conf
and normal user can not change it.
I would like to
Hi all,
I am just wondering - it there any way how to protect a conference from
being abused by someone?
I know I can request pin, but that pin is then hardcoded in meetme.conf
and normal user can not change it.
I would like to establish an admin user who could set a pin for the
conference to be
jamespev wrote:
> We are using only SIP trunks for our provider.(we have no POTS
> hardware) Is there an aggressive echo cancellation setting in this case?
No, sorry, only for Zap channels.
Moj
___
--Bandwidth and Colocation Provided by http://www.a
We are using only SIP trunks for our provider.(we have no POTS hardware) Is
there an aggressive echo cancellation setting in this case?
Could this be related to the audio buffers setting in meetme.conf?
Thanks for the ideas!
James> Are you using zap channels with 'aggressive' echo suppressi
Are you using zap channels with 'aggressive' echo suppression enabled?
That will make calls pretty half-duplex.
Moj
jamespev wrote:
>
>Hello. We are very successfully using asterisk (in the form of
> trixbox 2.2/asterisk 1.2). We run a few conference lines for customer
> teleconferences
I have not noticed this here at all -- although too much of talking
over each other makes a mess, but in both 1.2 and 1.4 I have not
noticed any such behavior. What are you using for a carrier?
on Wednesday 10/10/2007 jamespev([EMAIL PROTECTED]) wrote
>
> Hello. We are very successfully us
301 - 400 of 1037 matches
Mail list logo
